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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
Downloaded 27/11/2018 07:43:58
Link to Item http://hdl.handle.net/11285/569657

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

Y TECNOLOGÍAS DE INFORMACIÓN
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. Gabriel Campuzano Treviño, Ph.D.
Synodal 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 in me the passion for
telecommunications.
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 Scheme............................................... 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 LTE Physical layer .................................................................... 39
5.2.2 LTE MAC Protocol .................................................................... 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 Channel Capacity ..................................................................... 61
A.1.2 Wider bandwidths..................................................................... 63
A.2 Multi-carrier transmission ............................................................... 64
A.3 OFDM as a multi-carrier transmission ............................................ 64
A.3.1 OFDM implementation using IDFT/DFT ................................... 66
A.3.2 Cyclic-Prefix ............................................................................. 67
A.3.3 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 normal-CP.................................................. 14
Figure 2.9: Reference Symbols in a subframe ............................................ 15
Figure 2.10: Synchronization Signals.......................................................... 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
Figure 4.3: HARQ retransmission for the DL............................................... 34
Figure 4.4: Jitter buffer ................................................................................ 36
Figure 5.1: Simulation Scenario .................................................................. 38
Figure 5.2: LTE downlink PHY structure ..................................................... 40
Figure 5.3: Scheduler flow diagram ............................................................ 43
Figure 5.4: L1/L2 Control Region, RS, and Data Region ............................ 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
Figure 5.9: Jitter behavior (SNR = 12 dB) ................................................... 52
Figure 5.10: Jitter cell profile (SNR = 5 dB) ................................................. 53
Figure 5.11: Jitter cell profile (SNR = 8 dB) ................................................. 55
Figure 5.12: Jitter cell profile (SNR = 12 dB) ............................................... 58
Figure A.1: Operation regions ..................................................................... 62
Figure A.2: Subcarrier spacing ................................................................... 65
Figure A.3: a) FDM, b) OFDM ..................................................................... 65
Figure A.4: OFDM modulation and demodulation ....................................... 65
Figure A.5: Digital implementation of OFDM ............................................... 67
Figure A.6: There is no intra-cell interference for OFDM............................. 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 Resource configuration ....................................................... 14
Table 3.1: VoIP traffic model parameters .................................................... 24
Table 3.2: VoIP Traffic Simulation Results .................................................. 30
Table 5.1: Simulation parameters ............................................................... 40
Table 5.2: CQI and MCS recommendations ............................................... 45
Table 5.3: Cell jitter; SNR = 5 dB ................................................................ 49
Table 5.4: Cell jitter; SNR = 8 dB ................................................................ 50
Table 5.5: Cell jitter; SNR = 12 dB .............................................................. 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.
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
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
3 G P P L O N G T E R M 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 LTE. Both, the architecture and air interface are
presented.
2.1 T H 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 6
2.2 DESIGN TARGETS
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) DL average: 1.6 - 2.1 bit/s/Hz (3 to 4 x HSPA Rel. 6 )
b) UL average: 0.66 -1.0 bit/s/Hz (2 to 3 x HSPA Rel. 6)
c) DL cell edge : 0.04 - 0.06 bit/s/Hz (2 to 3 x HSPA Rel. 6)
2
d) UL cell edge : 0.02-0.03 bit/s/Hz (2 to 3 x HSPA 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 percentile - 95% of the users have better performance
th
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 is connected to the Evolved Packet Core (EPC) through the
3
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.
• eNB: Enhanced Node B, or base

M M E / S A E Gateway M M E / S A E Gateway station
• UE: User Equipment
• EPC: Evolved Packet Core
o MME: Mobility Management

Entity (Control Plane)
E-UTRAN o SAE: System Architecture

eNB eNB Evolved (User Plane)
• E-UTRAN: Evolved Universal

Terrestrial Radio Access Network
eNB
UE
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.
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
MAC scheduler Hybrid ARQ Hybrid-ARQ
Redundancy version
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)
Figure 2.2: DL LTE protocols
2.4 L T E PHYSICAL LAYER
2.4.1 BANDWIDTHS, FREQUENCY BANDS A N D 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
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 1920 - 1980 2110 - 2170 FDD Europe, Asia
2 1850 - 1910 1930 - 1990 FDD Americas (Asia)
3 1710 - 1785 1805 - 1880 FDD Europe, Asia (Americas)
4 1710 - 1755 2110 - 2155 FDD Americas
5 824 - 849 869 - 894 FDD Americas
6 830 - 840 875 - 885 FDD Japan
7 2500 - 2570 2620 - 2690 FDD Europe, Asia
8 880 - 915 925 - 960 FDD Europe, Asia
9 1749.9 - 1784.9 1844.9 - 1879.9 FDD Japan
10 1710 - 1770 2110 - 2170 FDD Americas
11 1427.9 - 1452.9 1475.9 - 1500.9 FDD Japan
12 698 - 716 728 - 746 FDD Americas
13 777 - 787 746 - 756 FDD Americas
14 788 - 798 758 - 768 FDD Americas
17 704 - 716 734 - 746 FDD -
33 1900 - 1920 1900 - 1920 TDD Europe, Asia (not Japan)

34 2010 - 2025 2010 - 2025 TDD Europe Asia
35 1850 - 1910 1850 - 1910 TDD -
36 1930 - 1990 1930 - 1990 TDD -
37 1910 - 1930 1910 - 1930 TDD -
38 2570 - 2620 2570 - 2620 TDD Europe
39 1880 - 1920 1880 - 1920 TDD China
40 2300 - 2400 2300 - 2400 TDD Europe, Asia
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 fo
The subcarrier spacing is Δf = 15 KHz . Likewise the OFDM symbol duration is Tu=1/Δf = 66.7 µs. Both co
4
of an OFDM signal is given by the relationship BW = Nc. Δf, where Nc is thenumberofsubcarri
Δf = 1/TU
Pulse shape
T = 1/Δf
u
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 should be preferably selected asN =2
FFT FFT
n
for some integer , so OFDM can be performed by means of the efficient

radix-2 IFFT/FFT. The sampling rate is given as fs = Δf • NFFT = 15,000 •NFFT,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 , corresponding to the sample
s
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
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
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].
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 = 3 072 00xT ;= 10 ms

f s
One slot, T slot = 15360xT ; = 0.5 ms

s
#0 #1 #2 #3 #18 #19
One subframe
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.
LTE slot: 0.5 ms

15360 samples
(Assumed Sampling Frequency f = 30.72 MHz) s
5.2 µs 4.7 µs
160 samples 144 samples
Normal CP
Δf=15kHz
Special OFDM symbol: 66.7 µs OFDM symbol:

71.9 µS 2048 samples 71.3 µs
2208 samples 2192 samples
16.7 µs
512 samples
Extended CP
Δf = 15kHz
66.7 µs
OFDM symbol:
2048 samples 83.3 µs
2560 samples
Figure 2.7: Time slot
5
Spatial-domain is also considered for MIMO communications (1 grid per antenna)
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
normal-CP and extended-CP respectively . Figure 2.8 illustrates a RB for normal-
6
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 7/6
per subframe
(Normal/Extended)
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
6
Hence, a RB is formed by 84 or 72 resource elements.
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 R E F E R E N C E 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 . These 7
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.
Figure 2.9: Reference Symbols in a subframe
Additionally, there also exist the UE-specific reference signals (to be used for an explicit
7
UE) and MBSFN reference signals (for multi-cell broadcast).

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 subcarrier or 6 resource blocks 1.08 Mhz

0 1 2 3 4 56 0 1 2 3 4 5 6 0 1 2 3 4 0 1 2 3 4 5 6
Systembandwidth
O F D M symbol
Secondary Synchronization Signal
Other resource allocation

Primary Synchronization Signa
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).
• 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 Reference
Control region signaling symbols
(1-3 OFDM symbols)
Figure 2.11: L1/L2 control region
2.7 L I N K 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].
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

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).
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.
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 [6]: 8
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.
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.
• 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 /N gain, there is also a coding gain (Figure 2.16).
b 0
a) Chase Combining b) IR Combining

Figure 2.16: Soft combining
Chapter 3
VOICE-OVER-IP
V o I P (Voice-over-IP) is simply the transmission of voice traffic over IP-based

networks. T h e Internet Protocol (IP) w a s originally designed for data networking.
T h e success of IP in becoming a world standard for data networking has led to its
adaption to voice networking.
3.1 V O I P CODECS
G S M networks started with the Full rate (FR) speech codec a n d evolved to
E n h a n c e d Full R a t e (EFR). T h e Adaptive Multi-Rate ( A M R ) codec w a s added to
3 G P P Release 98 for G S M to enable codec rate adaptation to the radio conditions.
A M R data rates range f r o m 4.75 Kbps to 12.2 Kbps. T h e highest A M R rate is equal
to the E F R . A M R uses a sampling rate of 8 KHz, which provides 300-3400 Hz
audio bandwidth.
T h e A M R - W i d e b a n d ( A M R - W B ) codec w a s a d d e d to 3 G P P Release 5.
A M R - W B uses a sampling rate of 16 kHz, w h i c h provides 50-7000 Hz audio
bandwidth a n d substantially better voice quality a n d mean opinion score ( M O S ) . A s
the sampling rate of A M R - W B is double the sampling rate of A M R , A M R is often
referred to as A M R - N B (narrowband). A M R - W B data rates range from 6.6 Kbps to
23.85 Kbps. T h e typical rate is 12.65 Kbps, which is similar to the normal A M R of
12.2 Kbps. A M R - W B offers clearly better voice quality than A M R - N B with the s a m e
data rate a n d can be called w i d e b a n d audio with narrowband radio transmission.
T h e bandwidths for the A M R - N B a n d A M R - W B codecs are illustrated in

Figure 3.1.a, while a comparison of these codec with audio bandwidth is
exemplified in Figure 3.1.b. T h e smallest bit rates, 1.8 a n d 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 H z
Wideband A M R
50-7000 H z
Narrowband
AMR
300-3400 H z
a) Codec bandwidth b) Audio bandwidth

Figure 3 . 1 : V o I P Codecs: A M R - N B and A M R - W B
LTE specifications suggest that for V o I P performance tests, the A M R 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]. T h e
resulting capacity of the A M R - N B 1 2 . 2 Kbps w o u l d also be approximately valid for
A M R - W B 12.65 Kbps.
Table 3 . 1 : V o I P traffic model parameters
Parameter Value
Voice C o d e c RTP A M R 12.2, Source Rate 12.2 Kbps
Encoder frame 2 0 ms
Voice activity factor (VAF) 50% (a=b=0.01)
SID payload SID Packet every 160 ms during silence
15 bytes (5 Bytes + Header)
Protocol overhead with 10bits+padding (RTP-pre-header), 4 bytes (RTP/UDP/IP),
c o m p r e s s e d header 2 Bytes (RLC/Security), 16 bits (CRC)
Total voice payload on air interface 4 0 Bytes
There are two types of V o I P f r a m e s for the A M R 12.2 codec: Voice frames
and SID f r a m e s (see Figure 3.2):
• Voice Frames. At a voice source rate of 12.2 Kbps, a voice frame generated
every 2 0 ms consists of 244 bits. T h e total protocol overhead per voice
frame includes 10-bits of RTP pre-header and 2-bits padding resulting in a
total of 2 3 6 bits (32 bytes). Furthermore, a c o m p r e s s e d 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 R L C and security
header and 2 bytes C R C , the total V o I P payload size transmitted over the air
interface b e c o m e s 320 bits (40 bytes) every 20 ms [7].
• SID Frames. These frames contain comfort noise, and connection

information. SID f r a m e s are delivered during silent states every 160 ms (8
voice frames) consisting of 5 bytes of information and 10 bytes of header.
For the A M R 12.2 codec, 120 bits must be carried every 160 ms.
VoIP packets on the active period S I D packets on the silent period
20 ms
160 ms
Headers Payload
Figure 3.2: V o I P 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 V o I P packets in time. Instead, certain degree of missing
packets can be tolerated without notably affecting the Q o S perceived by the users.
For voice services, usually 1 % is tolerated [7].
T h e voice quality perception degrades as the end-to-end delay increases as

depicted in Figure 3.3 [10]. LTE a s s u m e s 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 e N o d e B to UE).
100
Users
Very Satisfied
90
Users
Satisfied
E-Model Rating R
80
Some Users
Dissatisfied
70
Many Users
Dissatisfied
60
Nearly All
Users
Dissatisfied
50
0 100 200 300 400 500
Mouth-to-Ear-Delay/ms
Figure 3.3: Voice Quality (Source: ITU)
T h e system capacity for V o I P is defined as the n u m b e r of users supported in

the cell w h e n more than 9 5 % of the users are satisfied. A V o I P user is satisfied if
9 8 % of its packets experience a delay of less than 5 0 ms [8].
3.3 V O I P TRAFFIC M O D E L
Consider t h e two-state voice activity model s h o w n in Figure 3.4. T h e

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 ) . T h e updates are made at the speech
encoder frame rate R= 1/T, w h e r e T is the encoder frame duration (20ms).
Silence Talking
(1 - a )
(State 0) (State 1) (1-b)
Figure 3.4: V o I P Traffic Model

T h e probabilities of being in state 0 and state 1 denoted as and

respectively:
b
P =
a + b
O
a
P= 1
a+b
T h e Voice Activity Factor is the probability of being in taking state, that is,
state 1:
a
VAF = P 1 =
a+b
T h e mean silence duration and mean talking duration in terms of n u m b e r of

voice f r a m e s can be written as:
1
1
E [TS] =
a
1
E [TS] =
b
T h e probabilities that silence duration or a talking duration is n voice f r a m e s

long are given by:
P Ts = a( 1 - a ) n - 1
, n = 1,2 , . . . (3.1)
(P =b(1-b) ,n=1,2,...)
Ts
n-1
(3.2)
Since the states transitions f r o m state 1 to state 0 and vice versa are
independent, the mean E [ T ] b e t w e e n active state entries is given simply by the
A E
s u m of the m e a n time in each state, That is:
1
A voice frame duration is 20 ms.
1 1
E [ T ] = E[TS] + E [ T ] =
AE T
a +b
Accordingly, the mean rate of arrival R AE of transitions into the active state is
given by
RA E = 1 / E[TA E ]
3.4 GENERATING V O I P TRAFFIC
The V o I P model traffic c a n serve as a guide on the number of resource

allocation requests. Likewise, it can also be used to generate V o I P traffic as will be
described in this section.
The number of packets in silence a n d talking states must be determined to

simulate V o I P traffic. T h e discrete inverse transform (DIT) w o u l d be used for such
purpose [12]. Consider the probability mass functions in Equation 3.1 a n d Equation
3.2 describing the probability of a user staying in talking a n d silence duration
respectively. For any probability mass function the following condition must hold:
P(N = n ) = p , i = 1 , . . . , Σ p
i i i = 1
i
T o generate N, the discrete random variable representing the number

packets in any state, generate a uniform random number u a n d set:
N = n if p + i ...+ p i - 1 +p i
That is,
n,
1 u ≤p 1
n,
2 p <u≤p +p 1 1 2
N = n,
3 Pi + P2 < u ≤pi+ p + p 3 2
n t p + ... + Pi- <u<p

1 1 1 + -p _ i 1 + pi
Because is uniform distributed on (0, 1), it follows that for ( 0 < a < b < 1 ) :
P(a< u ≤ b)=b-a
Consequently,
i-1 i
P Σ pj < u≤ Σ pj = Pi
j=1 j=1
w h i c h proves that N has the desired probability mass function.
Graphically, it represents a mapping between the C D F and the n u m b e r of

packets. For example, consider Equation 3.1 for the talking state case. T h e C D F is
easily obtained and plotted in Figure 3.5. Since u is a uniform n u m b e r between 0
and 1, it represents the probability that the duration in talking state is n packets.
Find the value n that produces u.Finallyset N=n.
Figure 3.5: Inverse Discrete Transform
3.5 V O I P TRAFFIC SIMULATOR
T h e V o I P traffic simulator w a s developed applying the V o I P traffic model

c o n c e p t s ; its respective parameters were initialized according to the 3GPP
recommendations in Table 3.1 [8]. Figure 3.4 depicts the theoretical and simulated
probabilities that a talking subframe is n frames long. Since the voice activity factor
is 5 0 % , the silence probability mass function will be exactly the s a m e . T h e obtained
results of the simulation are displayed in Table 2 . 1 . Simulation and theoretical
results do agree, proving the validity of the results
Figure 3.3: V o I P Traffic Model
Table 3.2: V o I P Traffic Simulation Results
Parameter Value
Voice frame periodicity 2 0 ms
SID frame periodicity 160 ms
Voice Activity Factor (VAF) 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
M e a n talking duration E [ T ]T 100 voice packets (2 sec)
M e a n silence duration E [ T ] S 100 voice packets (2 sec)
M e a n successive transitions into the 1 state E [ T A E ] 2 0 0 voice packets (4 sec)
M e a n 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 T H E JITTER C O N C E P T
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
of frame. The kth packet is expected to arrive at the receiver at a fixed time
t = kT + t , where t is the propagation time. However, as the kth packet travels
k p p
along the PS network it suffers delay variations t , called jitter. Then, the packet
n
would be received at a time t = kT +t + t.

k p
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 toJitter 32
compensated in a deterministic way. However, often exhibits a random behavior

[3].
Transmitter
T 2T 3T 4T
Packet 1 Packet 2 Packet 3 Packet 4
Time
Receiver
Packet 1 Packet 2 Packet 3 Packet 4
tp tp tp tp
T1 = 0 T2 T3 T4
tp - Propagation time
T - Packet periodicity
Ti - Jitter
Figure 4.1: Jittered packets
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 L T E JITTER SOURCES
4.2.1 SCHEDULER B U F F E R
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 H A R 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 , in the subframe n of the receiver. Then the UE will attempt to decode the
p
received signals during a time t , possible after soft combining. In the subframe n
UE
+ 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.
Figure 4.3: HARQ retransmission for the DL
4.2.3 RADIO L I N K C O N T R O L 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
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 O T H E R 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
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 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.
Chapter 5.VoIPJitter in LTE 39
3. Retransmission requests. Due to the wireless channels impairments,

packets could arrive corrupted to the receiver. In this case a retransmission
would be requested. 8 HARQ processes are used in LTE, implying a
retransmission delay of at least 8 subframes.
The Jitter T experienced by the kth packet, of the nth UE, is defined as the
n
k
time difference between the moment the packet was successfully received by the
UE, t (n)
UEk, and the instant when it arrived to the eNodeB, t (n)
eNBk . That is, T k
(n)
= t(n)
UEk -
t(n)
eNB k
.
The time transmission interval is not considered for jitter calculations since it
is a fixed delay. The propagation delay is neglected. Of course, the nth user will
receive K packets, comprising the user jitter profile. In addition, the cell jitter profile
can be obtained by taking the overall behavior of the N users in the cell. Cell jitter
profiles would be analyzed in further sections.
5.2 SIMULATION DESCRIPTION
The simulation can be split into 3 sections. VoIP traffic (see Chapter 3.5,
page 29), PHY layer, and MAC protocol (Scheduler).
5.2.1 L T E PHYSICAL LAYER
Developing a whole new simulator for the LTE air interface would be an
overwhelming task and beyond the scope of this thesis. However, results as real as
possible are desired. The Institute of Communications and Radio Frequency
Engineering of the Vienna University of Technology has developed the LTE link-
level simulator f o r the Matlab platform, and provided under academic use. It has
quickly gain popularity because of its features and reliability [9]. The structures of
the transmitter and receiver are shown in Figure 5.2 for the downlink.
Nevertheless, neither traffic models nor the required algorithms to study the
jitter phenomenon had been implemented, i.e., a continuous data stream was
supposed. The original simulator was enhanced, by the author of this thesis, to
support a variety of traffic models - VoIP in this case. The PHY layer simulation
parameters used in the simulations are described in Table 5.1.
Figure 5.2: LTE downlink PHY structure
Table 5.1: Simulation parameters

Parameter Value
Scenario/direction 1 Cell / Downlink
SNR 5, 8, 12
Channel Bandwidth 5 MHz
Number of RBs 25
Subcarrier Spacing 15 KHz
Cyclic Prefix Normal
Symbols per subframe 14
L1/L2 control region 3 OFDM symbols
Reference symbols per subframe 4
Duplexing FDD
HARQ mode Adaptive asynchronous
HARQ soft combining Off
HARQ processes 8
HARQ max retransmissions Infinite
CQI delay 1 ms
CQI resolution 1 RB
Possible MCSs MCS = CQI
Antenna configuration SISO
Channel model Typical Urban
Some comments about the simulation parameters:
 The SNRs are a sample of a worst, mean, and best case scenario.
 Although a semi-persistent scheduler will be used, 3 OFDM symbols are
considered for the simulation. The interest is to analyze the VoIP
performance under strict restrictions.
 Soft combining is not used to analyze a baseline system.
 The maximum number of retransmission was set to infinite, so that packet
losses did not impact in the jitter behavior.
 The MCS is adjusted according to the CQI recommendations. This will be
clarified in further sections.
5.2.2 LTE MAC PROTOCOL

The scheduler plays a key role in the performance of LTE networks.
However, the scheduling and rate-adaptation algorithms are vendor specific. In this
master thesis a fair user, delay-optimized resource allocation algorithm is
proposed. It is fair in the sense that every user has the same probability to be
chosen. It is delay optimized, signifying that decisions are taken based in time.
Note, that the term delay is used in this context. The reason for this is that if the
maximum jitter experienced by a packet is covered by the jitter buffer size, it will be
traded-off for delay. Thus, the goal is to keep the maximum number of packets with
minimum jitter.
The scheduler flow diagram is illustrated in Figure 5.3. Instructions are

performed sequentially. Every section of the scheduler is described in the following
subsections.
Data and Control Regions
The total number of physical resources depends on the bandwidth assigned

to the LTE system. For VoIP performance evaluations a 5 MHz channel bandwidth
is suggested, and results are scaled accordingly for other bandwidths [12]. This
corresponds to 25 RB-pairs. For each subframe, the scheduler must determine
how many of these radio resources will be employed for user data and how many
for signaling and control purposes. The number of data subcarriers varies between
subframes. However, an approximation can be obtained as follows:
 1 ms subframe size, 5 MHz BW

 Each subframe comprises of 14 OFDM symbols1 and 300 REs per symbol
 First 3 OFDM symbols comprises of control + pilot = 900 REs (worst case)
 5th, 8th and 12th symbol comprises additional pilots; 50 x 3 = 150 REs
 PSS and SSS 2 take 72 + 60 = 132 REs around the DC subcarrier;
subframes 0, 5
 Data subcarriers (RE) in DL: 300 x 14 – 900 – 150 – 132 = 3,018 (worst case)
 Number of overhead RE’s: 1182 REs (worst case)
Figure 5.4 illustrates the L1/L2 control region, reference symbols, and the
data region. Synchronization signals are omitted for clarity.
1
OFDM symbols
2
5th symbol contains 2 RS, hence 72 – 2 x 6 =60
Start The Scheduler makes decisions every

1 ms
Feedback Packets received Free

Input Yes
ACK/NACK correctly? HARQ Buffer
Retransmission
staus
Save
No Scheduler
Buffer
Read Any packet needs Free

# retransmissions < Free
Scheduler retransmission? Yes Yes Scheduler
Max. retransmissions? HARQ Buffer
Buffer Buffer
No
Priority 1
Read Free
Any other packet in the Time scheduler buffer Free
Scheduler Read Yes Yes Scheduler
scheduler buffer? > time-bound? HARQ Buffer
Buffer Buffer
No
Priority 2
Incoming
packets New packets?
Input Yes Priority 3
(VoIP Traffic) (VoIP Traffic)
Save
Read RBs
No Scheduler
Priority list available?
Buffer
Yes
Feedback
Input CQI > 1? No
CQI
Yes
Choose MCS Save

End
AMC HARQ Buffer
Figure 5.3: Scheduler flow diagram

OFDM symbol 0 1 2 3 4 5 6 7 8 9 10 11 12 13
RB = 1
R R
12 subcarrier R R
180 KHz
R R
R R
1 Slot 1 Slot
0.5 ms 0.5 ms
RB = 2
R R
12 subcarrier R R
180 KHz
25 RBs
R R
300 subcarriers
4.5 MHz
R R
L1/L2 Data
Control Region Region
...
RB = 25
R R
12 subcarrier R R
180 KHz
R R
R R
1 Subframe
TTI = 1 ms
Figure 5.4: L1/L2 Control Region, RS, and Data Region

Link Adaptation
Link adaptation is a functionality of the LTE scheduler. The MCS determines

both the modulation alphabet and the Effective Code Rate (ECR) of the channel
encoder. To help on the scheduler decisions, the UEs sends recommendations
about the maximum supported MCS that ensures aBLER ≤1 0 , the so-called
- 1
Channel Quality Indicator. The possible MCS suggested by a mobile terminal are
described in Table 5.2.
Table 5.2. CQI and MCS recommendations

CQI Index Modulation Coding Rate Efficiency
(Bits per resource element)
0 No transmission - -
1 QPSK 78/1024 0.1523
2 QPSK 120/1024 0.2344
3 QPSK 193/1024 0.3770
4 QPSK 308/1024 0.6016
5 QPSK 449/1024 0.8770
6 QPSK 602/1024 1.1758
7 16QAM 378/1024 1.4766
8 16QAM 490/1024 1.9141
9 16QAM 616/1024 2.4063
10 64QAM 466/1024 2.7305
11 64QAM 567/1024 3.3223
12 64QAM 666/1024 3.9023
13 64QAM 772/1024 4.5234
14 64QAM 873/1024 5.1152
15 64QAM 948/1024 5.5547
To obtain the BLER for the MCS corresponding to each CQI value, AWGN
simulations were performed. Figure 5.5 shows the BLER results of CQIs 1-15
without using HARQ. Each curve is spaced approximately 2 dB from each other.
The SNR-to-CQI mapping required to achieve this goal can thus be obtained by
plotting the 10% BLER values of the curves in Figure 5.5 over SNR, like it is shown
in Figure 5.6. Using the obtained line, an effective SNR can be mapped to a CQI
value [14].
BLER, 1.4MHz, SISO AWGN, 5000 subframes

0
10
-1
10
BLER
-2
10
-3
10
-10 -5 0 5 10 15 20
SNR [dB]
Figure 5.5: BLER curves obtained from SISO AWGN simulations for all 15 CQI
values. From CQI 1 (leftmost) to CQI 15 (rightmost)
SNR-CQI mapping model

15
14
13
12
11
10
8
CQI
0
-10 -5 0 5 10 15 20
SNR [dB]
Figure 5.6: SNR-CQI mapping
For VoIP traffic the used BLER target in DL is in the order 10% for the first
transmission [10]. Since CQI provides approximately such BLER, the MCS can be
adjusted according to the CQI suggestions.
Priority System
Before actual radio resource assignment takes place, a priority system

orders the packets according to the following criteria
1. Retransmission requests. Since retransmissions take at least 8 ms to be

completed, it gets the higher priority.
2. Buffer time. This weight is obtained as the sum of the time in the queue
experienced by all the packets of the same user.
3. New packets. Incoming VoIP/SID packets get the lower priority.
Note that the priority system applies for packets not for users. That is, all the
users are treated equally, but packets experiencing more delay jitter get higher
priority. It is important to mention that, if certain UE is experiencing bad
instantaneous channel conditions (CQI < 2), it will not be scheduled until the next
subframe even if it has the higher priority. Instead, the next user in the priority list
will be chosen.
Resource assignment
It will be assumed a semi-persistent scheduling implementation. That is,

users will always transmit in the same frequency allocation, but scheduling
decisions can be taken in the time domain.
The number of users in each simulation has been chosen as a multiple of

25, since there 25 RBs in 5 MHZ. Thus for 25, 50 and 75 UEs there would be 1, 2,
and 3 users per RB respectively.
5.3 SIMULATION RESULTS
Simulations are computationally intensive because of the complexity of

physical and MAC layers simulations. Coding, decoding, modulation, channel
estimation, link adaptation, scheduling, and other functions are performed
continuously for each user, packet and fragment. The tests were run for 12, 000
subframes (12 seconds), for different SNR scenarios. Before analyzing the results,
it is important to describe some concepts. The cell outage probability is defined as.
P{outage} = P(T > 50)

Where T is the jitter experienced by all the packets in the cell. Likewise, LTE
specifies the capacity and satisfaction criteria as [14].
"System capacity 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 ms"
Note that satisfaction is a stricter criterion than cell outage probability, since
it is based in the number of satisfied users and not only in the total jitter
experienced in the cell.
5.3.1 SNR = 5 dB
Consider the simulation results in Table 5.3 and Figure 5.7, describing the
jitter behavior for a SNR = 5 dB. It can be concluded the following. The satisfaction
quality criterion is completely accomplished for 25 and 50 users, although 0.22 %
and 0.34% of the packets are above the desired bound (50 ms). However, when
the number of mobile terminals increases to 75, the satisfaction criterion is not
strictly achieved (93.3% < 95%). Nevertheless, jitter is a minor issue in this case,
since the mean of the delay is 12.2 and the deviation is only 10.2. This indicates
that jitter could be tolerated for the 0.61% of the packets.
Now, let us study the case for 100 users. At a glance it could seem from
Figure 5.7 as if jitter were under acceptable limits since the mean is 18.9 ms, and
the standard deviation is 27.6 ms. However, the satisfaction percent drops
dramatically to 66%; indeed, the outage probability is 5.8%. This is not acceptable.
If users keep increasing up to 125 and 150, the means are 129.5 and 171.4, clearly
above the 50 ms target. Finally, for 175 UEs the satisfaction percent is only 3.4,
and the outage probability is 69%.
Now, let us analyze the jitter cell profile shown in Figure 5.10. Consider the
3
case for 25 users. It can be observed, that jitter is concentrated inside the 50 ms
bound, so 100% capacity is achieved. Most of the packets (10.5%) suffer a jitter of
2 ms due to packet fragmentation (the source is clear, since packets does not have
to wait because there is not contention). However, as expected, the retransmission
scheme will introduce jitter. Recall that a packet sent in subframe n, takes n + 8
subframes to be retransmitted. Thus, erroneous packets generate "replicas" with a
3
Jitter cell profiles show the probability that a packet is jittered by T milliseconds
ms
periodicity of 8 ms. As the number of users keeps increasing, the jitter cell profile
spreads, leading to the subsequent satisfaction drop.
Table 5.3: Cell jitter; SNR = 5 dB

Users Mean Deviation Mode Max Poutage Satisfaction
25 10.2 8.9 2 72 2.29 x 10-3 100
50 11.4 9.4 2 75 3.46 x 10-3 100
-3
75 12.2 10.2 2 97 6.16 x 10 93.3
-2
100 18.9 27.6 2 479 5.81 x 10 66
-1
125 57.2 87.1 2 838 2.72 x 10 23.2
150 171.4 240.5 2 2450 5.20 x 10-1 4.7
175 292.7 379.5 2 4815 6.90 x 10-1 3.4
Jitter behavior
700
650 Mean
Dispersion
600
550
500
450
400
Jitter (ms)
350
300
250
200
150
100
50
0
25 50 75 100 125 150 175
Users
Figure 5.7: Jitter behavior (SNR = 5 dB)

5.3.2 SNR = 8 dB
Now, let us study the case for the average case (SNR = 8 dB). Consider
Table 5.4 and Figure 5.8. The jitter behavior seems steady up to 100 users, with a
satisfaction criterion of 100%. Note that under the strict capacity criterion, capacity
has increased from 50 to 100 compared to the previous case (SNR = 5 dB).
Although the satisfaction is 88% for 125 UEs, the outage probability is only
1%. At 150 users, 81.3% satisfaction is achieved with an outage probability of
3.69%. Operation for 175 and 200 users is not suggested since the outage
probability is 22.6 % and 45.1 % respectively.

25 7.1 7.4 1 60 4.07 x 10-4 100
-3
50 7.6 7.9 1 68 1.26 x 10 100
-3
75 7.8 8.1 1 83 1.54 x 10 100
-3
100 8.1 8.2 1 86 1.18 x 10 100
125 9.5 10.5 1 110 1.07 x 10-2 88.8
150 15.8 36.2 1 734 3.69 x 10-2 81.3
175 49.4 97 1 1093 2.26 x 10-1 40
200 147.2 233.3 1 2003 4.51 x 10-1 14.5
It can be noted from Figure 5.8 that network congestion has a small impact
on jitter up to 125 users. In this zone, jitter is mainly due to HARQ retransmission
(see the jitter profiles in Figure 5.8). Then, jitter will increase exponentially with the
number of users.
Jitter cell profiles for this scenario are illustrated in Figure 5.11. Under no
contention (25 users), 24% of the packets are jittered by 1 ms; an excellent
performance measure. Nevertheless, network congestion will cause that the jitter
cell profile spreads, as the previous case, leading to the subsequent reduction in
satisfaction.
Jitter behavior
400
Mean
350 Dispersion
300
250
Jitter (ms)
200
150
100
50
0
25 50 75 100 125 150 175 200
Users
5.3.3 SNR = 12 dB
As can be noted from Table 5.5 and Figure 5.9, the capacity for this case is
225 users. Network congestion has small impact up to this case. Then, when the
number of UEs in cell reaches 250, the satisfaction criterion drops to 78.4% and
the cell outage probability is 2.8%.
From the jitter cell profile depicted in Figure 5.12, it is easy to see that from
25 to 100 users the packets are concentrated around 0 and 1 ms. Due to
retransmissions, with a periodicity of 8 ms, “replicas” appears along the jitter
profile.
As the number of users increases, then packets will be placed in the

scheduler buffer. This will lead to the spreading of the jitter cell profile, and the
subsequent reduction in the satisfaction quality criterion.

25 4.2 6 0 43 - 100
-5
50 4.3 6.1 0 51 3.69 x 10 100
-4
75 4.8 6.5 1 58 1.71 x 10 100
100 4.8 6.5 1 58 2.07 x 10-4 100
125 5 6.6 1 59 2.58 x 10-4 100
150 5.1 6.6 1 60 2.74 x 10-4 100
175 5.3 6.6 1 61 2.76 x 10-4 100
200 5.9 7.1 1 66 3.79 x 10-4 100
225 6.5 7.4 1 73 5.71 x 10-4 100
250 10 18.9 1 365 2.80 x 10-2 78.4
275 32.1 75.6 1 1252 1.4 x 10-1 46.9
-1
300 58.3 121.7 1 1498 2.80 x 10 55.3
Jitter behavior
175
Mean
Dispersion
150
125
100
Jitter (ms)
75
50
25
0
25 50 75 100 125 150 175 200 225 250 275 300
Users

Chapter 5.VoIP Jitter in LTE
SNR = 5 dB SNR = 5 dB
0.12 0.12
0.1 0.1
0.08 0.08
Probability
Probability
0.06 0.06
0.04 0.04
0.02 0.02
0 0
0 10 20 30 40 50 60 70 0 10 20 30 40 50 60 70
Jitter (ms) Jitter (ms)
a) 25 UEs b) 50 UEs
0.12 0.12
0.1 0.1
0.08 0.08
Probability
Probability
0.06 0.06
0.04 0.04
0.02 0.02
0 0
0 10 20 30 40 50 60 70 80 90 0 50 100 150 200 250 300 350 400 450
c) 75 UEs d) 100 UEs

Figure 5.10: Jitter cell profile (SNR = 5 dB)
53
0.25 0.25
0.2 0.2
0.15 0.15
Probability
Probability
0.1 0.1
0.05 0.05
0 0
0 10 20 30 40 50 60 0 10 20 30 40 50 60
a) 25 UEs b) 50 UEs
0.25 0.25
0.2 0.2
0.15 0.15
Probability
Probability
0.1 0.1
0.05 0.05
0 0
0 10 20 30 40 50 60 70 80 0 10 20 30 40 50 60 70 80
54
0.25 0.25
0.2 0.2
0.15 0.15
Probability
Probability
0.1 0.1
0.05 0.05
0 0
0 20 40 60 80 100 0 100 200 300 400 500 600 700
e) 125 UEs f) 150 UEs

55
0.35 0.35
0.3 0.3
0.25 0.25
0.2 0.2
Probability
Probability
0.15 0.15
0.1 0.1
0.05 0.05
0 0
0 5 10 15 20 25 30 35 40 0 10 20 30 40 50
a) 25 UEs b) 50 UEs
0.35 0.35
0.3 0.3
0.25 0.25
0.2 0.2
Probability
Probability
0.15 0.15
0.1 0.1
0.05 0.05
0 0
0 10 20 30 40 50 0 10 20 30 40 50
56
0.35 0.35
0.3 0.3
0.25 0.25
0.2 0.2
Probability
Probability
0.15 0.15
0.1 0.1
0.05 0.05
0 0
0 10 20 30 40 50 60 0 10 20 30 40 50 60
e) 125 UEs f) 150 UEs

0.35 0.35
0.3 0.3
0.25 0.25
0.2 0.2
Probability
Probability
0.15 0.15
0.1 0.1
0.05 0.05
0 0
0 10 20 30 40 50 60 0 10 20 30 40 50 60
g) 175 UEs h) 200 UEs
57
0.35 0.35
0.3 0.3
0.25 0.25
0.2 0.2
Probability
Probability
0.15 0.15
0.1 0.1
0.05 0.05
0 0
0 10 20 30 40 50 60 70 0 50 100 150 200 250 300 350
i) 225 UEs j) 250 UEs

0.06 0.06
0.05 0.05
0.04 0.04
Probability
Probability
0.03 0.03
0.02 0.02
0.01 0.01
0 0
0 200 400 600 800 1000 1200 0 200 400 600 800 1000 1200 1400
k) 275 UEs l) 300 UEs

58
Chapter 6
CONCLUSIONS AND FUTURE WORK
6.1 GENERAL CONCLUSIONS

3GPP Long Term Evolution is the next step towards 4G mobile networks,
promising higher data rates, increased capacity, and reduced delay. LTE is a flat,
all-IP network, where voice services will be provided through Voice-over-IP (VoIP).
VoIP services are highly sensitive to jitter, the delay variation, since it
causes speech intelligibility disruptions. The main jitter sources in LTE were
identified. Network congestion, packet fragmentation and HARQ retransmissions
influence the arrival variability of voice packets. Furthermore, the radio channel
conditions dictate the performance of all of these components as demonstrated in
the VoIP traffic, PHY and MAC layer simulations.
The scheduler plays a key role in the queue dynamics; therefore jitter.
Scheduler strategies should consider that jitter does not exceed the quality bound,
(50 ms for LTE), such that the jitter buffer can handle effectively this phenomenon.
Indeed, for VoIP service is crucial to assure a minimum data rate (according to the
requirements of the VoIP codec) at any time.
Three different scenarios were analyzed corresponding to diverse channel

conditions. In every scenario the MCS was selected to achieve a 10% BLER, as
suggested by the CQI reports sent by the UE. In this way a reference point
between each scenario was established.
In poor radio conditions (SNR = 5 dB), low-order modulation and coding

schemes induced packet fragmentation. This phenomenon increased network
congestion since several TTIs were required to send a single packet. Likewise,
every fragment could be retransmitted several times provoking the exponential
59
Chapter 6. Conclusions and Future Work 60
growth of jitter. According to the 3GPP VoIP satisfaction criterion, up to 50 users

could be supported in this case.
For average (SNR = 8 dB) and good (SNR = 12 dB) channel conditions,
packet fragmentation was reduced since higher order MCSs were used. The main
jitter source for few users in cell was due HARQ retransmissions. However, as the
number of mobile terminals increased, packets began to be placed in the scheduler
buffer, leading to network congestion. The number of supported users was
determined as 100 and 225 respectively.
Comparing the results of the different scenarios the following conclusions

can be established. Network congestion leads to the spreading of the jitter cell
profiles (see Figure 5.10, Figure 5.11 and Figure 5.12), while HARQ
retransmissions generate “replicas” of such widening with a periodicity of 8 ms.
Indeed packet fragmentation, inducing a jitter granularity of 1 ms, is

preferable over HARQ retransmissions that take at least 8 ms to be accomplished.
Even more, it can be inferred that conservative MCS could improve the jitter
behavior even in the worst case scenario. It comes from the fact that, although the
number of fragments would increase, along the jitter spreading, the BLER and the
number of HARQ retransmissions would be decreased. Then, the jitter replicas
would be diminished. Of course, this implies that the jitter caused by the added
fragments does not surpass the delay bound.
Under the strict settings considered in this thesis (maximum overhead,

simple HARQ retransmissions, time-based scheduling), LTE prove to be able to
handle jitter. However, aspects such as network congestion have to be considered
in the deployment stage. For instance, the cell size determines the number of
users, and radio channel conditions; key component in jitter dynamics.
6.2 FUTURE WORK

The following ideas can suggested for further research:
 Analyze jitter under conservative MCSs.

 Study the case for HARQ soft combing.
 Consider the effects of dynamic scheduling.
Appendix A
MULTI-CARRIER TRANSMISSION
In modern communications systems intensive multimedia applications such

as video streaming, videoconferencing, voice, etc., are being d e m a n d e d by users.
T h e necessity for higher data rates in a limited spectrum has developed a series of
transmission s c h e m e s w h i c h improve the spectrum efficiency, increment capacity,
and reduce the corruption c a u s e d by radio-channel time dispersion. O n e of such
s c h e m e s is the so-called, Orthogonal Frequency Division Multiplexing ( O F D M ) , a
multi-carrier system.
A.1 T H E MUTI-CARRIER C O N C E P T
A.1.1 CHANNEL CAPACITY
S h a n n o n provided the theoretical tools to determinate the m a x i m u m rate C,

k n o w n as channel capacity, w h e r e the radio link is impaired only by additive white
G a u s s i a n noise. This relationship is given b y :
S
C = B W ·1og ( l + 2
N)
BW is the available bandwidth, S is the signal power a n d N d e n o t e s the

power noise. It is clear that c h a n n e l capacity d e p e n d s on the signal-to-noise ratio
a n d the bandwidth available. S u p p o s e a c o m m u n i c a t i o n s y s t e m with rate R. T h e
signal power c a n be e x p r e s s e d a s S = E · R w h e r e E is the energy per information
b b
bit, a n d N = N · BW is the power spectral density in W / H z . S i n c e the

0 actual
information rate cannot e x c e e d the capacity of the s y s t e m :
S E -R
≤C = B W · lo g ( l + · log 2 (
b
R 2 ) = BW 1 +
N N -BW )
0
61
Appendix A. Multi-carrier transmission 62
Let γ = R/BW be defined a s the bandwidth utilization of the c h a n n e l .
E b
)
γ ≤log (1 + γ
N
2
0
It is convenient to e x p r e s s this inequality a s a lower b o u n d on the required

r e c e i v e d energy per information bit, n o r m a l i z e d to the noise power density,
n e c e s s a r y for a given data rate.
E -1
{E } 2γ
=
b b
≥min
N 0 N 0 γ
T h e lower bound is plotted in Figure A . 1 . T w o regions c a n be identified from

this plot: a power-limited region a n d a bandwidth limited region.
Figure A . 1 : Operation regions
Power-limited region. W h e n the bandwidth utilization is less than 1, the

channel is underutilized; an increment in the bandwidth of the
c o m m u n i c a t i o n s y s t e m will not improve significantly the performance of the
c h a n n e l . T h e actual constrain is the power of t r a n s m i s s i o n .
Bandwidth-limited region. If the bandwidth is not increased, a very high

power is required to a c h i e v e higher data rates.
Working on a bandwidth-limited region is highly inefficient since un¬

proportional signal-to-noise ratios are required to a c h i e v e high data rates. T h e
s y s t e m s h o u l d be operated in the power-limited region. T h i s is the reason w h y
wider bandwidths are n e c e s s a r y for the next generation of w i r e l e s s c o m m u n i c a t i o n
networks s u c h a s 3 G P P L o n g T e r m Evolution. L T E c a n handle bandwidths up to
20 MHz.
A.1.2 WIDER BANDWIDTHS
Larger bandwidths are n e c e s s a r y to provide high data rates in a power

efficiently way. However, transmitting a single wideband signal has some
constraints.
• Allocating continuous spectrum is c o m p l i c a t e d s i n c e it is a s c a r c e and

e x p e n s i v e resource.
• T h e use of w i d e r bandwidths i n c r e a s e s the complexity of transmitters a n d

receivers. F o r e x a m p l e , the amplifiers must work on a w i d e r linear area.
Furthermore, the time dispersion problem associated with wider-band

transmission arises naturally. T i m e dispersion o c c u r s w h e n the transmitted signal
propagates to the receiver via multiple paths with different delays. In a frequency
domain, a time-dispersive channel corresponds to a non-constant channel
frequency response. T h i s radio-channel frequency selectivity will corrupt the
frequency d o m a i n spectrum of the transmitted signal a n d lead to higher error rates
for a given signal-to-noise/interference ratios. Every radio c h a n n e l is subject to
frequency selectivity, at least at s o m e extend. However, the extent to w h i c h the
frequency selectivity impacts the radio c o m m u n i c a t i o n d e p e n d s of the bandwidth of
the transmitted signal with, in general, larger impact for w i d e - b a n d transmissions.
The amount of radio-channel frequency selectivity also depends on the
environment with typically less frequency selectivity (less time dispersion) in c a s e
of small cells a n d in environments with few obstructions a n d potential reflectors
s u c h a s rural environments.
A s a mobile terminal is moving through the environment, the detailed

structured of the multi-path propagation, a n d thus also the detailed structure of the
c h a n n e l frequency response, may vary rapidly in time. T h e rate of the variations in
the c h a n n e l frequency r e s p o n s e is related to the c h a n n e l D o p p l e r s p r e a d .
A.2 MULTI-CARRIER TRANSMISSION
In muti-carrier t r a n s m i s s i o n s instead of transmitting a single w i d e b a n d

signal, multiple narrowband signals, often referred a s subcarriers, are frequency
multiplexed a n d transmitted over the s a m e radio-link. T h e impact in terms of signal
corruption due to radio-channel frequency selectivity d e p e n d s on the bandwidth of
e a c h subcarrier.
C o n s i d e r a w i d e b a n d s y s t e m with a total transmission bandwidth BW a n d a

data rate R. T h e c o h e r e n c e time of the c h a n n e l is a s s u m e d to be B < B, s o the
c
signal experiences selective fading. T h e basic principle behind multi-carrier

transmission is to split the s y s t e m into N s u b s y s t e m s p l a c e d in parallel. E v e r y sub-
stream will h a v e a bandwidth B = BW / N a n d a data rate R
N N = R/N. If is c h o s e n
sufficiently large then B << B a n d every subcarrier will e x p e r i e n c e approximately
N c
flat fading. In the time domain this implies that the symbol time of every subcarrier
T = 1/ B is much bigger than the delay spread in the channel σ = 1 /Bc so little
N N τ
ISI corruption is e x p e r i e n c e d .
A d r a w b a c k of conventional multi-carrier transmission is that the spectrum of

e a c h subcarrier cannot overlap to a v o i d distortion. Typically frequency guard b a n d s
are a d d e d between subcarriers, resulting in low frequency spectrum utilization. A
second disadvantage is that multi-carrier s y s t e m s usually have big peak-to-
a v e r a g e power ratios ( P A P R ) , w h i c h is an a s p e c t to c o n s i d e r in mobile terminals.
A.3 O F D M AS A MULTI-CARRIER TRANSMISSION
Orthogonal Frequency Division Multiplexing is a kind of multi-carrier

transmission. T h e main differences between O F D M a n d a straightforward multi-
carrier transmission s c h e m e are.
• T h e use of relative large n u m b e r of narrowband subcarriers. A s a n e x a m p l e ,

W C D M A multi-carrier evolution consists of four subcarriers in a bandwidth of
2 0 M H z , e a c h with a bandwidth of 5 M H z . In contrast, O F D M could consist
of h u n d r e d s of subcarriers transmitted at the s a m e time over the s a m e radio-
channel.
• Tight frequency-domain p a c k i n g with a subcarrier s p a c i n g Δf = 1/T w h e r e u
T u is the per-carrier modulation-symbol time (Figure A . 2 ) . Basically, the

spectrum overlaps without causing interference d u e to the orthogonally

property between subcarriers as s h o w n in Figure A . 3 .
Figure A . 2 : Subcarrier spacing
Figure A.3: a) F D M , b) O F D M
A n O F D M signal is p r o d u c e d through a bank of Nc modulators, w h e r e e a c h

modulator c o r r e s p o n d s to a n O F D M subcarrier, like the o n e s h o w n in Figure A . 4 . a .
Therefore, the O F D M signal x(t), during the time mT u ≤t<(m+1)T , U c a n be
e x p r e s s e d as:
N -1
c
k=0
a) Bank of Modulators b) Bank of correlators

Figure A . 4 : O F D M modulation and demodulation
.
W h e r e x (t) k is the kth modulated subcarrier with frequency f =k k Δf a n d
(m)
ak is the modulation s y m b o l applied to the kth subcarrier during the mth O F D M
s y m b o l interval. Note that the serial-to-parallel operation "enlarges" the modulation
s y m b o l time by a factor N . T h e modulation s y m b o l s could be from a n y alphabet
c
s u c h a s Q P S K , 1 6 Q A M or 6 4 Q A M . T w o modulated subcarriers x ( t ) a n d x ( t ) k 1 k 2
are orthogonal with e a c h other in the interval mT U ≤t<(m+ 1)T

u if:
(m+1)T u
(m+1)T u
( )
Xk (t)x* t dt
1 k2 = a k 1 a* k 2 e j 2 π k Δft j2πk
1 e- 2 dt = 0 Δft
k 1 ≠k2
mT u mT u
Demodulation of an O F D M signal is carried out through a bank of correlators

(Figure A . 4 b), o n e for e a c h subcarrier. Although the spectrum between subcarriers
overlaps there is not interference between the O F D M subcarriers.
A.3.1 OFDM IMPLEMENTATION USING IDFT/DFT

As the number of subcarriers increases, the bank of modulators a n d
correlators b e c o m e a serious issue. H o w e v e r , O F D M c a n be implemented digitally
through the I D F T / D F T transforms. S u p p o s e an O F D M signal is s a m p l e d at uniform
intervals .
Where:
a,
k 0 ≤k < N c
a'k= { a, N c ≤k < N
T h i s c o r r e s p o n d s to the IDFT of the O F D M signal. B y selecting the IDFT

s i z e N a s a power of 2, the O F D M modulation c a n be implemented through the
efficient radix-2 IFFT (Figure A . 5 a). Similarly, the bank of N correlators in the c
receiver c a n be replaced by a D F T block a s illustrated in Figure A . 5 b).

a) IDFT implementation (TX) b) DFT implementation (RX)

Figure A . 5 : Digital implementation of O F D M
S i n c e the subcarriers are orthogonal there is not inter-carrier interference

(ICI). In Figure A . 6 , the receiver s a m p l e s at the p e a k of the kth subcarrier w h i c h is
the z e r o e d value for all the other subcarriers.
Figure A . 6 : T h e r e is no intra-cell interference for O F D M
A.3.2 CYCLIC-PREFIX
A n uncorrupted O F D M signal c a n be d e m o d u l a t e d without any interference

between subcarriers. However, a time dispersive c h a n n e l will provoke a partial loss
of orthogonally. T h e time integration interval for o n e interval s y m b o l will overlap
with the s y m b o l boundary of another path, a s illustrated in Figure A . 7 . In this c a s e
there will be not only inter-symbol interference within a subcarrier but also
interference between subcarriers.
Direct path
Reflected path
Integration interval for

demodulation of direct path
Figure A . 7 : Corruption due to time dispersion
A time guard band, T C P , is u s e d in O F D M transmission to deal with this

problem, the s o - c a l l e d cyclic-prefix. T h e value of the cyclic prefix s h o u l d c o v e r the
m a x i m u m length of the time dispersion e x p e c t e d in the radio-link. Basically, the C P
is a c o p y of the last part of the O F D M s y m b o l a p p e n d e d at the beginning. T h e
cyclic prefix m a k e s a n d O F D M insensitive to time dispersion as long the C P is
larger than the delay s p r e a d present in the c h a n n e l . However, a lost in capacity is
implied, s i n c e the s y m b o l durations T + T u CP (see Figure A . 8 ) , a n d T CP is not
information u s e d by the receiver (it is discarded).
Figure A . 8 : Cyclic-prefix insertion
A.3.3 OFDM SUBCARRIER SPACING

T h e subcarrier s p a c i n g is o n e of significant factors which dictate the
performance of the O F D M s y s t e m . W h a t subcarrier s p a c i n g to use d e p e n d s of
what type of environments the s y s t e m is to operate in, including the m a x i m u m
radio-channel frequency selectivity ( m a x i m u m e x p e c t e d time dispersion) a n d the
m a x i m u m rate of c h a n n e l variation ( m a x i m u m e x p e c t e d D o p p l e r spread). Basically:
• T h e subcarrier s p a c i n g s h o u l d be a s small a s p o s s i b l e (T u a s large as

possible) to minimize the cyclic-prefix o v e r h e a d T CP = T /(T
CP CP + T ).
u
• A small subcarrier spacing increases the sensitivity to Doppler spread
In LTE the subcarrier spacing has been defined as Δf = 15 KHz.
A.3.4 NUMBER OF SUBCARRIERS
O n c e the subcarrier s p a c i n g has b e e n s e l e c t e d b a s e d on environment,

e x p e c t e d D o p p l e r s p r e a d a d time dispersion, etc., the n u m b e r of subcarriers c a n
be determinate b a s e d on the amount of spectrum available a n d the a c c e p t a b l e out-
of-band e m i s s i o n s . T h e bandwidth of a n O F D M signal e q u a l s BW = Nc . Δf, i.e. the
n u m b e r of subcarriers multiplied by the subcarrier s p a c i n g .
NOMENCLATURE
3GPP Third Generation Partnership Project
ACK Acknowledgment (in ARQ protocol)
AM Acknowledged Mode (RLC configuration)
AMC Adaptive Modulation and Coding
AMR-NB Adaptive MultiRate Narrowband
AMR-WB Adaptive MultiRate Wideband
ARQ Automatic Repeat-reQuest
AWGN Additive White Gaussian Noise
BER Bit Error Rate
BLER Block Error Rate
BW Bandwidth
CDF Cumulative Density Function
CP Cyclic Prefix
CQI Channel Quality Indicator
CS Circuit-switched
CTC Convolutional Turbo Code
DFT Discrete Fourier Transform
DL Downlink
EFR Enhanced Full Rate (VoIP Codec)
eNB eNodeB (LTE base station)
eNodeB E-UTRA Node B
EPC Evolved Packet Core
E-UTRA Evolved UTRA
E-UTRAN Evolved UTRAN
FDD Frequency Division Duplex
FDM Frequency-Division Multiplex
FDMA Frequency Division Multiple Access
FEC Forward Error Correction
FFT Fast Fourier Transform
70
Nomenclature 71
FR Full Rate (VoIP Codec)

GSM Global System for Mobile Communications
HARQ Hybrid-ARQ
HSPA High Speed Packet Access
IDFT Inverse DFT
IEEE Institute of Electrical and Electronic Engineers
IFFT Inverse Fast Fourier Transform
IP Internet Protocol
IR Incremental Redundancy
ITU International Telecommunications Union
LTE Long Term Evolution
MAC Medium Access Control
MBMS Multimedia Broadcast/Multicast Service
MCS Modulation and Coding Scheme
MME Mobility Management Entities
MOS Mean Opinion Score
NACK Negative ACK (in ARQ protocols)
OFDM Orthogonal Frequency-Division Multiplexing
OFDMA Orthogonal Frequency-Division Multiple Access
PAPR Peak Average to Power Ratio
PCFICH Physical Control Format Indicator Channel
PDCCH Physical Downlink Control Chanel
PDSCH Physical Downlink Shared Channel
PHICH Physical Hybrid-ARQ Indicator Channel
PHY Physical layer
PMF Probability Mass Function
PS Packet-Switched
PSS Primary Synchronization Signal
PSTN Public Switched Telephonic Network
QAM Quadrature Amplitude Modulation
QPSK Quadrature Phase-Shift Keying
Nomenclature 72
RAN Radio Access Network

RB Resource Block
RE Resource Element
RI Rank Indicator
RLC Radio Link Control
RRC Radio Resource Control
RS Reference Symbol
RTP Real Time Protocol
RTT Round-Trip delay Time
RX Receiver
S1 The interface between eNodeB and the EPC
SC-FDMA Single-Carrier FDMA
S-GW Serving Gateway
SID Silence Insertion Descriptor
SNR Signal-to-Noise Ratio
SSS Secondary Synchronization Signals
TDD Time Division Duplex
TDM Time Division Multiplexing
TDMA Time Division Multiple Access
TM Transparent Mode (RLC Configuration)
TTI Transmission Time Interval
TX Transmitter
UE User Equipment (LTE mobile terminal)
UL Uplink
UM Unacknowledged Mode (RLC Configuration)
UMTS Universal Mobile Telecommunication System
UTRA Universal Terrestrial Radio Access
UTRAN Universal Terrestrial Radio Access Network
VoIP Voice-over-IP
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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

Downloaded 27/11/2018 07:43:58

Link to Item http://hdl.handle.net/11285/569657

PROGRAMA DE GRADUADOS EN MECATRÓNICA

Christian Alberto Rodríguez García

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

Master of Science in Electronic Engineering

Monterrey, N.L. December 2009

División de Mecatrónica y Tecnologías de Información

Cristina García González, Juan José Rodríguez Uc

I would like to express my gratitude to my thesis advisor David Muñoz

Christian Alberto Rodríguez García

Christian Alberto Rodríguez García, B.S.

3GPP LTE is the next step towards 4G mobile communications with

Jitter is a common issue in packet-switched networks, where packets arrive

Figure 2.1: LTE Architecture ......................................................................... 8

Table 2.1: LTE Frequency bands ................................................................ 10

LTE is extremely flexible, using a number of defined channel bandwidths

1.1 PROBLEM DESCRIPTION

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

Jitter is a common issue in packet-switched networks, where packets arrive

1.5 THESIS ORGANIZATION

Final conclusions are presented in Chapter 6. Further research under the

3GPP Long Term Evolution is the next step towards 4G mobile

2.1 T H E STANDARDIZATION PROCESS

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

• Standardization starts with the requirement phase, where the

2.2 DESIGN TARGETS

1. Support scalable bandwidths

2. Peak data rate that scales with system bandwidth.

d) UL cell edge : 0.02-0.03 bit/s/Hz (2 to 3 x HSPA Rel. 6 )

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

Additionally, the related VoIP service requirements are:

2. VoIP should be supported with at least as good radio backhaul efficiency

3. Voice and other real-time services supported in the CS domain in Release 6

S1 interface which connects the eNodeBs to the Mobility Management Entities

• eNB: Enhanced Node B, or base

• UE: User Equipment

• EPC: Evolved Packet Core

o MME: Mobility Management

E-UTRAN o SAE: System Architecture

• E-UTRAN: Evolved Universal

Figure 2.1: LTE Architecture

The radio protocol architecture of E-UTRAN is specified for the control-plane

• Packet Data Convergence Protocol (PDCP) performs IP header

• Radio Link Control (RLC) is responsible for segmentation/concatenation,

• Medium Access Control (MAC) handles the HARQ retransmissions,

• Physical Layer (PHY) is responsible for coding/decoding,

eNodeB Mobile terminal (UE)

Figure 2.2: DL LTE protocols

2.4 L T E PHYSICAL LAYER

2.4.1 BANDWIDTHS, FREQUENCY BANDS A N D DUPLEXING

different, sufficiently separated, frequency bands. Time Division Duplex (TDD)

Figure 2.3: FDD and TDD

Table 2.1: LTE Frequency bands

17 704 - 716 734 - 746 FDD -

33 1900 - 1920 1900 - 1920 TDD Europe, Asia (not Japan)

LTE can be deployed in current cellular frequency bands (IMT-2000) and

of an OFDM signal is given by the relationship BW = Nc. Δf, where Nc is thenumberofsubcarri