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Title VoIP Jitter in 3GPP Long Term Evolution Networks -Edición Única
VoIP Jitter in
3GPP Long Term Evolution Networks
by
Thesis
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,
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.
Finally but not the last, I want to thank God for giving me strength, patience,
and the wonderful opportunity to be alive.
V
VoIP Jitter in
3GPP Long Term Evolution Networks
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.
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
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
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.
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.
Jitter is the variation of delay, where packets arrive at random times at the
receiver. In other words, the kth packet is expected to arrive at a time but it
is received at , where is jitter. When jitter is constant, it can be filtered
out or compensated in a deterministic way. However, it often exhibits a random
behavior [3]. Jitter results in speech intelligibility disruptions [4]; hence the end-to-
end jitter has to be small enough not to be noticeable.
3GPP Long Term Evolution, an all-IP based network, is not exempt from
jitter. Hence, research about this phenomenon is necessary to assure the feasibility
of VoIP services over LTE.
Chapter 1. Introduction 3
1.2 OBJECTIVE
In order to determine the feasibility of VoIP services over the LTE mobile
networks, the purpose of this thesis is to analyze the jitter phenomenon.
Particularly the impact on jitter caused by network congestion, retransmissions,
and the modulation and coding scheme, for different radio channel conditions is
studied. VoIP traffic, physical layer and MAC layer simulations are developed.
1.3 JUSTIFICATION
In the packet-switched LTE network, services must be provided in a fast,
efficient and reliable way, including services substituting their CS counterparts.
Voice services will be offered in the form of Voice-over-IP. Since voice is the most
widespread service, special care must be taken to assure a successful deployment
of future LTE networks.
In the literature exists a variety of studies about VoIP over LTE [5] [6] [7].
Nevertheless, they mainly focus on capacity, coverage, or scheduling issues.
However, there are not researches identifying the jitter phenomenon and its
behavior. This thesis pretends to research jitter under diverse channel conditions,
and its impact on the VoIP QoS.
1.4 CONTRIBUTION
3GPP LTE is the next step towards 4G mobile communications with
performance comparable to wire-line networks. Careful planning and design must
be carried out to assure a successful deployment for both, users and network
operators. LTE must be able to adapt to a variety of traffic such as data, voice, and
video. Services currently provided through circuit-switched systems are expected
to have a similar equivalent in LTE, an all-IP based network.
Voice is the most widespread service and represents the main revenue for
network operators. Users expect at least the same Quality of Service provided by
CS networks, while operators look for an increase in capacity and reduction of
costs. Such objectives can be reached through Voice-over-IP (VoIP). This service
has the following characteristics: Bursty low bitrate traffic, strict packet delay-based
Chapter 1. Introduction 4
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.
This is an iterative process since any phase can directly affect the others.
5
Chapter 2. 3GPP Long Term Evolution 6
Initiated in 2004, the Long Term Evolution project focused on enhancing the
Universal Terrestrial Radio Access Network (UTRAN) and optimizing 3GPP's radio
access architecture. The design targets were [5]:
1
Final specifications consider 1.4, 3, 5, 10, 15 and 20 MHz.
2
5 percentile - 95% of the users have better performance
th
Chapter 2. 3GPP Long Term Evolution 7
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.
2.3 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
3
E-UTRAN is the official standard's name for LTE, the entire radio network.
Chapter 2. 3GPP Long Term Evolution 8
eNB
UE
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
LTE can be operated in different bandwidth sizes. The main reason for this
is that the amount of spectrum available depends on the frequency band and the
particular operator's situation. Originally it was stated in [6] as a list of LTE
spectrum allocations from 1.25 to 20 MHz, although final specifications consider
only 1.4, 3, 5, 10, 15 and 20 MHz.
Pair and unpair spectrum, i.e. FDD and TDD modes, are supported.
Frequency Division Duplex (FDD) entails that downlink and uplink take place in
Chapter 2. 3GPP Long Term Evolution 10
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
Δf = 1/TU
Pulse shape
T = 1/Δf
u
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
period of a 20 MHz OFDM signal with a FFT size of 2048. That isTs = 1/f = 1/ (15
4
7.5 KHz is also considered for multi-cell broadcast messages
Chapter 2. 3GPP Long Term Evolution 12
OFDM symbol inserted at the beginning. The CP has been chosen to be slightly
longer than the longest expected delay spread in the radio channel. LTE defines
two cyclic prefix lengths, the normal CP and the extended CP. Normal CP is the
expected operation mode for LTE, and its size has been set at ~4.7 us, enabling
the system to cope with path delay variations up to about 1.4 Km. Since the length
of an OFDM symbol is ~66.7 us, about a reduction of 6.6% in the effective data
rates is experienced. Extended CP is designed to provide robustness against multi-
path effect in larger cells, and for use with multi-cell broadcast messages. It
provides protection for up to 10 Km delay spread with a capacity loss of 20%.
Subcarriers Subcarriers
User 2
User 3
OFDM OFDMA
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)
#0 #1 #2 #3 #18 #19
One subframe
5.2 µs 4.7 µs
160 samples 144 samples
Normal CP
Δf=15kHz
5
Spatial-domain is also considered for MIMO communications (1 grid per antenna)
Chapter 2. 3GPP Long Term Evolution 14
CP.
6
Hence, a RB is formed by 84 or 72 resource elements.
Chapter 2. 3GPP Long Term Evolution 15
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.
Additionally, there also exist the UE-specific reference signals (to be used for an explicit
7
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
O F D M symbol
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)
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].
Chapter 2. 3GPP Long Term Evolution 18
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.
Modulation
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 CQI provides the eNodeB information about the link adaptation
parameters the UE can support at the time (taking into account the transmission
mode, the UE receiver type, number of antennas and interference situation
8
Rank Indicator (RI) and Pre-coding Matrix Indicator reports are used for MIMO schemes.
Chapter 2. 3GPP Long Term Evolution 21
experienced at the given time) [2]. This report is represented by a CQI index which
indicates the modulation scheme and coding rate that should, preferably, be used
for the downlink transmission such that the BLER does not exceed 10%. Details
about the CQI will be given in Chapter 5.
2.9 HYBRID-ARQ
Transmissions over wireless channels are subject to errors, for example due
to variations in the received signal quality. LTE implements error detection and
correction through HARQ, which makes use of the following techniques:
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.
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.
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
Parameter Value
Encoder frame 2 0 ms
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):
Chapter 3. Voice-over-IP 25
• 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].
20 ms
160 ms
Headers Payload
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
3.3 V O I P TRAFFIC M O D E L
Silence Talking
(1 - a )
(State 0) (State 1) (1-b)
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
1
E [TS] =
a
1
E [TS] =
b
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
1
A voice frame duration is 20 ms.
Chapter 3. Voice-over-IP 28
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 ]
P(N = n ) = p , i = 1 , . . . , Σ p
i i i = 1
i
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
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
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
Parameter Value
4.1 T H E JITTER C O N C E P T
along the PS network it suffers delay variations t , called jitter. Then, the packet
n
31
Chapter 4. Introduction toJitter 32
Transmitter
T 2T 3T 4T
Time
Receiver
tp tp tp tp
T1 = 0 T2 T3 T4
tp - Propagation time
T - Packet periodicity
Ti - Jitter
4.2.1 SCHEDULER B U F F E R
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.
4.2.2 H A R Q RETRANSMISSIONS
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.
Chapter 4. Introduction to Jitter 34
The RLC protocol can be operated in three modes to adapt to the type of
transmission:
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.
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:
Since packets arrive at their destination at random times due to jitter, the
user may perceive anomalies in the stream, experienced as static, strange noise
effects, garbled words or even missed words or syllables. In Figure 4.4 a jitter
Chapter 4. Introduction to Jitter 36
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.
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
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.
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
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.
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.
The simulation can be split into 3 sections. VoIP traffic (see Chapter 3.5,
page 29), PHY layer, and MAC protocol (Scheduler).
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.
Chapter 5.VoIP Jitter in LTE 40
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.
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
Chapter 5.VoIP Jitter in LTE 43
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
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
Link Adaptation
Channel Quality Indicator. The possible MCS suggested by a mobile terminal are
described in Table 5.2.
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
Chapter 5.VoIP Jitter in LTE 46
in Figure 5.6. Using the obtained line, an effective SNR can be mapped to a CQI
value [14].
-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)
14
13
12
11
10
8
CQI
0
-10 -5 0 5 10 15 20
SNR [dB]
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.
Chapter 5.VoIPJitter in LTE 47
Priority System
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
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
Chapter 5.VoIP Jitter in LTE 49
periodicity of 8 ms. As the number of users keeps increasing, the jitter cell profile
spreads, leading to the subsequent satisfaction drop.
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
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.
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.
Chapter 5.VoIP Jitter in LTE 51
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.
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
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
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 80 90 0 50 100 150 200 250 300 350 400 450
Jitter (ms) Jitter (ms)
SNR = 8 dB SNR = 8 dB
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
Jitter (ms) Jitter (ms)
a) 25 UEs b) 50 UEs
SNR = 8 dB SNR = 8 dB
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
Jitter (ms) Jitter (ms)
54
Chapter 5.VoIP Jitter in LTE
SNR = 8 dB SNR = 8 dB
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
Jitter (ms) Jitter (ms)
55
Chapter 5.VoIP Jitter in LTE
SNR = 12 dB SNR = 12 dB
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
Jitter (ms) Jitter (ms)
a) 25 UEs b) 50 UEs
SNR = 12 dB SNR = 12 dB
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
Jitter (ms) Jitter (ms)
56
Chapter 5.VoIP Jitter in LTE
SNR = 12 dB SNR = 12 dB
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
Jitter (ms) Jitter (ms)
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
Jitter (ms) Jitter (ms)
57
Chapter 5.VoIP Jitter in LTE
SNR = 12 dB SNR = 12 dB
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
Jitter (ms) Jitter (ms)
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
Jitter (ms) Jitter (ms)
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.
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.
A.1 T H E MUTI-CARRIER C O N C E P T
S
C = B W ·1og ( l + 2
N)
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
E b
)
γ ≤log (1 + γ
N
2
0
E -1
{E } 2γ
=
b b
≥min
N 0 N 0 γ
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 .
Figure A.3: a) F D M , b) O F D M
N -1
c
k=0
.
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
(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
Where:
a,
k 0 ≤k < N c
a'k= { a, N c ≤k < N
A.3.2 CYCLIC-PREFIX
Direct path
Reflected path
70
Nomenclature 71
[2] Holma, Harri and Toskala, Antti., LTE for UMTS: OFDMA and SC-FDMA
Based Radio Access. s.l. : John Wiley & Sons, 2009.
[3] Munoz, D., et al., "Heavy tail jitter in mobile packet networks." IEEE,
2001, Vol. 3, pp. 2224-2228 vol.3.
[5] Wang, Haiming, Jiang, Dajie and Tuomaala, E., "Uplink capacity of VoIP
on LTE system." 2007. Communications, 2007. APCC 2007. Asia-Pacific
Conference on. pp. 397-400.
73
Bibliography 74
[11] Dahlman, Erik, et al., 3G Evolution: HSPA and LTE for Mobile
Broadband. s.l. : Elsevier, 2008.
[15] G.114, ITU-T Recommendation., "One way transmission time." One way
transmission time. 2003.
[16] Ross, Sheldon M., Introduction to Probability and Statics for Engineers
and Scientists. [ed.] Academic Press. s.l. : Elsevier, 2009.
[23] Sesia, Stefania, Toufik, Issam and Baker, Matthew., LTE: The Long
Term Evolution. From Theory to Practice. s.l. : John Wiley & Sons, 2009.