3GPP LTE system model analisys and simulation of

video transmission

Teodor Iliev
Department of Telecommunication
University of Ruse
Ruse, Bulgaria
tiliev@ecs.uni-ruse.bg
Grigor Mihaylov
Department of Telecommunication
University of Ruse
Ruse, Bulgaria
gmihaylov@uni-ruse.bg


Abstract The era of new wireless communications is upon us.
Telecommunications technologies are evolving at an unbelievable
pace, driven by innovation, speed, quality, convenience, and cost.
Long-Term Evolution (LTE) is the standard recently specified by
the 3GPP on the way towards fourth-generation mobile. LTE
leads a big technological improvement as compared with the
previous 3G standard. In this paper are presented the main
technical features of LTE standard. An analysis of LTE physical
layer is also given and simulation of video transmission with
different resolution and frame rates was conducted.
Keywords- LTE; mobile network; video transmission; jitter;
throughput
I. INTRODUCTION

3rd Generation Partnership Project (3GPP) Long Term
Evolution (LTE) is the evolution of the Third-generation of
mobile communications, UMTS, to the Fourth generation
technology, that is essentially a wireless broadband Internet
system with voice, multimedia and other services. The
specifications related to LTE are formally known as the
Evolved Universal Terrestrial Radio Access (E-UTRA), for
support of wireless broadband data service up to 300 Mbps in
the downlink and 75Mbps in the uplink [1]. As multimedia
communications become increasingly popular, mobile
communications are expected to reliably support high data rate
transmissions. Multiple input multiple output (MIMO)
technologies has been treated as an emerging technology to
meet the demand for higher data rate and better cell coverage
even without increasing average transmit power or frequency
bandwidth, since it was proved that MIMO structure
successfully constructs multiple spatial layers where multiple
data streams are delivered on a given frequency time resource
and linearly increases the channel capacity. In LTE MIMO
technologies have been widely used to improve downlink peak
rate, cell coverage, as well as average cell throughput.
LTE is designed to increase data rates and cell edge
bitrates, improve spectrum efficiency (unicast as well as
broadcast) and allow spectrum flexibility (1.25, 2.5, 5, 10, 15
and 20 MHz) for flexible radio planning. LTE has also to
reduce packet latency, the main restriction for real-time
services, such as VoIP, video-conferencing, stream video, etc.,
reduce radio access network cost as well as cost-effective
migration from earlier 3GPP releases and simplify its network
to a flat all-IP packet-based network architecture where all the
user plane radio functionalities are terminated at the enhanced
Node B (eNB).
II. 3GPP LTE: MAIN FEATURES
The radio access network of LTE is called E-UTRAN and
one of its main features is that all services, including real-time,
will be supported over shared packet channels. This approach
will achieve increased spectral efficiency which will turn into
higher system capacity with respect to current UMTS and
HSPA. An important consequence of using packet access for
all services is the better integration among all multimedia
services and among wireless and fixed services [2].
The main philosophy behind LTE is minimizing the
number of nodes. Therefore the developers opted for a single-
node architecture. The new base station is more complicated
than the Node B in WCDMA/HSPA radio access networks,
and is consequently called eNB (Enhanced Node B). The eNBs
have all necessary functionalities for LTE radio access network
including the functions related to radio resource management.
The new core network is a radical evolution of the one of third
generation systems and it only covers the packet-switched
domain. Therefore it has a new name: Evolved Packet Core.
Following the same philosophy as for the E-UTRAN, the
number of nodes is reduced. EPC divides user data flows into
the control and the data planes. A specific node is defined for
each plane plus the generic gateway that connects the LTE
network to the internet and other systems. The EPC comprises
several functional entities.
The overall structure of LTE is shown in Fig. 1, where: the
Mobility Management Entity (MME) is responsible for the
control plane functions related to subscriber and session
management; The Serving Gateway is the anchor point of the
packet data interface towards E-UTRAN. Moreover, it acts as
the routing node towards other 3GPP technologies; The Packet
Data Network (PDN Gateway) is the termination point for
sessions towards the external packet data network. It is also the
router to the Internet; The Policy and Charging Rules Function
(PCRF) controls the tariff making and the IP Multimedia
Subsystem (IMS) configuration of each user.
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MME / S-GW
MME / S-GW
eNB
eNB
S1
X2

Figure 1. LTE architecture
The most important technologies included in the new radio
access network are Orthogonal Frequency Division
Multiplexing (OFDM), multidimensional (time, frequency)
dynamic resource allocation and link adaptation, Multiple Input
Multiple Output (MIMO) transmission, turbo coding and
Hybrid Automatic Repeat reQuest (HARQ) with soft
combining. These technologies are shortly explained in the
following paragraphs.
Orthogonal Frequency Division Multiplexing is a kind of
multicarrier transmission technique with a relatively large
number of subcarriers. OFDM offers a lot of advantages. First
of all, by using a multiple carrier transmission technique, the
symbol time can be made substantially longer than the channel
delay spread, which reduces significantly or even removes the
intersymbol interference (ISI). In other words, OFDM provides
a high robustness against frequency selective fading. Secondly,
due to its specific structure, OFDM allows for low-complexity
implementation by means of Fast Fourier Transform (FFT)
processing. Thirdly, the access to the frequency domain
(OFDMA) implies a high degree of freedom to the scheduler.
Finally, it offers spectrum flexibility which facilitates a smooth
evolution from already existing radio access technologies to
LTE [3,4].
In the FDD mode of LTE each OFDM symbol is
transmitted over subcarriers of 15 or 7.5 kHz. One subframe
lasts 1ms, divided in two 0.5ms slots, and contains several
consecutive OFDM symbols (14 and 12 for the 15 and 7.5 kHz
modes, resp.).
In the uplink, Single Carrier Frequency Division Multiple
Access (SC-FDMA) is used rather than OFDM. SC-FDMA is
also known as DFT-spread OFDM modulation. Basically, SC-
FDMA is identical to OFDM unless an initial FFT is applied
before the OFDM modulation. The objective of such
modification is to reduce the peak to average power ratio, thus
decreasing the power consumption in the user terminals [5].
A. OFDM
OFDM systems break the available bandwidth into many
narrower sub-carriers and transmit the data in parallel streams.
Each subcarrier is modulated using varying levels of QAM
modulation, e.g. QPSK, QAM, 64QAM or possibly higher
orders depending on signal quality. Each OFDM symbol is
therefore a linear combination of the instantaneous signals on
each of the sub carriers in the channel. Because data is
transmitted in parallel rather than serially, OFDM symbols are
generally much longer than symbols on single carrier systems
of equivalent data rate.
Each OFDM symbol is preceded by a cyclic prefix (CP),
which is used to effectively eliminate ISI. Second, the sub-
carriers are very tightly spaced to make efficient use of
available bandwidth, yet there is virtually no interference
among adjacent sub-carriers (Inter Carrier Interference, or ICI).
These two unique features are actually closely related. In order
to understand how OFDM deals with multipath distortion, its
useful to consider the signal in both the time and frequency
domains.
The OFDM symbol consists of two major components: the
CP and an FFT period (TFFT). The duration of the CP is
determined by the highest anticipated degree of delay spread
for the targeted application. When transmitted signals arrive at
the receiver by two paths of differing length, they are staggered
in time as shown in Fig. 2.
Within the CP, it is possible to have distortion from the
preceding symbol. However, with a CP of sufficient duration,
preceding symbols do not spill over into the FFT period; there
is only interference caused by time-staggered copies of the
current symbol. Once the channel impulse response is
determined (by periodic transmission of known reference
signals), distortion can be corrected by applying an amplitude
and phase shift on a subcarrier-by-subcarrier basis. OFDM has
two principle weaknesses relative to single carrier systems:
susceptibility to carrier frequency errors (due either to local
oscillator offset or Doppler shifts) and a large signal peak-to-
average power ratio (PAPR). As discussed above, OFDM
systems can achieve zero-ICI if each subcarrier is sampled
precisely at its center frequency [6].

Signal
delay < CP
CP =
4.6875sec
TFFT =
66.667sec

Figure 2. OFDM eliminates ISI via longer symbol periods and a cyclic
prefix
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The time-sampled OFDM signal is converted into the
frequency domain by means of a fast Fourier transform
(FFT)which is a highly efficient means of implementing a
discrete Fourier transform (DFT). The DFT renders a discrete
finite sequence of complex coefficients which are given by:
1 ,..., 1 , 0 ,
1
0
/ 2
= =

N k e x X
N
n
N nk j
n k
t
(1) o + | = _. (1) (1)
The resulting Fourier spectrum has discrete frequencies at:
1 ,..., 1 , 0 = N k NT k
s
, (2) o + | = _. (1) (1)
where T
s
is the sample interval in the time domain and N is the
number of samples. Thus, the frequencies in the Fourier
representation are completely defined by the sample frequency
(1/T
s
) and the number of samples taken within the FTT period.
III. SYSTEM MODEL AND INFORMATION RATE
A. OFDMA and the LTE generic frame structure
OFDMA is an excellent choice of multiplexing scheme for
the 3GPP LTE downlink. Although it involves added
complexity in terms of resource scheduling, it is vastly superior
to packet-oriented approaches in terms of efficiency and
latency. In OFDMA, users are allocated a specific number of
subcarriers for a predetermined amount of time. These are
referred to as physical resource blocks (PRBs) in the LTE
specifications. PRBs thus have both a time and frequency
dimension. Allocation of PRBs is handled by a scheduling
function at the 3GPP base station (eNB).
In order to adequately explain OFDMA within the context
of the LTE, we must study the PHY layer generic frame
structure. The generic frame structure is used with FDD.
Alternative frame structures are defined for use with TDD. As
shown in Fig. 3, LTE frames are 10 msec in duration. They are
divided into 10 subframes, each subframe being 1.0 msec long.
Each subframe is further divided into two slots, each of 0.5
msec duration. Slots consist of either 6 or 7 ODFM symbols,
depending on whether the normal or extended cyclic prefix is
employed [7].
0 2 3 19 18 1
1 subframe, 1 ms
1 slot,
0.5 ms
1 frame,10 ms
1 3 4 2 5 6 0
10 11
1 3 4 2 5 6 0
7 OFDM symbols
Cyclic prefix

Figure 3. LTE generic frame structure
The total number of available subcarriers depends on the
overall transmission bandwidth of the system. The LTE
specifications define parameters for system bandwidths from
1.25 MHz to 20 MHz as shown in Table I. A PRB is defined as
consisting of 12 consecutive subcarriers for one slot (0.5 msec)
in duration. A PRB is the smallest element of resource
allocation assigned by the base station scheduler.
TABLE I. AVAILABLE DAWNLINK BANDWIDTH IS DIVIDED INTO
PHYSICAL RESOURCE BLOCKS
Bandwidth (MHz)
1,25 2,5 5,0 10,0 15,0 20,0
Subcarrier
bandwidth (kHz)
15
Physical resource
block (PRB)
bandwidth (kHz)
180
Number of
available PRBs
6 12 25 50 75 100
B. SC-FDMA
Consider the SC-FDMA uplink transmission scheme
depicted in Fig. 4 and equivalent block diagram of SC-FDMA
system shown on Fig.5. The only difference from OFDMA is
the addition of the N point DFT at the transmitter and the N
point IDFT at the receiver.
The transmitted signal block d = [d
0
, . . . , d
N1
]
T
of size N
spreads onto the N sub-carriers selected by the sub-carrier
mapping method. In other words, the transmitted signal vector
is pre-encoded by DFT before going to the OFDMA
modulator. For OFDMA transmission, a specific set of sub-
carriers is assigned to the user through the sub-carrier mapping
stage. Then multi-carrier modulation is performed via a Q point
IFFT (Q>N), and a cyclic prefix (CP) longer than the
maximum channel delay is inserted to avoid interblock
interference. The frequency selective channel can be
represented by a tap delay line model with the tap vector h =
[h
0
, h
1
, . . . , h
L
]
T
and the additive white Gaussian noise
(AWGN) v N(0,N
0
). At the receiver, the CP is removed and a
Q point FFT is performed. A demapping procedure consisting
of the spectral mask of the desired user is then applied,
followed by zero forcing equalization which involves a scalar
channel inversion per sub-carrier. For SC-FDMA, the
equalized signal is further transformed to the time domain
using an N point IDFT where decoding and detection take
place.

N-point
DFT
N-point
IDFT
Subcarrier
mapping
Subcarrier
demapping
Q-point
IFFT
Q-point
FFT
DAC
ADC
Add cyclic
prefix
Remove
cyclic prefix
H

c
h
a
n
n
e
l
v
d x
M
y
r
EQ
d
x
M
H
^
^
OFDMA SC-FDMA

Figure 4. LTE generic frame structure
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We assume that in the uplink the users channels are
perfectly measured by the base station (BS), where the resource
allocation algorithm takes place and its decision is then sent to
the users via a signaling channel in the downlink. For
simplicity, we start with the single-user single-input single-
output system and then extend it to the multi-user case with
multiple antennas at the BS. F
N
is the NN Fourier matrix with
the (n, k) entry | | ( )
N nk j
k n n
e N F
/ 2
,
1
t
= and
H
N
F denotes the
inverse Fourier matrix. Further on, the assignment of data
symbols x
n
to specific sub-carriers is described by the Q N
sub-carrier mapping matrix M with the entry m
q,n
=1 if the n
data is assigned to the q sub-carrier and m
q,n
=0 otherwise,
0qQ-1, 0nN-1.
After CP removal at the receiver, the received block can be
written as:
v M HF y
x
H
Q
+ = , (3) o + | = _. (1) (1)
where x = [x
0
, . . . , x
N1
] is the transmitted block of the
OFDMA system, and H is a QQ circulant matrix with the first
column h = [h
0
, . . . , h
L1
, 0, . . . , 0]
T
.
The achievable sum rate of an OFDMA system can be
calculated as the sum of the rates of the assigned sub-carriers,
which is given by:

=
|
|
|
|
.
|

\
|
+ =
1
0 0
2
2

1 log
N
n
n n
OFDMA
N
h P
C , (4) o + | = _. (1) (1)
where P
n
is the power allocated to the nth sub-carrier. Note that
the employment of a zero forcing (ZF) equalizer performing
channel inversion for each sub-carrier preserves the capacity
since the resulting signal-to-noise ratio (SNR) for each sub-
carrier remains unchanged. To maximize the OFDMA rate
subject to the total transmit power constraint P
total
, the
assignment of the transmit power to the n independent
Gaussian sub-channels should follow the water filling
principle, and so the optimal power P
n
of the n
th
sub-carrier is:

|
|
|
|
.
|

\
|
=
2
0

, 0 max
n
n
h
N
P , (5) o + | = _. (1) (1)
where the positive constant must be chosen in order to fulfill
the total transmit power constraint [8].

N-point DFT N-point IDFT
OFDM channel
ZF
equalizer
d0
dN-1
x0
xN-1
0
N-1
1/
1/
x0
xN-1
d0
dN-1
0
^
h
1
^
N h
1
^
N h
0
^
h
^
^
^
^

Figure 5. Equivalent block diagram of SC-FDMA systems
IV. SIMULATION RESULTS
For transmission simulation we use NS-2 network
simulator with two additional modules. The first one is
EvalVid a complete framework and tool-set for evaluation of
the quality of video transmitted over a real or simulated
communication network [9]. Besides measuring QoS
parameters of the underlying network, like loss rates, delays,
and jitter, it supports also a subjective video quality evaluation
of the received video based on the frame-by-frame PSNR
calculation. The tool-set has a modular construction, making it
possible to exchange both the network and the codec. The
processing of the data takes place in 3 stages. The first stage
requires the timestamps from both sides and the packet types.
The results of this stage are the frame-type based loss rates and
the inter-packet times. Furthermore the erroneous video file
from the receiver side is reconstructed using the original
encoded video file and the packet loss information. This video
can now be decoded yielding the raw video frames which
would be displayed to the user. At this point a common
problem of video quality evaluation comes up. Video quality
metrics always require the comparison of the displayed
(possibly distorted) frame with the corresponding original
frame. In the case of completely lost frames, the required
synchronization cannot be kept up. The second stage of the
processing provides a solution to this problem. Based on the
loss information, frame synchronization is recovered by
inserting the last displayed frame for every lost frame. This
makes further quality assessment possible. The thus fixed raw
video file and the original raw video file are used in the last
stage to obtain the video quality.
The second additional module we use is LTE/SAE module,
introduced in [10]. The system model includes LTE/SAE
network model, traffic model and flow control model. In the
network model, some network configuration parameters can
easily be changed, for example it can be defined any number of
UEs, bandwidth between the network elements and the usage
of the optimization features; others cannot be changed so easily
due to the limitation of the implemented model, such as eNB
and aGW number. In the simulated LTE/SAE network, the
following network elements are included 1 server (provide
HTTP, FTP and signaling services), 1 aGW (provide HTTP
cache, flow control), 1 eNB (provide flow control information)
and many UEs. In this module three types of traffic are
provided Conversational, Streaming and Interactive traffic.
The Conversational traffic simulates transmission of Voice
over IP, Video conferencing calls and Telephony speech.
Interactive traffic simulates Web browsing and e-mail
reading/sending. The most important traffic type for us is the
Streaming traffic. Its one-way traffic and it is used for Audio
and Video streaming. The QoS mechanisms provided in the
cellular network have to be robust and capable of providing
reasonable QoS resolution. More accurate flow controls, better
performance the system can get. However, the systems
resource, such as CPU and memory, is limited. Balance
between the system performance and resource is needed. HVQ
flow control provides the good solution to get the high ratio of
performance over resource. The basic principle of the virtual
queue based flow control is shown in the following figure.
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Arrivals
Virtual queue
Real queue
C
k*C

Figure 6. Virtual queue based flow control
A virtual queue is a fictitious queue with a capacity less
than the actually available capacity. The motivation for using
the virtual queue is that it provides advance warning of
congestion. For each real queue there is a corresponding virtual
queue or a set of virtual queues, one for each differently treated
traffic class. When packets arrive into the real queue, the
virtual queue length is also updated by the length of the
received packet. The draining rate of the virtual queue is k*C
(k1), where C is the draining rate of the real queue. When the
virtual queue length upper threshold reaches, the flow control
information is send from air interface to the S1 interface. S1
interface will take the corresponding flow control actions. If
the flow control information is about UE, the packets belong to
the UE are blocked; if the flow control information is about
cell, all the packets belong to the cell where the UE locates are
blocked; if the flow control information is about the eNB, all
packets belong to the eNB where the UE locates are blocked.
The blocking is cancelled when the lower threshold is
triggered.
Fig. 7 shows the architecture of the simulation. During the
simulation study, three video clips with different frame rates,
resolution and size are used. The parameters of the clips are
shown in Table II.
TABLE II. PARAMETERS OF TRANSMITTED VIDEO CLIPS
Clip Resolution FPS Duration, min
Size,
Mb
A Bit of a Pickle 1920x1080 24 03:45 90,6
Massive Attack 640x358 30 05:20 85,0
Tatu 368x208 30 03:37 18,0

It can be seen the most important parameter in wireless
video transmission is the jitter. From the conducted simulation
and the graph, shown in Fig. 8, it can be seen, that this
parameter has a very low value, which is prerequisite for
excellent delivery and play in user side.

DropTail
DLS1Queue
ULS1Queue
D
L
A
irQ
ue
ue
ULAirQueue
Server
aGW
eNB
UEs

Figure 7. Architecture of the system

Figure 8. Jitter of transmitted video clips
From the diagram also can be seen, that the clip with
highest resolution has the lowest jitter, which is very good for
contemporary video consumption.
On Fig. 9 we represent the delay of the conducted
simulation. From diagram can be seen, that the video clip with
lowest resolution and size has the smallest delay.

2,038
2,039
2,04
2,041
2,042
2,043
2,044
User
1
User
2
User
3
User
4
User
5
User
6
User
7
User
8
User
9
User
10
m
s
Delay
A Bit of a Pickle
Massive Attack
Tatu

Figure 9. Delay in video transmission
Fig. 10 shows the results for throughput from conducted
simulation. This throughput depends on duration and size of
the clip. It can be seen that the video file with lowest resolution
368x208 has the smallest value of throughput, since it is the
smallest in size (Mb).

0
0,05
0,1
0,15
0,2
0,25
0,3
0,35
0,4
0,45
User
1
User
2
User
3
User
4
User
5
User
6
User
7
User
8
User
9
User
10
M
b
/
s
Throughput
A Bit of a Pickle
Massive Attack
Tatu

Figure 10. Throughput of the video clips

0
1
2
3
4
5
6
7
8
9
10
User
1
User
2
User
3
User
4
User
5
User
6
User
7
User
8
User
9
User
10
x
1
0
-
6
Jitter
A Bit of a Pickle
Massive Attack
Tatu

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V. CONCLUSIONS AND FUTUREWORK
LTE has been designed as a future technology to cope with
next user requirements. We have presented a framework for an
analytical comparison between the achievable information rate
in SC-FDMA and that in OFDMA. Based on the conducted
simulation it can be seen that video files with higher resolution
(1920x1080) have lower jitter than the video files with
resolution (368x208).
This paper has also given an initial insight into the new
technical proposals currently under discussion in the
framework of 3GPP. Future work will focus on more detailed
analysis, design and simulation of transmission of video
information with high resolution to more subscribers (UEs).
Finally, combining channel codding influence and channel
capacity is an emerging area that will be the subject of future
research.
REFERENCES

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LTE-Advanced, EURASIP Journal onWireless Communications and
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Evolved UTRAN (E-UTRAN), v.8.0.0, December 2008.
[3] 3GPP R1-072578, Summary of Downlink Performance Evaluation,
May 2007.
[4] On theWay towards Fourth-GenerationMobile: 3GPP LTE and LTE-
Advanced, EURASIP Journal onWireless Communications and
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[5] 3GPP R1-072261, LTE Performance EvaluationUplink Summary,
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[6] N. Wang and S. Blostein, Comparison of CP-based single carrier and
OFDMA with power allocation, IEEE Transactions on
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[8] R. S. Cheng and S. Verdu, Gaussian multiaccess channels with ISI:
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[9] J. Klaue, B. Rathke, and A. Wolisz, "EvalVid - A Framework for Video
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[10] Q. Qiu, J. Chen, L. Ping, Q. Zhang, X. Pan, LTE/SAE Model and its
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E
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