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sustainability

Article
Transmission Algorithm with QoS Considerations for
a Sustainable MPEG Streaming Service
Sang-Hyong Kim, Kwan-Jong Yoo and Yoojae Won *
Department of Computer Science and Engineering, Chungnam National University, Daejeon 34134, Korea;
kims@cnu.ac.kr (S.-H.K.); kjyoo@cnu.ac.kr (K.-J.Y.)
* Correspondence: yjwon@cnu.ac.kr; Tel.: +82-42-821-6294

Academic Editors: James J. Park and Han-Chieh Chao


Received: 23 December 2016; Accepted: 25 February 2017; Published: 2 March 2017

Abstract: With the proliferation of heterogeneous networks, there is a need to provide multimedia
stream services in a sustainable manner. It is especially critical to maintain the Quality of Service
(QoS) standards. Existing multimedia streaming services have been studied to guarantee QoS on
the receiving side. QoS has not been ensured due to the fact that the loss of streaming data to be
transmitted has not been considered in network conditions. With an algorithm that considers the QoS
and can reduce the overhead of the network, it will be possible to reduce the transmission error and
wastage of communication network resources. In this paper, we propose a scheme that improves the
reliability of multimedia transmissions by using an adaptive algorithm that switches between UDP
(User Datagram Protocol) and TCP (Transmission Control Protocol) based on the size of the data.
In addition, we present a method that retransmits essential portions of the multimedia data, thus
improving transmission efficiency. We simulate an MPEG (Moving Picture Experts Group) stream
service and evaluate the performance of the proposed adaptive MPEG stream service.

Keywords: sustainability; MPEG; scalability; layered coding; Quality of Service; Forward Error Correction

1. Introduction
The simultaneous advances in computing and information communication technologies have
enabled the processing of massive volumes of data using personal computers. Several studies analyze
multimedia data services [1–8] for a variety of communication media and devices. The process of
providing real-time multimedia services over mobile and broadband Internet connection has different
requirements in terms of the Quality of Service (QoS) [9–27]. This prompts the division of multimedia
data into two layers—the base layer and the enhancement layer. This modularization helps in the
customization of encoding and decoding operations as per the user connection capabilities. The layers
can be divided based on spatial and temporal aspects; the fundamental characteristics and scalability
of these have been previously analyzed [28–41]. Here, scalability refers to the adaptability of the
multimedia data service to a dynamic network environment with the aim of reducing data loss and
providing best QoS.
Several measures have been proposed to improve the efficiency of multimedia data services.
However, these are not suitable for deployment in resource constrained environments, where the
users experience poor QoS. A multi-layered methodology has been proposed to implement adaptive
multimedia transmission to improve efficiency [42–46]. In this, depending on the streaming bandwidth,
either TCP or UDP sessions are chosen for transmission through the heterogeneous network [47–56].
To prevent the loss of packets during the transfer, important data segments are selected using the
Forward Error Correction (FEC) methodology [57–65] and these are retransmitted.

Sustainability 2017, 9, 367; doi:10.3390/su9030367 www.mdpi.com/journal/sustainability


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The
The rest
rest of
of the
the paper
paper isis organized
organizedas asfollows.
follows.InInSection
Section2, 2,
the layered
the coding
layered codingmethodology,
methodology,a
data division method for adaptive MPEG system, is described. A description of the
a data division method for adaptive MPEG system, is described. A description of the adaptive QoS adaptive QoS
algorithm
algorithm based
based on
on session
session selection
selection and
and FEC
FEC isis provided
provided in
in Section
Section 3.3. The
The evaluation
evaluation of
of the
the proposed
proposed
algorithm
algorithm is provided in Section 4. Section 5 concludes the paper by providing an overall summary
is provided in Section 4. Section 5 concludes the paper by providing an overall summary
and
and describing
describing further
further research tasks.
research tasks.

2.
2. Related
Related Work
Work

Layered Coding Methodology


2.1. Layered
In order
order to
toadapt
adapttotochanging
changingnetwork
networkenvironments,
environments,multimedia datadata
multimedia must be divided.
must The QoS
be divided. The
is improved
QoS by adding
is improved decoding
by adding data.
decoding When
data. Whenthe the
network condition
network is good,
condition thethe
is good, multimedia
multimedia data is
data
transmitted
is transmittedas as
a single unitunit
a single without anyany
without interference. However,
interference. when
However, the network
when is relatively
the network slow,
is relatively
slow, we perform
we perform basicencoding
basic data data encoding to improve
to improve the QoS.the QoS.

2.1.1. Spatial
2.1.1. Spatial Division
Division Method
Method
The type
The type of
of division
division method
methodto tobe
beimplemented
implementeddepends
dependsononthethedata
dataresolution.
resolution. The encoding
The encoding of
the basic layers is performed first and the extension area is encoded based on the in
of the basic layers is performed first and the extension area is encoded based on the in betweenbetween differential.
Figure 1 illustrates
differential. Figure the process for
1 illustrates anprocess
the object using theobject
for an minimum
usingvalue. There is an
the minimum expansion
value. There ofis the
an
layers withofanthe
expansion increase
layers in thean
with value of theinexpression
increase layer.
the value of the The left upper
expression area
layer. Theis designated as the
left upper area is
low frequency area while the right lower area has higher frequency. The area information
designated as the low frequency area while the right lower area has higher frequency. The area is used to
determine the
information is spatial
used toresolution.
determine the spatial resolution.

Figure
Figure 1.
1. Spatial
Spatial division
division method.
method.

2.1.2. Time Division Method


2.1.2. Time Division Method
In the time division method, the process of dividing the base layer and extended layer is the
In the time division method, the process of dividing the base layer and extended layer is the
same as that in the spatial division method. However, the time division method utilizes the time
same as that in the spatial division method. However, the time division method utilizes the time
difference between the base layer and the extended layer. This is done by identifying different
difference between the base layer and the extended layer. This is done by identifying different streams
streams in the frame expression and segregating them. When the base layer is subject to continuous
in the frame expression and segregating them. When the base layer is subject to continuous replay, the
replay, the result is an expression of wide movement. A smooth streaming replay can be acquired by
result is an expression of wide movement. A smooth streaming replay can be acquired by adding the
adding the extended layers to the wide movement frame.
extended layers to the wide movement frame.
The time division method uses the time axis and is shown in Figure 2. The redundancy is
The time division method uses the time axis and is shown in Figure 2. The redundancy is removed
removed for coding movements and can be categorized by its application methods.
for coding movements and can be categorized by its application methods.
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Figure 2. Time division method.

2.1.3. Relationship between Layers


Theoretically, an MPEG streamFigure can be Time
divided intomethod.
192 different layers by applying time and
Figure 2.
2. Time division
division method.
other division methods. The time division method produces up to 15 video layers that exhibit inter-
dependencies
2.1.3. as shown in Figure 3. Here, “n” represents the number of layers generated using spatial
2.1.3. Relationship
Relationship between
between Layers
Layers
division. The base layer T_layer1 is a time division layer used to decode T-layer2, which in turn is
usedTheoretically, an MPEG
to decode T_layer3.
Theoretically, an MPEG stream
stream cancan be
be divided
divided into
into 192
192 different
different layers
layers by
by applying
applying time
time and
and
other division
otherLayers
division methods.
that
methods. The
are dividedThetime
based division
time method
ondivision
the definitionproduces
method up to
of produces
an image up15to
also video
have layerslayers
that exhibit
dependencies.
15 video Each
that inter-
layer
exhibit
dependencies
exists as
inter-dependencies shown
independently, in
as but
shownFigure 3. Here,
is dependent
in Figure 3.on“n” represents
other
Here, layers
“n” the number
for replay.
represents of layers generated
For decoding
the number using
a generated
of layers spatial
lower layer, the
using
division.
higher The base
layer should
spatial division. layer T_layer1
havelayer
The base already is a time division
beenisdecoded.
T_layer1 layer used
This results
a time division to decode
in the
layer used T-layer2,
existence
to decode which
of a base
T-layer2, in turn
layer
which and
in turnis
used to
arbitrary decode
is used tolower
decodeT_layer3.
layers for encoding and decoding operations.
T_layer3.
Layers that are divided based on the definition of an image also have dependencies. Each layer
exists independently, but is dependent on other layers for replay. For decoding a lower layer, the
higher layer should have already been decoded. This results in the existence of a base layer and
arbitrary lower layers for encoding and decoding operations.

Figure 3. Relationship between the layers.

2.2. Adaptive MPEG


Layers that are System
divided based on the definition of an image also have dependencies. Each layer
existsAn
independently,
optimum streambut is dependent
transfer on other
takes layers
into for replay. For
consideration thedecoding
real-timea lower
networklayer,bandwidth
the higher
Figure 3. Relationship between the layers.
layer should have already been decoded. This results in the existence of a base
measurement. Figure 4 represents the proposed MPEG system structure that consists of an algorithm layer and arbitrary
lowersplits
that layers for
a MPEG encoding stream
multimedia and decoding
into manyoperations.
streams and transfers each of them to the client. The
2.2. Adaptive System
transferred streams are merged into a single multimedia stream and are sent to the client through the
2.2. Adaptive
Anplayer. MPEG System
optimum stream cantransfer takes into service
consideration thethereal-time
MPEG This process provide a better by using real-timenetwork bandwidth
network bandwidth
measurement.
An optimum
information toFigure 4 represents
stream
estimate transfer the into
takes
the network proposed
and MPEG
loadconsideration system
minimize datastructure
the real-time that
loss.network
The consists system
bandwidth
proposed of measurement.
an algorithm
uses an
that splits a multimedia
Figure 4 represents
autonomous real-time stream
thebandwidth into
proposed MPEG many streams
system
calibration and
structure
method transfers
that consists
to monitor each of them to
of an algorithm
the changing the client.
that The
splits
stream information. a
It
transferred
multimedia stream into many streams and transfers each of them to the client. The transferred streams the
streams are merged into a single multimedia stream and are sent to the client through are
MPEG
mergedplayer. This multimedia
into a single process canstream
provide anda better
are sentservice by using
to the client the real-time
through the MPEGnetwork bandwidth
player. This process
information to estimate the network load and minimize data loss. The proposed
can provide a better service by using the real-time network bandwidth information to estimate the system uses an
autonomous real-time bandwidth calibration method to monitor the changing stream information. It
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network load and minimize data loss. The proposed system uses an autonomous real-time bandwidth
calibration
sends method
Sustainability
first 2017, 9, to
metadata 367monitor the changing
to the client streamstream
and initiates information. It first
service. Thissends
helpsmetadata
to adapttoto
thebandwidth
client and
4 of 13
initiates
changes stream service.
and provide Thisstream
better helps to adapt to bandwidth changes and provide better stream service.
service.
first sends metadata to the client and initiates stream service. This helps to adapt to bandwidth
changes and provide better stream service.

Figure 4.
Figure 4. Adaptive
Adaptive MPEG
MPEG system transfer module.
system transfer module.
Figure 4. Adaptive MPEG system transfer module.
3. Adaptive
3. Adaptive QoS QoS Transfer Algorithm
Transfer Algorithm
3. Adaptive QoS Transfer Algorithm
Figure 55illustrates
Figure illustratesthe the temporal-fidelity
temporal-fidelity scalability
scalability method
method usedused in transferring
in transferring data afrom
data from server a
server Figure
to the 5 illustrates
client. Steps the show
1–4 temporal-fidelity
the initial scalability
server and method
client used The
states. in transferring
server sends data
a from a
requested
to the client. Steps 1–4 show the initial server and client states. The server sends a requested multimedia
server to the
multimedia fileclient.
to the Steps
client 1–4 show
first and the
theinitial
actualserver and clientdata
multimedia states. Theasserver
later shown sends a requested
in received
steps 5–7.file
The
file to the client
multimedia
first
file to
and
the
the actual
client first
multimedia
and the actual
data later as shown
multimedia data
in steps
later as
5–7. The
shown in steps 5–7. this
The is
is
received
merged file
into ais merged
single file into
using a single
the file
merging using
module.the merging
As shown module.
in steps As shown
8–11, this in
is steps
played 8–11,
back to the
received file is merged into a single file using the merging module. As shown in steps 8–11, this is
playedvia
client backplayer.
to the The
client viaMonitor
a player.obtains
The QoS theMonitor obtains the current session information based
played aback QoS
to the client via a player. The QoS currentobtains
Monitor sessionthe
information basedinformation
current session on the multimedia
based
on the
data multimedia datainformation
(steps 10–11). This information isQoS
resent to the serverdetermines
QoS Adapter, which
on(steps 10–11). This
the multimedia data (steps 10–11). is resent to the serveris
This information Adapter,
resent to the which
server QoS Adapter, thewhich
data to
determines
be determines the
transferredthe data
next to be
(steps transferred
12–13). next (steps 12–13).
data to be transferred next (steps 12–13).

Figure5.5.Adaptive
Figure Adaptive QoS
QoS algorithm.
algorithm.
Figure 5. Adaptive QoS algorithm.
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Sustainability 2017, 9, 367 5 of 13

3.1. Session-Based Adaptive QoS Algorithm


3.1. Session-Based Adaptive QoS Algorithm
Existing streaming services run over UDP, which enables them to meet real-time requirements.
Existing streaming services run over UDP, which enables them to meet real-time requirements.
However, there is a possibility of losing important data when using UDP, which could result in a loss
However, there is a possibility of losing important data when using UDP, which could result in a loss
of integrity. On On
of integrity. thetheother
otherhand,
hand,thetheTCP
TCP ensures reliabledelivery
ensures reliable delivery butbut at the
at the costcost of increased
of increased time time
latency. The The
latency. proposed
proposed session
sessiontransfer
transferalgorithm
algorithm exploits theadvantages
exploits the advantages of both
of both the the protocols.
protocols. It usesIt uses
UDPUDPto transfer real-time
to transfer datadata
real-time andandTCPTCPto to
transfer critical
transfer criticalhigh-priority
high-priority data. The categorization
data. The categorizationofof the
data the data is determined
is determined by clientbyfeedback
client feedback
and based andonbased
this,on
thethis, the session
session group isgroup is increased
increased or
or decreased.
decreased.
Figure 6 illustrates the session size (group) calculation at time instances A and B. The client
side countsFigure
the6number
illustratesofthe sessionnsize
packets for(group) calculation
the final transferatand
timethe
instances A and
playback B. The
time client
from side n to
packet
counts the number of packets n for the final transfer and the playback time from packet n to packet n
packet n + 1, where n >0, to determine whether the receiving operation is complete. When the time of
+ 1, where n >0, to determine whether the receiving operation is complete. When the time of
completion is smaller, it indicates that there is still time to receive more packets. The difference value
completion is smaller, it indicates that there is still time to receive more packets. The difference value
can be
canused to determine
be used to determine thethe
time
timerequired
required to sendmore
to send morepackets
packets and
and thethe
nextnext session
session packet
packet number.number.
The steps in the session size (group) calculation algorithm are shown
The steps in the session size (group) calculation algorithm are shown in Algorithm 1.in Algorithm 1.

Figure6.6.Session
Figure Session group
groupcalculation.
calculation.

Algorithm 1. Session size calculation


Algorithm theSession
NRT : 1. time it size
takescalculation
to play the next GoV
NRT : the time it takes to playcomplete
CRT : the time it takes to the next the
GoV current session
CSPN : the number of packets received
CRT : the time it takes to complete the current in the current session
session
CSPNiLayerPackNo
: the number :of the number
packets of packets
received in acurrent
in the particular spatial and temporal layer obtained from the metafile
session
NSS : the size
iLayerPackNo : theofnumber
the nextofsession to in
packets beatransmitted
particular spatial and temporal layer obtained from the metafile
NSS : PN
the:size
the number
of the nextof extra packets
session to bethat can be sent during the free time
transmitted
PN : the number of extra packets that can be sent during the free time
1 T = NRT-CRT;
1 T = 2NRT-CRT;
PN = ROUND((T * CSPN)/CRT);
2 PN 3= NSS = 0;
ROUND((T * CSPN)/CRT);
3 NSS = 0;
4 for4 (procedure
for (procedure of Spatial
of Spatial layer)
layer) { {
5 5for (procedure
for (procedure of Temporal
of Temporal layer)
layer) { {
6 6 if ((NSS if ((NSS + iLayerPackNo)
+ iLayerPackNo) > CSPN
> CSPN + PN)
+ PN)
7 7 break; break;
8 8 else NSS else+=
NSS += iLayerPackNo;
iLayerPackNo;
9 9} }
10 } 10 }

3.2. FEC-Based Adaptive QoS Algorithm


3.2. FEC-Based Adaptive QoS Algorithm
The existing FEC method does not use client feedback information. The server calculation is
The existing FEC method does not use client feedback information. The server calculation is
performed for transmission operations only and places a heavy load on the network bandwidth. It is
performed for for
not suited transmission operations
a variable network only andasplaces
environment a heavy
it requires load
that the on the network
transmitted bandwidth.
data segments have It is
not suited
the same size. To address this issue, we propose that client feedback information be used to obtain have
for a variable network environment as it requires that the transmitted data segments
the same size. To
knowledge address
of the networkthiscondition.
issue, weThis
propose that client
information is usedfeedback information
to implement be used to
data redundancy obtain
and
knowledge of the network condition. This information is used to implement data redundancy and
minimize data loss. Figure 7 illustrates the indirect loss situation that arises during network transfer
after termination assuming that there is no higher layer.
Sustainability 2017, 9, 367 6 of 13

minimize2017,
Sustainability data9, loss.
367 Figure 7 illustrates the indirect loss situation that arises during network transfer
6 of 13
after termination assuming that there is no higher layer.

Figure 7. Indirect packet loss.


Figure 7. Indirect packet loss.

Consider a five-layered data (T1F1~T1F5) scenario, where the layers decrease in importance
Consider
from a five-layered
top to bottom, i.e., T1F1data (T1F1~T1F5)
> T1F2 > T1F3 > T1F4scenario,
> T1F5.where the layers
Here, T1F2 decrease
needs T1F1 in importance
for replay and so
from top to bottom, i.e., T1F1 > T1F2 > T1F3 > T1F4 > T1F5. Here, T1F2 needs T1F1 for replay
on. Therefore, when transferring data, T1F1 is retransmitted to prevent indirect loss of other levels. and
so The
on. proposed
Therefore, when transferring
algorithm uses feedbackdata, T1F1 is retransmitted
information from the clienttoside
prevent indirect
to improve theloss
QoS.ofThe
other
levels. Theofproposed
amount algorithm
data transmitted byuses
the feedback
server andinformation
received byfrom
the the client
client side to improve
is calculated based on thethe
QoS.
Thefeedback
amountinformation and error rate
of data transmitted of the
by the current
server andnetwork.
receivedThe
by metrics used
the client to quantify based
is calculated this is on the
feedback information and error rate of the current network. The metrics used to quantify this is
=1− (1)
Gi Gi
Erate = 1 − ∑ SRLi / ∑ SSLi (1)
where Gi is the number of transmitted GoV, SSLi=is 0 the sizei =of
0 the transmitted data in GoV units, and
SRL is the size of the received data in GoV units.
where G Toi isreduce
the number of transmitted
the network GoV,
error rate, dataSSL
canisbe
theretransmitted,
size of the transmitted
but this isdata in GoV
a waste units, and
of network
SRL is the size The
bandwidth. of the
proposed data in GoV
receivedalgorithm units.
uses client feedback information to determine an adaptive
To reduce
quantity the network
of data error rate,Indata
for retransmission. thiscan be retransmitted,
mechanism, but this is are
the key challenges a waste of to
related network
the
determination
bandwidth. of the desired
The proposed QoS and
algorithm usesthe associated
client feedback redundancy.
informationThe QoS is determined
to determine an adaptiveby quantity
using
of feedback information from
data for retransmission. the mechanism,
In this client. We assume
the keythat the same
challenges areamount
relatedof
todata is sent again and
the determination of the
therefore the size of the next data packet is known. The data quantity decision
desired QoS and the associated redundancy. The QoS is determined by using feedback informationalgorithm is shown in
Algorithm 2.
from the client. We assume that the same amount of data is sent again and therefore the size of the
next data packet is known. The data quantity decision algorithm is shown in Algorithm 2.
Algorithm 2. Data quantity decision
iSizeofNet : the predictable network bandwidth
Algorithm 2. Data quantity decision
1 for (procedure of Temporal layer) {
iSizeofNet
2 : the predictable
for (procedure network
of Spatial layer)bandwidth
{
1 for
3 (procedure of Temporal
iSizeofLayer = size oflayer)
layer;{
2 4for (procedure
iSizeofSend of Spatial
= 0; layer) {
3 5 iSizeofLayer = size
if (expansion of layer;
layer)
4 6 iSizeofSend = 0; += iSizeofLayer;
iSizeofSend
5 7 if (expansion layer)
else (base layer)
6 8 iSizeofSend += +=
iSizeofSend iSizeofLayer;
(iSizeofLayer * amount of redundancy)
7 9 elseif(base layer)
(iSizeofSend > iSizeofNet) break;
8 10 } iSizeofSend += (iSizeofLayer * amount of redundancy)
9 11 ifif(iSizeofSend
(iSizeofSend>>iSizeofNet)
iSizeofNet) break;
break;
10 12} }
11 if (iSizeofSend > iSizeofNet) break;
12 } The error-rate calculation, transfer determination algorithm, transfer data error rate, and data
redundancy should be determined. For meaningful MPEG streaming, higher layer data associated
The error-rate calculation, transfer determination algorithm, transfer data error rate, and data
redundancy should be determined. For meaningful MPEG streaming, higher layer data associated
with the current layer should be transmitted first. The proposed algorithm guarantees that this higher
Sustainability 2017, 9, 367 7 of 13

layer data is transferred with a higher reliability. The probability of redundant transfer can be derived
Sustainability 2017, 9, 367 7 of 13
from Equations (2) and (3).
with the current layer should be transmitted first. The proposed algorithm guarantees that this higher
N11
layer data is transferred with a higher reliability. The probability of redundant transfer can be derived
( T1 S1 ) = P111 × P112 × · · · × P11N11 = ∏ P11k (2)
from Equations (2) and (3). k =1

where Pijn is the probability that the nth= packet


× of × ⋯T
the ×i Sj layer=will be transmitted, Nij is the (2)
number
of packets in Ti Sj layer, and ∆P is the size of the packet.
where Pijn is the probability that the nth packet
! of the TiSj layer will be transmitted, Nij is the number
N11
of packets in TiSj layer, and ∆ is the size of the packet.
∏ P11k × Dup > ProBotLmt (3)
k =1
× > (3)
where ProBotLmt is the minimum probability that a layer should be transmitted and DuP is the amount
of redundant transmission.
where ProBotLmt is the minimum probability that a layer should be transmitted and DuP is the
amount of redundant
4. Experimental transmission.
Evaluation and Performance Analysis
4. Experimental
The performance Evaluation and Performance
of the proposed algorithmAnalysis
was analyzed by simulation using a video stream.
The adaptive QoS algorithm’s
The performance performance
of the proposed was was
algorithm compared
analyzedwith a transferusing
by simulation technique
a videothat did not
stream.
consider
The adaptive QoS algorithm’s performance was compared with a transfer technique that did not size
the QoS. The direct and indirect packet losses were computed in terms of the GoV packet
by processing
consider the theQoS.
merged data at
The direct andthe clientpacket
indirect side. The
lossesresult
wereproved
computedthat
in the proposed
terms of the GoVadaptive
packet QoS
size byisprocessing
algorithm better than the merged
the data
current at the client side. The result proved that the proposed adaptive
methods.
QoS algorithm
Figure is better
8 illustrates than the current
a simulation methods. for the performance evaluation of the proposed
environment
Figure 8 illustrates a simulation
algorithms. The streaming server was implemented inenvironment forVisual
the performance
C++ underevaluation
Windowsof the proposed
2000, and the client
algorithms. The streaming server was implemented in Visual C++ under Windows 2000, and the
experimented with a laptop using an Intel Core (TM) i3 CPU. The TS (Transport Stream) Divider is
client experimented with a laptop using an Intel Core (TM) i3 CPU. The TS (Transport Stream)
responsible for dividing; the TS Sender is in charge of sending the divided stream; the TS Writer is
Divider is responsible for dividing; the TS Sender is in charge of sending the divided stream; the TS
in charge of data receiving; and the TS Merger decodes the received data. The Error Rate Controller
Writer is in charge of data receiving; and the TS Merger decodes the received data. The Error Rate
determines the determines
Controller loss of the thepacket
loss sent
of theand the sent
packet Errorand
Inspector
the Errormeasures
Inspector the indirect
measures thedata lossdata
indirect between
the higher layer and lower layer in accordance with the presented error rate.
loss between the higher layer and lower layer in accordance with the presented error rate.

Figure 8. Simulation diagram.


Figure 8. Simulation diagram.
The streaming data used in the experiment is an MPEG-4 stream called Foreman, which is 176 ×
The streaming
144 pixels in size.data usednumber
The total in theofexperiment
frames is 400isframes,
an MPEG-4
and thestream
number called Foreman,
of frames which
per second is is
176 × 144 pixels in size. The total number of frames is 400 frames, and the number of frames per
30 frames. We divide the data into 12 layers through the divider of the server and assume that the
seconddata
is that can be transmitted
30 frames. We dividein each
the time
data unit
into 12islayers
the same. We controlled
through the error
the divider of theofserver
the network in
and assume
between 5% and 50% for comparison of send efficiency.
that the data that can be transmitted in each time unit is the same. We controlled the error of the
network in between 5% and 50% for comparison of send efficiency.
Sustainability 2017, 9, 367 8 of 13
Sustainability 2017, 9, 367 8 of 13

4.1. Performance of the Session-Based Adaptive QoS and Non-QoS Techniques


4.1. Performance of the Session-Based Adaptive QoS and Non-QoS Techniques
The data transmitted using the non-QoS and session-based adaptive QoS techniques was not
The data transmitted using the non-QoS and session-based adaptive QoS techniques was not large
large in size. However, the final merged multimedia data showed big differences because the
in relationship
size. However, the final
between themerged multimedia
layers was data
different. showed
Figure 9a–cbig differences
show because the
the comparison relationship
results. This
between
demonstrates that the lower layer can only be decoded by using higher layer data and that the lossthe
the layers was different. Figure 9a–c show the comparison results. This demonstrates that
lower layerlayer
of higher can only
data be
candecoded
result in by
lossusing higher
of lower layer
layer data.data and that the
To minimize this,loss of higher layer
the session-based data can
method
result in loss of lower layer data. To minimize this, the session-based method transmits
transmits this layer using TCP, resulting in a more reliable transfer as compared to UDP. The transfer this layer using
TCP,
timeresulting in a more
is reduced, however,reliable transfer
reliable as compared
transfer is acquired.to UDP. The transfer
This difference time is
widens asreduced,
the packethowever,
size
reliable transfer
increases. is data
As the acquired.
transferThis
ratedifference
decreases,widens
the dataastransfer
the packet size
failure increases.
possibility Asthe
and theindirect
data transfer
loss
rate decreases,
size increase. the data transfer failure possibility and the indirect loss size increase.

(a)

(b)

(c)
Figure 9. (a) Non-QoS and Session-based adaptive QoS-1; (b) Non-QoS and Session-based adaptive
Figure 9. (a) Non-QoS and Session-based adaptive QoS-1; (b) Non-QoS and Session-based adaptive
QoS-2; (c) Non-QoS and Session-based adaptive QoS-3.
QoS-2; (c) Non-QoS and Session-based adaptive QoS-3.
Sustainability 2017, 9, 367 9 of 13
Sustainability 2017, 9, 367 9 of 13

4.2.4.2.
FEC-Based
FEC-BasedAdaptive
AdaptiveQoS
QoSand
andNon-QoS
Non-QoS Performance Compare
Performance Compare
Figure
Figure10a–c
10a–cshow
showthe
thecomparison
comparison of of results obtained
obtained using
usingthe
theFEC-based
FEC-basedadaptive
adaptive QoS
QoS
algorithm
algorithmand
andthe
the non-QoS technique.The
non-QoS technique. The FEC-based
FEC-based adaptive
adaptive QoS algorithm
QoS algorithm transfertransfer
result is result
lower is
in comparison
lower with with
in comparison non-QoS. This result
non-QoS. is a consequence
This result of the of
is a consequence adaptive QoS algorithm’s
the adaptive flow
QoS algorithm’s
control.
flow control.

(a)

(b)

(c)
Figure 10. (a) FEC method adaptive QoS and non-QoS-1; (b) FEC method adaptive QoS and non-QoS-2; (c)
Figure 10. (a) FEC method adaptive QoS and non-QoS-1; (b) FEC method adaptive QoS and non-QoS-2;
FEC method adaptive QoS and non-QoS-3.
(c) FEC method adaptive QoS and non-QoS-3.
Sustainability 2017, 9, 367 10 of 13

To compare the two techniques, the same amount of data is transferred using both. As the sizes of
all the layers are not equal, we observe, from Figure 10a–c, that the transferred data is not continuous.
The amount of data received using the FEC-based adaptive QoS method is smaller than that received
from the non-QoS technique. Nevertheless, the whole data transfer loss rates are similar because of
the direct loss factor. In the real-time data quantity comparison, the FEC-based adaptive QoS system
shows better results. This is caused by FEC-based redundant retransmission of data. The higher level
data is repeatedly sent, resulting in a lower transfer failure rate as compared to the non-QoS system.
The FEC-adaptive QoS transfer method reduces the indirect loss of the transfer.

4.3. Performance Analysis Result


The performance of the adaptive QoS transfer algorithm and the existing transfer algorithm are
compared. The experimental results show that the proposed method exhibits better data transfer
results. During the transfer of a large multimedia file, the indirect loss of data caused by network
limitations is comparable to the direct loss. It is almost impossible to compensate for direct loss
at the client side. Therefore, data recovery techniques based on the transfer data specification that
reduce indirect losses play a vital role. The proposed method is based on the principle that the higher
layer transmission must be reliable in order to improve overall transmission efficiency. The server
determines the data transfer rate by using the real-time bandwidth information. This results in
an efficient data transfer system. In order to quantify the efficiency of the system, we measured the
error rate in data. The results indicate that the adaptive QoS algorithm system performs better than
the non-QoS technique.

5. Conclusions
Multimedia transmissions have gained research interest due to advances in networking.
The emergence of mobile communication technologies and improved device performance are the
key factors that have motivated studies on multimedia data transfer. This paper proposed two
adaptive QoS transfer algorithms. The first method involved the selection of important data based
on relationships and the other advocated setting of data transmission rates in accordance with the
network bandwidth. When applied to a network that involves multimedia transmission, this adaptive
QoS algorithm can be used to solve different problems. The proposed adaptive QoS data transfer
algorithm can be adapted for use in wired and wireless networks. It can be considered as a service
for server/client-based network structure that guarantees reliable service. In our future research,
we will be analyzing new QoS algorithms and transfer-error prevention methods to address various
heterogeneous network quality issues.

Acknowledgments: This research was supported by the MSIP (Ministry of Science, ICT and Future Planning),
Korea, under the University Information Technology Research Center support program (IITP-2016-R2718-16-0003)
supervised by the IITP (Institute for Information & Communications Technology Promotion).
Author Contributions: All authors developed the research design and contributed to the writing of the paper.
Sang-Hyong Kim and Yoojae Won contributed to the study design, algorithm improvement, and drafted the
manuscript. Kwan-Jong Yoo was involved in data acquisition and analysis, worked on aspects of the experiment
evaluation, and drafted the manuscript. All authors have read and approved the final manuscript.
Conflicts of Interest: The authors declare no conflict of interest.

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