Sci.Int.

(Lahore),26(2),613-620,2014

ISSN 1013-5316; CODEN:SINTE 8

613

PERFORMANCE ENHANCEMENT OF END-TO-END QUALITY OF

SERVICE IN WCDMA WIRELESS NETWORKS
Waseem Abbas1, Nasim Abbas2 and Uzma Majeed3
1

Computer Engineering Department, UET Taxila,

Electronic Engineering Department, MAJU Islamabad,
3
Computer Engineering Department, UET Taxila Pakistan
[1waseem_abbas@live.com, 2nasim_abbas67@yahoo.com, 3uzmamajeed14@gmail.com]
2

ABSTRACT: In this paper we proposed user satisfaction architecture for packed-switched services in 3rd
generation cellular networks for end-to-end Quality of Service (QoS) provisioning in Diffserv IP network
environment. The paper focuses on mapping of QoS classes from Diffserv to UMTS, Admission control,
Buffering and scheduling. The Diffserv code point was utilized in the end-to-end quality of service
provisioning to differentiate various type of multimedia real time traffic. This paper proposes the
WCDMA based prioritized uplink call admission control that combines the QoS tolerance and service
differentiation for data and multimedia traffic by priority. This algorithm reserves some bandwidth
margin, number of users and power consumption to decrease handoff failures to give preference to high
priority calls, such as handoff calls additionally Low latency queuing (LLQ) is implemented to improve
the quality of service. LLQ is used with the key idea of mapping voice and video traffic in two different
queues but at the same time using priority queuing within LLQ for both voice and video traffic for all
other QoS classes. The results obtained from simulations demonstrate that proposed algorithm meet the
QoS requirement.
Keywords: CAC, DiffServ, EURANE, LLQ, NS- 2, QoS, Scheduling

1. INTRODUCTION
Third generation mobile communication technologies have
gone through a very rapid growth and Universal Mobile
Telecommunication System (UMTS) has emerged as a
leading standard for the provisioning of third generation
cellular networks. Wireless 3G network is anticipated to
convey multimedia traffic such as VOIP, video telephony,
data and other applications. The requirement of QoS while
transmitting multimedia traffic on same medium is one of
the major issues in order to design and analyze 3G wireless
networks. Service classification and efficient resource
management are quite demanding tasks due to increasing
number of applications and ubiquitous bandwidth
limitations for multimedia applications such as voice and
video streaming.
Key advantage of UMTS is its ability to provide different
services with QoS guarantees [1]. IETF standardized two
different mechanisms for providing QoS in IP networks
Diffserv [2] and InterServ [3]. DiffServ can be
implemented in UMTS with no management complexity,
while InterServ have scalability and complexity problems.
DiffServ uses Codepoints known as Differentiated Services
code points (DSCP) attached with IP header of a packet to
classify traffic with different PHBs (Per Hop Behavior) at
the boundaries of the network. PHB definitions do not
specify any particular implementation mechanism and
therefore implementation problem of PHB has gained
noteworthy attention. On the other hand, QoS concept and
architecture for UMTS network as specified in 3GPP [1]
focuses on QoS signaling from user equipment to GGSN
only, and it does not support the QoS mechanisms for data
transport. For end-to-end QoS provisioning in UMTS, it is
required to map the IP traffic classes to the UMTS network

and propose a QoS mechanism for the transport of user

data.
Existing work on end-to-end QoS in UMTS [4] have
analyzed advanced mapping between voice and video
telephony but do not take other UMTS traffic classes into
account. Other work investigates the access control (AC),
in both the wired and wireless network part, a multi-class
AC scheme on the wired network part was proposed in [5],
which supports the DiffServ approach. Authors in paper
[6] have combine WFQ and LLQ, but the main flaw of this
idea is the property that delay in video conferencing could
be reduce but at the same time voice traffic got the
maximum delay time. The SEACORN [7] project
contributed in the development and implementation of
wireless part i-e. Radio Resource management RRM
algorithms for QoS provisioning in the UMTS network.
One of the major contributions of the SEACORN project is
a UMTS extension for the Network Simulator 2 (NS-2) [8]
known as Enhanced UMTS Radio Access Network
Extension (EURANE) [9]. We choose EURANE as the
tool to simulate an E2E UMTS QoS scenario.
This paper proposes detailed algorithm in which concept
of mapping voice and video telephony to different QoS
classes, idea of implementing LLQ scheduler and packet
treatment strategies for the UMTS core network and
analyzes them in a large set of simulation experiments.
The latter focus on revealing the impact of non-real time
services on real-time services in different scenarios with
and without full QoS provisioning mechanisms. The
mapping will be done in the GGSN and scheduling,
policing and multiple queuing mechanisms will be
implemented in the UMTS Core Network.

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This paper is structured as follows. Section II presents the

proposed algorithms and mechanism, Section III explains
the simulation scenario and simulation results. Section IV
concludes the paper and describes direction of future work.
2. END-TO-END QOS MECHANISM
The aim of this section is to give the brief idea of
development of E2E QoS algorithm in terms of QoS
mapping, Call Admission Control, policing, buffering and
scheduling.
2.1 General Assumptions and optimization target
In our proposed algorithm which is depicted in Figure 1
for provisioning of E2E QoS algorithms. There are three
main components of UMTS System architecture User
Equipment UE or Terminal Equipment TE, UMTS
Terrestrial Radio Access Network (UTRAN) and Core
Network. Core network is connected to external IP domain
network in which there are edge router and application
servers which can send only one type of application data.
GGSN and SGSN are the entities of UMTS core network
and UTRAN contains Radio Network controllers RNCs
and Node-Bs, which are responsible for all functionality
related to radio access. In our scenario both external
network and UMTS network supports DiffServ.
Application Servers send downlink data packets to the UEs
and these packets will be controlled by Edge router, GGSN
and SGSN before it is segmented to RLC PDUs in RNC
and forwarded to Node B and in our proposed algorithm
only PDUs transmitted via Dedicated Transport Channel to
UEs are considered.
UMTS CORE
NETWORK

UTRAN

EXTERNAL
NETWORK
VOIP

UE
UE

UE

VIDEO

Node-B

RNC

SGSN

GGSN

EDGE

UE

Sci.Int.(Lahore),26(2),613-620,2014

In our scenario, the bottleneck of the downlink

transmissions is presumed to be the outgoing link from the
GGSN to SGSN hence the design target is to enhance the
bottleneck link consumption i-e minimizing session
blocking rate and higher link throughput while
maintaining the E2E QoS requirements of each UMTS
class. The boundary conditions of each service type are
shown in Table 2.
Table 2: UMTS QoS Requirements of each service class
Service Conversational Streaming
Interactive BackType
ground
SDU
Loss
< 10-2
<10-1
<10-3
<10-3
Rate
End-toEnd
<100 ms
<250ms
N/A
N/A
Delay

2.2 DiffServ to UMTS QoS Mapping

The 3GPP standard defines a layered architecture for the
provisioning of end-to-end QoS. To understand QoS for a
certain network, a Bearer Service (BS) with appropriately
given functionalities has to be implemented from the
source to the destination of a service and contains all
characteristics to facilitate provisioning of a constricted
QoS. UMTS bearer service attributes from a QoS profile
and defines the level of service provided to the UMTS BS
user.
UMTS specification [10] defines four QoS classes in
UMTS packet domain: Conversational, Streaming,
Interactive, and background. These classes are categorized
on the basis of Delay factor The streaming and
conversational classes are the most delay sensitive classes
and defined for real time applications, while interactive
and background classes are meant for delay-insensitive
classes and treat the traffic without restricted transfer delay
requirements, e-g Transmission control Protocol.

WEB

Table 3: UMTS vs. DiffServ Mapping

FTP

Figure: 1 End-to-End UMTS QoS Scenario

The basic assumptions on the traffic model are given in

Table 1 and sessions of the individual types of traffic arrive
according to Poisson Processes.
Table 1: Traffic Model assumptions
Traffic Type
Application
Application
Holding
Level
Time Model
Traffic
Model
Conversational
Voice
EXP On/Off
Exponential
Streaming
Video
EXP On/Off
Exponential
Streaming
Interactive
Web
Pareto
Log-Normal
On/Off
Background
FTP
CBR
Pareto

Application
Type
VOIP
Video
Streaming
Web HTTP

UMTS Service
Class
Conversational
Streaming

FTP

Background

Interactive

DiffServ
DSCP
EF
AF11
AF 12
AF21
AF22
BE

DSCP
Value
46
10
12
18
20
0

QoS mapping between IP DiffServ and UMTS services as

described in Table 3 is very essential for keeping
appropriate end-to-end delay for real time traffic.
The Expedited Forwarding Per Hop Behavior (EF-PHB)
[11] is known for low packet loss, low jitter and less delay
services. In case of EF traffic is treated at minimum service
rate for both short and long intervals. The EF packets that

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go beyond the particular arrival rate will be dropped in

advance. This DiffServ class is very suitable with the
traffic of conversational class as traffic of conversational
class i-e VOIP is very delay sensitive but comparatively
insensitive of packet loss. IETF specifies Assured
Forwarding (AF) PHB group in [12].
AF-PHB gives guarantee of delivery provided that traffic
will not surpass subscribe rate. In case of congestion traffic
that goes beyond the particular rate has high probability of
being dropped.
Four different independent classes with three different
dropping precedence levels each were implemented in AFPHB for IP packet delivery. The corresponding PHB is
called AFij, in which i represent AF class and j shows
dropped precedence. Every class is implemented with
different shares of bandwidth with different configurations
of buffering and dropping. The AF-PHB class supports
streaming class as it is delay sensitive but less than
conversational class. The Interactive class is mapped to
AF31, with no explicit delay boundary and do not have
extra requirements apart from reliability and in last
Background class of UMTS network is mapped with
default PHB class i-e Best Effort
2.3 Call Admission Control
The main purpose of CAC is to restrict the interference by
constraining the capacity of new calls admitted in the
network, as traffic load offered by uplink and downlink
transmissions is different so CAC need to perform
separately for each other. The preconditions of uplink and
downlink CAC must be attained by each new user while
entering into the system. Due to mobility in wireless
networks CAC turn out to be more complicated.
The call dropping happens in the network in most of the
case during handoff in which the UE is moving away from
the covered area of one cell and entering in the region
covered by another cell. During the process of call handoff
there is a possibility of not getting enough resources in the
next cell due to limited resources in wireless networks
[13]. Thus there should be a priority base mechanism for
treating new calls and handoff calls in terms of resource
allocation. Normally higher priority is given to handoff
calls than new calls [14]. Handoff priority-based CAC
schemes can be classified into two broad categories NCAC
(Number based) and ICAC (Interference based).
NCAC is the number based call admission control that
considers limited number of channels and the QoS
parameter bit-error-to-interface ratio for NCAC is mapped
to the number of users contained in the network while
ICAC algorithm does not limit the number of channels and
it uses SIR parameter to accept or reject the newly
originated call. ICAC have been further classified in three
schemes [15].
Wideband Power Based CAC, Throughput Based CAC and
Signal to Noise Based CAC.
2.3.1 Principle

615

Figure 2 shows flow chart for handoff prioritization

algorithm which is basic logic.

i. Estimate load factor threshold thresh (in term of No. of

user, BW, Power consumption) [16].
ii. Calculate the load increase of incoming call i along

with the load factor of current cell , prior to accept

incoming call [17].
iii. Calculate the up to date load of target cell and compare
it with threshold of load factor for newly originated call.
The new call seeking admission will only be admitted in
target cell when sum of load increase and current cell load
factor is equal to or less than threshold of desired load
factor for incoming call or else the incoming call will be
rejected or queued. Soft handoff calls which are in queue
can be admitted if enough bandwidth is available [21]
otherwise rejected because of timeout.
Initialize Network parameters
New Call Seeking Admission

Determine thresh and QoS requirements

Calculate i and

Yes

i + < thresh

Admit Call

No

Reject New Call

No

Check Queue
Availability
Yes

Call Queued

Terminate Call

No

If sufficient Bandwidth
available before
timeout
Yes

Admit Call
Update Network Parameters
End

Figure: 2 Flow chart of Handoff Prioritization algorithm

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2.4 Policing, buffering and scheduling.

Policing, buffering and scheduling algorithms deals with
packets, these mechanisms are executed each time a packet
is sent or received. Policing and buffering algorithms
decide whether or not an incoming packet is accepted. A
full implementation structure of these algorithms is
depicted in figure 3.
DiffServ Code Point DSCP marking is implemented by a
policer at the edge router and also two code points (red and
green) are assigned to each packet, it means that IP
packets with different color can be handled later with
different physical/virtual queues and different treatment.
Edge
Router

GGSN
SGSN

Application
Server

EF
Scheduler

AF11-12

Meter/
Marker

IP Packets

RED AF2X
RED

BE

Figure: 3 policing buffering and scheduling algorithm

implementation

Multiple queues have been designed to provide different

treatment to different UMTS classes according to their
DSCP field. Since Random Early Detection (RED) [18]
queues only impact TCP based traffic streams and cannot
help in congestion control of UDP based traffic, two droptail queues are designed for UDP based real time services,
Conversational (C) and Streaming (S), while RED queues
are for TCP based non-real time services, Interactive (I)
and Background (B). The queue size was optimized
according to their different delay and packet loss
requirements simultaneously. Preliminary simulations
show that in the RED queue design, multiple virtual
queues did not improve the QoS for non real time traffic,
i.e., the green packets of Web/Http traffic actually
experienced higher packet dropping than the red
packets. Hence the coloring was not use in our QoS
concept.
Scheduling improves the capacity of system by sharing the
common recourses dynamically among different classes.
The main responsibility of packet scheduling algorithms is
to assign available wireless network resources to group of
mobile users who are allowed to receive data. When a link
is congested there is a need of scheduler and queuing
mechanisms are busy, though Scheduler has to decide for
the next queue to treat first. If there is no congestion on
link then packets are sent as they arrive.
It is obvious that each UMTS service class should be
assigned a priority levels from 1 to 4 due to the different
delay sensitivity levels in which 1 is the highest. We
choose Low Latency Queuing LLQ because the relative
importance of each class can be modified easily according
to change of traffic mix, hence it is more flexible. LLQ
scheduling is an enhancement of Class Based Weighted

Sci.Int.(Lahore),26(2),613-620,2014

Fair Queuing CBWFQ in which one or more strict priority

queues are integrated to solve latency issues in multimedia
applications, that makes it ideal for jitter and delay
sensitive applications. LLQ have properties of both PQ and
CBWFQ. LLQ queues delay sensitive traffic and allow it
to be dequeued and if there are no more packets in that
queue then WFQ executes the rest of the traffic and the
normal scheduler logic applies to other non latency queues,
giving them their guaranteed bandwidth. The main
advantage of LLQ over the classic PQ is that the LLQ
strict priority queue is policed which minimize the chances
of starvation of other queues.

3. EXPERIMENTAL RESULTS
In this paper, we simulated our proposed scheduling
algorithm and CAC algorithm in UMTS cellular networks
for different transmission scenarios. We have used NS-2 as
our simulation tool which is a discrete event-driven tool
[7] and independent developed module for user defined
networks. The proposed CAC algorithm is investigated
based on three parameters: probability of blocking of calls

requesting for new admission, probability of

termination of calls before completion forcibly and
overall system carried traffic. Overall Grade of Service to
calculate performance of the network is defined as [19]:
GoSj = *Phj + Pnj
(1)
Where Ph is probability of the blocking of handoff calls, Pn
is probability of the blocking of new calls and priority level
of handoff over new calls are shown in terms of . The
system gives good performance when GoS is small. The
system capacity is investigated by evaluating total carried
traffic which is defined as [20]:

Figure 4: GoS for Voice Calls

CT = h1 (1-Ph1) + h2 (1-Ph2) + n1 (1-Pn1) + n2,j (1-Pn2)

(2)
Where h1, h2, shows the rates for voice and data handoff
calls and n1, n2, j shows the rates for voice and data calls
seeking new admission respectively.

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3.1 CAC Simulation Results

In our proposed algorithm higher priority is assigned to
handoff calls over newly arrived calls. Following three
scenarios are investigated in our simulations: (1) all calls
should be equally treated when there is alike load threshold
and in this scenario queuing techniques are not
implemented, (2) handoff calls are queued till the
availability of resource or time out reach, (3) over and
above second case higher load threshold for handoff than
newly arrived calls (proposed). This mechanism is
evaluated with disparate channel holding times. In our
simulation general service time is 180 sec and each service
has different arriving rate. Different service times (180,
120, 90) are used in case of third scenario. Figure 4 shows
GoS of voice calls vs. arrival rates, Figure 5 shows GoS of
Data calls vs. arrival rates.
It can be clearly seen from these graphs that performance
is improved by using queue and soft guard channel.
In case of increase in mobility there is decrease in channel
holding time, overall system performance will be improved
and in case of decrease in service time the call in queue
waiting for getting channel have more chance to get
admitted in the system before time out reach. Figure 6
depicts the comparison of system carried traffic and total
offered traffic which clearly shows that in third case
overall system capacity increases and it will be enhanced
more when channel holding time decreases.

617

the better bottleneck link bandwidth utilization while

keeping the end-to-end delay and IP data Packet Loss
within their boundary as shown in table 2. So the Key
Performance indicators are end-to-end delay, IP packet
loss, bottleneck link bandwidth utilization. Simulation
scenario is shown in Figure 7 in which there are four
application servers located in external network. The IP
data Packets are sent to edge router, which is simplified

as both the Ingress and Egress router to the external

DiffServ domain, here IP data packets are marked with
DSCP according to their application type and forward to
GGSN. The GGSN outlink differentiates each IP data flow
according its DSCP (the GGSN does not change the DSCP
assigned from the external network) and transmits them
with the queuing and scheduling schemes described in the
previous chapter. The SGSN receives these packets and
forwards them to the RNC, at which the IP packets are
converted into RLC SDUs.
Multiple Queuing
Scheduling

Ingress/Egress Router
DSCP Marking
VOIP

UE
UE

UE

VIDEO

Node-B

RNC

SGSN

GGSN

EDGE

UE
Bottleneck Link

WEB

FTP

Figure 7: Implementation of End-to-End QoS provisioning

algorithms

Figure 5: GoS for Data Calls

Figure 6: System Carried Traffic

3.2 LLQ Scheduling Simulation Results

In this section overall end-to-end QoS investigated
scenarios and experiments are discussed in which all the
QoS mechanisms which are designed in last sections are
applied. Our main target in this simulation is to achieve

The radio link settings are shown in Table4. The DCH

works in Acknowledge Mode, and the maximum RLC
layer retransmission time is unlimited. This setting is due
to the EURANE [9] limitations. Hence all the erroneous
RLC PDUs will be recovered, and the E2E SDU loss will
only be caused by IP packet dropping in the bottleneck link
queue, which is Early Drop for real-time traffic or queue
overflow for all other traffic. On the other hand the
unlimited RLC PDU retransmission will result in much
more uncertainty for the E2E delay, and make the delay
control more difficult.
Table 4: Network Parameters
Wired Part
Link Bandwidth Link Delay (ms)
(Mbps)
Server-Edge router
10
20
Edge router-GGSN
10
2
GGSN-SGSN
1
2
SGSN-RNC
10
2
RNC-Node B
10
5
Wirleless
Simulated cell
1
Number
Active UE number
20
DCH bandwidth
384Kbps
Fast Power Control
Ideal
Radio link RLC PDU
Uniform, mean=0.01
error
Mobility model
No

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Sci.Int.(Lahore),26(2),613-620,2014

Theses parameters are investigated using different

simulation experiments in which different scheduling
algorithms WRR, PQ and LLQ are implemented and
evaluated for different scenarios of link congestion. In
start link is overloaded with 10% traffic and gradually
increases to 150 %. Traffic parameters are given in Table5.
Conversational class produces 20 % of overall generated
traffic, streaming class produce 70 %, interactive class 3%
and background class produce 7 % of overall generated
traffic. In case of PQ scheduler, voice has given highest
priority, while background class has the least priority. In
case. In other schedulers weight represent output port
bandwidth percentage i-e Weight 20, weight 70, weight 3,
weight 7 for conversational, streaming, interactive and
background respectively.
Table 5: Average throughput of NRT Traffic
Voice
Video
HTTP
FTP
Traffic
Traffic
Traffic
Traffic
Transport
UDP
UDP
TCP
TCP
layer Protocol
Size of IP
120
160
240
480
Packet
Traffic Source
Exp
Exp
Pareto
Pareto
model
on/off
on/off
on/off
on/off
Holding time
Exp
Exp
LogPareto
distribution
normal
Distributed
Bottleneck
1000
Bandwidth
BW(Kbps)

It is clear from the Figure 8 that if PQ and LLQ are

implemented average end-to-end delay for conversational
class remains below 100ms even in case of high
congestion. The average end-to-end delay of video
streaming is shown in Figure 9. All schedulers have almost
same performance and provide satisfactory level of QoS.
Figure 10 depicts average jitter, and in all experiments for
conversational class jitter stays within 50ms and same
results are observed in case of video streaming as depicted
in Figure 11. Figure 12 depicts packet loss rate for
conversational class, it is clear that link congestion has
great effect on WRR scheduler performance while packet
loss rate stay within boundary conditions in case of PQ and
LLQ. Similarly packet loss in streaming class stays below
2x10-3 for PQ and LLQ but exceed in case of WRR.
Average throughput for interactive and background class is
given in Table 6. In case of PQ in high link congestion
both classes have least throughput. Whereas LLQ provides
fair level of bandwidth and gives the best results for
interactive and background classes even in case of high
link congestion.
Table 6: Average throughput of NRT Traffic
Traffic Class
Throughput (Kbps)
WRR
PQ
LLQ
Interactive Class
8.57
4.31
8.24
Background Class
16.21
10.04
17.23

Figure 8: Average End to End Delay of Conversational Class

Traffic

Figure 9:Average End to End Delay of Streaming Class Traffic

Figure 10: Average End to End Jitter of Conversational Class

Traffic

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619

simulation results, we conclude that this algorithm reduces

the dropped handoff calls by handoff prioritization and
shows adequate performance with respect to end-to-end
delay, packet loss ratio and achieves good bottleneck link
utilization with in all QoS limitations and at the same time
gives the fair service even in case of highly congested links
to all traffic classes and enhance the overall system
capacity. Future research work will focus on enhancing the
network capacity with convolution code with QoS for real
time application users by selecting suitable value of BER.

Figure 11: Average End to End Jitter of Streaming Class Traffic

Figure 12: Packet Loss of Conversational Class Traffic

Figure 13: Packet Loss of Streaming Class Traffic

4. CONCLUSIONS
This paper focuses on investigating provisioning of QoS
for real-time UMTS traffic as well as to improve the
system performance. The enhanced algorithm combines
Diffserv-UMTS QoS mapping, WCDMA based prioritized
uplink call admission control, multiple queuing and LLQ
scheduling with optimized parameters. The performance of
this CAC with multiple queuing and LLQ scheduling is
investigated with various scenarios. On the basis of

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