CSFB VolTE-Thesis Report

Voice over LTE
Master of Science Thesis
PRASANNA GURURAJ
RAGHAVENDRARAO
Wireless and Mobile Communications Group

Department of Telecommunications
Faculty of Electrical Engineering, Mathematics and
Computer Science
Delft University of Technology
II
Prasanna Gururaj
Raghavendrarao
III
Voice over LTE

For the degree of Master of Science in

Wireless and Mobile Communications Group (WMC)
at Department of Telecommunications
at Delft University of Technology
Prasanna Gururaj
Raghavendrarao
29.6.2012
Faculty of Electrical Engineering, Mathematics and Computer Science

Delft, The Netherlands
Prasanna Gururaj
Raghavendrarao
IV
Prasanna Gururaj
Raghavendrarao

Department of
Telecommunications
The undersigned hereby certify that they have read and recommend to the Faculty of
Electrical Engineering, Mathematics and Computer Science for acceptance a thesis
entitled
Voice over LTE
by
Prasanna Gururaj
Raghavendrarao
in partial fulfillment of the requirements for the degree of
Master of Science.
Dated: 29.6.2012
Supervisors:
dr.ir. Jos Adema (KPN)
ir. Gerard Fossung (KPN)
dr. R.R. Venkatesha Prasad

Readers:
dr.ir.Jos Weber
dr.ir. Bert Jan Kooij
Abstract
Long Term Evolution (LTE) is the latest high speed mobile broadband technology that
is gaining widespread attention due to its high data rates and improved Quality of
Service (QoS). Initially, LTE was seen as a technology for supporting high speed data,
but there is a growing interest in the industry to support voice over LTE. The support
of voice over LTE has lot of challenges owing to the fact that both voice and data traffic
are to be carried over the same radio and core networks. The optimum usage of resources in the radio network is of high importance as there is a growing need to improve
the capacity at reduced cost. The transport network is another key area that needs
to be carefully planned according to the capacity of the radio network. Differentiation
and scheduling of resources in the transport network plays a key role in guaranteeing
good end to end performance for both voice and data services.
In this thesis, the impact of differentiation and scheduling of resources in the transport network on the end to end performance of voice over LTE is investigated. The
results indicate that without proper prioritization and scheduling of resources in the
transport network , the performance of voice is severely affected when the transport
network is congested with data traffic. To overcome this scenario, we prioritize voice
over data traffic and analyse its performance for different transport network scheduling
algorithms. From the results, it is clear that with proper classification and scheduling of
resources in the transport network, significant increase in voice capacity is observed. On
the other hand, by totally prioritizing voice, performance of the data traffic is affected
to a large extent. Hence, to achieve a balance, voice users are classified into different
priority levels and the performance of voice and data in this scenario is investigated.
The analysis for all these scenarios are based on simulations using OPNET simulation
tool.
Prasanna Gururaj
Raghavendrarao
ii
Prasanna Gururaj
Raghavendrarao
Table of Contents
Acknowledgements
xi
1 Introduction
1-1 Solutions for Supporting Voice over LTE . . . . . . . . . . . . . . . . . .
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1
1-1-1
Circuit Switch(CS) fallback . . . . . . . . . . . . . . . . . . . . .
1-1-2
Voice over LTE via IP Multimedia Subsystem (VoLTE) . . . . . . .
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2-6-2
Weighted Round Robin (WRR) Scheduling . . . . . . . . . . . . .
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2-6-3
Weighted Fair Scheduling . . . . . . . . . . . . . . . . . . . . . .
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3 Simulation Model
3-1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3-2 Introduction to OPNET Modeller . . . . . . . . . . . . . . . . . . . . . .
3-2-1 Overview of LTE Model in OPNET . . . . . . . . . . . . . . . . .
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1-2
1-3
1-4
1-5
Motivation for the Thesis

Problem Definition . . . .
Related Work . . . . . .
Organization of the Thesis
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2 Background
2-1 Introduction . . . . . . . . . . . .
2-2 LTE Network Architecture . . . . .
2-3 QoS Architecture in LTE . . . . . .
2-4 IMS Network Architecture . . . . .
2-5 Differentiated Services Architecture
2-6 Scheduling Strategies . . . . . . .
2-6-1 Strict Priority Scheduling .
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Prasanna Gururaj
Raghavendrarao
iv
Table of Contents
3-2-2 Issues in LTE model . . . . . . . . .

3-3 Changes in LTE model . . . . . . . . . . . .
3-3-1 LTE S1 process model In E-Node B .
3-3-2 LTE S1 NAS Process model in EPC .
3-3-3 GTP Process model in E-Node B and
3-4 Simulation Environment . . . . . . . . . . .
3-4-1 Mobile Node . . . . . . . . . . . . .
3-4-2 E-Node B . . . . . . . . . . . . . .
3-4-3 IMS Model . . . . . . . . . . . . . .
3-4-4 Application Configuration . . . . . .
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EPC
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4 Results
4-1 Introduction . . . . . . . . . . . . . . . . . .
4-2 QoS parameters for Voice . . . . . . . . . . .
4-3 Scenario 1 . . . . . . . . . . . . . . . . . . .
4-3-1 Description of the Scenario . . . . . .
4-3-2 Analysis of results . . . . . . . . . . .
4-4 Scenario 2 . . . . . . . . . . . . . . . . . . .
4-4-3 Impact on FTP traffic . . . . . . . . .
4-4-4 Summary of the Results for Scenario 2
4-5 Scenario 3 . . . . . . . . . . . . . . . . . . .
4-5-3 Impact on FTP traffic . . . . . . . . .
4-5-4 Summary of results for Scenario 3 . . .
4-6 Scenario 4 . . . . . . . . . . . . . . . . . . .
4-6-1 Description of the scenario . . . . . . .
4-6-2 Analysis of Results . . . . . . . . . . .
4-7 Comparison of Scenarios . . . . . . . . . . . .
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5 Technical Details of Voice over LTE via IMS based solution - An Operator
Perspective
5-1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
5-2 VoLTE Architecture . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
5-3 Options for integrating LTE with existing CS/PS networks . . . . . . . . .
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5-3-1
Independent PS based solution . . . . . . . . . . . . . . . . . . .
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5-3-2
Enhanced Single Radio Voice call continuity (SRVCC) / IMS Centralized Services (ICS) . . . . . . . . . . . . . . . . . . . . . . . .
53
Prasanna Gururaj
Raghavendrarao
Table of Contents
6 Conclusion
6-1 Conclusion . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
6-2 Future Work . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
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Glossary
List of Acronyms . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
List of Symbols . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
59
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Prasanna Gururaj
Raghavendrarao
vi
Prasanna Gururaj
Raghavendrarao
Table of Contents
List of Figures
1-1 Circuit Switch Fallback [2] . . . . . . . . . . . . . . . . . . . . . . . . . .
1-2 VoLTE [2] . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
2-1 LTE Network Architecture . . . . . . . . . . . . . . . . . . . . . . . . . .

2-2 QoS Architecture in LTE [8] . . . . . . . . . . . . . . . . . . . . . . . . .
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2-3 IMS Network Architecture . . . . . . . . . . . . . . . . . . . . . . . . . .
11
3-1 LTE Network Model . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

3-2 Data Flow in LTE Network [13] . . . . . . . . . . . . . . . . . . . . . . .
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3-3
3-4
3-5
3-6
3-7
3-8
3-9
3-10
3-11
3-12
Protocol Architecture . . . . . . . . . . . . . . . . . . .
GTP Encapsulated Packet . . . . . . . . . . . . . . . . .
E-Node B Node Model . . . . . . . . . . . . . . . . . .
lte _ s1 Process model in E-Node B . . . . . . . . . . .
Node Model in EPC . . . . . . . . . . . . . . . . . . . .
lte _ s1 _ nas Process model in EPC . . . . . . . . . . .
GTP Process Model . . . . . . . . . . . . . . . . . . . .
LTE Simulation Network . . . . . . . . . . . . . . . . . .
Mobile Configuration . . . . . . . . . . . . . . . . . . . .
IMS Proxy session control function configuration attribute
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4-1
4-2
4-3
4-4
4-5
4-6
Voice Packet End to End Delay vs No. of VoIP

Packet Delay Variation Vs No. of VoIP users .
End to End Delay Vs No. of VoIP Users . . .
S1 Delay Vs No. of VoIP Users . . . . . . . .
PDV vs No. of VoIP users . . . . . . . . . . .
Packet Loss Rate vs No. of VoIP users . . . .
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users
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Prasanna Gururaj
Raghavendrarao
viii
List of Figures
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5-1 VoLTE Architecture . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

5-2 LTE-3G Integrated architecture . . . . . . . . . . . . . . . . . . . . . . .
50
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5-3 SRVCC/ICS Architecture . . . . . . . . . . . . . . . . . . . . . . . . . .
53
4-7
4-8
4-9
4-10
4-11
4-12
4-13
4-14
4-15
4-16
4-17
4-18
4-19
4-20
4-21
4-22
Mean Opinion Score vs No. of VoIP users . . . . . . .

FTP Transfer time vs No. of VoIP users . . . . . . .
FTP Throughput vs No. of VoIP users . . . . . . . .
End to End Delay vs No. of VoIP users . . . . . . . .
S1 delay vs No. of VoIP users . . . . . . . . . . . . .
Packet delay variation vs No. of VoIP users . . . . . .
Packet Loss Rate vs No. of VoIP users . . . . . . . .
Mean Opinion Score vs No. of VoIP users . . . . . . .
Mean FTP transfer time vs No. of VoIP users . . . .
FTP Throughput vs No. of VoIP users . . . . . . . .
Packet end to end delay for high priority VoIP users .
Packet end to end delay for normal priority VoIP users
Packet delay variation . . . . . . . . . . . . . . . . .
Packet Loss Rate . . . . . . . . . . . . . . . . . . . .
Mean Opinion Score . . . . . . . . . . . . . . . . . .
FTP Transfer time . . . . . . . . . . . . . . . . . . .
Prasanna Gururaj
Raghavendrarao
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List of Tables
2-1 EPS QOS Bearer Definitions [9] . . . . . . . . . . . . . . . . . . . . . . .
10
2-2 Assured Forwarding Drop Precedence Classification . . . . . . . . . . . . .
13
3-1 EPS Bearer to DSCP Mapping . . . . . . . . . . . . . . . . . . . . . . .

3-2 E-Node B Configuration Parameters . . . . . . . . . . . . . . . . . . . . .
3-3 VoIP Configuration Parameters . . . . . . . . . . . . . . . . . . . . . . .
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4-1 MOS satisfaction level . . . . . . . . . . . . . . . . . . . . . . . . . . . .

4-2 Scenario Description . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
4-3 Number of satisfied VoIP users . . . . . . . . . . . . . . . . . . . . . . .
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48
5-1 VoLTE Relevant Interfaces and Protocols . . . . . . . . . . . . . . . . . .
50
Prasanna Gururaj
Raghavendrarao
Prasanna Gururaj
Raghavendrarao
List of Tables
Acknowledgements
First, I would like to thanks my supervisors at KPN Dr.ir. Jos Adema and ir. Gerrard
Fossung for their valuable inputs and suggestions during the writing of this thesis. I
would also like to thank Perry Jackson for giving me an opportunity to do this thesis at
KPN and the Mobile Innovation Voice Team for their kind support and encouragement
during the course of this thesis. Next, I would like to thank my supervisor Dr. R.R.
Venkatesha Prasad for his invaluable support and constant encouragement during the
writing of this thesis. Last but not the least, I would like to thank my family and
friends for their continuous love and support.
Delft
29.6.2012
Prasanna Gururaj Raghavendrarao
Prasanna Gururaj
Raghavendrarao
xii
Prasanna Gururaj
Raghavendrarao
Acknowledgements
Chapter 1
Introduction
Long Term Evolution (LTE) is a fourth generation technology which is standardized

in the Release 8 specifications by the 3GPP. It is capable of providing high data rates
(100 Mbps in downlink and 50 Mbps in uplink) as well as support high speed mobility.
It has a completely packet switched core network architecture unlike its predecessor
UMTS which is capable of supporting both the Circuit Switched (CS) as well as Packet
Switched (PS) core networks.
1-1
Solutions for Supporting Voice over LTE
The absence of CS domain in the LTE network has led the industry and standardization
bodies like the 3GPP to propose various solutions to support voice in the LTE network.
The two most important among them widely being considered for deployment are as
follows:
1-1-1
Circuit Switch(CS) fallback
The Circuit Switch fallback solution defined in [1], provides a convenient way in reusing
the existing GSM/UMTS network to support voice in LTE network. This solution is
standardized in [1] and provides the operators with flexibility to roll out LTE as a data
only overlay network and use the existing CS network for supporting voice functionality.
The network architecture of CS fallback is shown in the Figure 1.1.
The user performs a combined registration with both the LTE as well as GSM/UMTS
network during the initial registration procedure. This combined registration is facilitated by the Mobility Management Entity(MME) in the LTE network which performs
the registration in the 2G/3G network on behalf of the user. During the initiation of
Prasanna Gururaj
Raghavendrarao
Introduction
Figure 1-1: Circuit Switch Fallback [2]
the voice call by the user, the MME redirects the request towards the MSC server in
the CS domain. On successful reservation of the resources in the CS domain for the
call, the MSC server shall respond to the MME on the status of the request. The
MME then instructs the E-Node B to request the user to perform a handover to the
GSM/UMTS network. The ongoing data session for the user in the LTE network is
suspended if the destination network is a GSM network. If the destination network is
an UMTS network, then a separate handover of the existing data bearers from LTE to
UMTS network takes place after registration by the user in the UMTS network.
This solution has several disadvantages like increase in call set up time due to the
handover procedure and disruption of data transmission throughout the duration of the
voice call when the user falls back to a GSM network. This solution can be used during
the initial roll out when LTE is more used for high speed data and voice is completely
handled by legacy circuit switched networks. Hence CS fallback is being seen only as
a temporary solution during the initial roll out of the LTE network.
1-1-2
Voice over LTE via IP Multimedia Subsystem (VoLTE)
In this solution, voice functionality is provided by the IP Multimedia Subsystem (IMS).

IMS is a core network architecture that is integrated on top of the LTE network as shown
in Figure 1.2. The IMS network is mainly used to provide all the basic services for voice
Prasanna Gururaj
Raghavendrarao
1-2 Motivation for the Thesis
that are provided by the existing CS networks. In addition, it also provides enhanced
multimedia services like video conference, real time gaming etc. The main advantage of
Figure 1-2: VoLTE [2]
using an IMS based solution is that it completely utilizes the LTE architecture rather
than relying on the existing CS networks for supporting voice feature. The IMS network
is also capable of integrating with the legacy 2G/3G networks and thus can support
voice call continuity even when the subscriber moves out of LTE coverage. Hence, the
subscriber can experience the same services even when roaming into legacy networks.
This solution is being projected as the long term solution as it is capable of providing
enhanced services to the LTE network and also supports integration with the existing
2G/3G networks.
1-2
Motivation for the Thesis
The VoLTE solution mentioned in the above section is widely being considered for
deployment by operators across the world as it provides simultaneous support of both
voice and data in the LTE network. In VoLTE, voice is carried in the LTE network as
Voice over IP (VoIP) packets. Hence the VoLTE architecture is significantly different
from the 2G/3G networks which have distinct CS capabilities for voice. IP based
networks are mainly designed for best effort services which do not provide any strict
guarantees on the quality of service demands of the various services that are offered
to the users. In legacy networks like GSM/UMTS, IP based networks were mainly
used for carrying data services like FTP, HTTP etc. However, with the growth of
mobile broadband technologies like High Speed Packet Access (HSPA) and LTE, there
is a growing need for carrying both voice and data in the same IP based network.
Such an architecture could lead to significant reduction in the costs for operation and
maintenance of the networks. It also enables the operators to introduce new IP based
services like Rich Communication Suite (RCS), that can provide the users with improved
quality of experience at reduced costs. Hence the VoLTE solution should provide the
users with a better quality of experience at reduced costs than the existing CS networks.
Prasanna Gururaj
Raghavendrarao
Introduction
In VoIP based networks, the user perceived Quality of Experience (QoE) depends on
various QoS parameters like delay, jitter, latency, packet loss etc. In addition, since
both the data and voice are carried over the same PS network in LTE, there needs to be
proper classification among them for scheduling of network resources in the radio and
core network domains. During congestion periods, scheduling algorithms used in both
the radio and core networks for allocation of resources play a critical role in meeting
the stringent delay and packet loss requirements of VoIP service as well as the packet
loss requirements of the data service. The capacity of LTE radio network is very high
and it can provide a peak cell throughput of around 300 Mbps in the downlink in the
4x4 MIMO configuration. This places a direct challenge on the transport network with
respect to the scheduling of the resources. The motivation of this thesis is to study
the effects of congestion in the transport network and to analyse its impact on the
performance of voice in LTE network.
1-3
Problem Definition
In mobile broadband networks like LTE, the high performance of the radio network
can be realized with proper scheduling of resources for different types of services. But
proper scheduling of resources in the radio network alone is not sufficient to guarantee
a good end to end performance. During periods of high congestion, packet losses might
occur in the transport network which can reduce the overall performance of the service
that is offered to the user. Hence, the transport network between the radio and core
networks is another area which needs proper dimensioning and scheduling of resources
for various types of services. The transport network is not aware of the QoS architecture
of LTE. This implies that the various bearers that are used to classify the services in
LTE domain needs to be mapped to IP based QoS techniques.
The Differentiated services architecture (Diffserv) which is commonly used in IP based

networks is used to classify the various types of services in the LTE transport network.
The Diffserv architecture needs to be integrated with the LTE QoS architecture to
guarantee good end to end performance. The scheduling of resources in the transport
network is another area which needs proper attention as the choice of scheduling algorithms is pivotal for optimum usage of resources. There are various scheduling strategies
like Weighted Fair scheduling, Strict Priority scheduling and Weighted Round Robin
scheduling that are used to schedule the packets based on the priority of each type
of service. For real time traffic like VoIP, the role of the classification and scheduling
strategies is of paramount importance as they play a crucial role in guaranteeing the
end to end quality of service to the users. During periods of congestion, real time
services like VoIP can be severely impacted if there is a marginal increase in the end
to end delay between VoIP packets or there is a packet loss in the transport network.
The aim of this thesis is to study the various transport network scheduling strategies
and to analyse their impacts on VoIP traffic during congestion periods. The analysis
is done based on simulations using OPNET simulation tool.
Prasanna Gururaj
Raghavendrarao
1-4 Related Work
1-4
Related Work
The transport of voice over LTE has a lot of challenges with respect to QoS as mentioned
earlier. In the literature, there are a number of studies which are focussed on the
optimum scheduling of resources for supporting VoIP service. In [3], Siomina et.al. have
analysed the impact of prioritizing VoIP over other services in the radio network. The
performance of prioritized VoIP is compared with Best Effort VoIP and the advantage
in terms of increase in capacity is explained. In [4], Zaki, et.al. have studied the impact
of dynamic packet scheduling on the performance of VoIP in LTE. Puttonen, J [5] and
Yasir Zaki [6], have studied the impacts of MAC scheduling algorithm for different
types of services. Most of these studies are focussed on the scheduling of resources in
the LTE radio network. To the best of my knowledge there are very few studies that
have been done on analysing the impact of scheduling in the LTE transport network.
The most relevant study in this aspect is done in [7] in which Li, et.al. have studied
the impact of dimensioning in the transport network. In this study, analytical models
have been proposed for dimensioning the transport network for real time and non real
time services and the proposed models are verified by simulations.
1-5
Organization of the Thesis
The thesis is organized as follows.

In chapter 2, the background information related to the network architecture of
LTE and IMS networks is introduced followed by a brief explanation on the QOS
concepts in LTE. The chapter also gives an overview on the Diffserv architecture
that will be used in the transport network for classification of various services
like voice, FTP etc. The chapter concludes with the explanation on the various
scheduling strategies that will be used in the transport network.
In chapter 3, the details of the OPNET modeller are presented. The limitations
of the LTE model in OPNET and the changes that were done on the various
process models are explained. The configuration details of the various nodes in
the LTE network and the final OPNET simulation environment that will be used
for performing the analysis is presented at the end of this chapter.
In chapter 4, the results of the simulation are presented. The chapter begins with
a brief introduction of the various metrics that were used to perform the analysis
followed by the evaluation of different congestion scenarios.
In chapter 5, the technical impacts on deploying VoLTE solution is presented.
This chapter begins with an introduction on the VoLTE architecture followed
by various scenarios that are being considered for integration of VoLTE with
the existing 2G/3G networks. The idea behind this study is to get an industry
perspective on the technical impacts of VoLTE solution.
Prasanna Gururaj
Raghavendrarao
Introduction
In chapter 6, the main results of the thesis are summarized and topics for further
research have been proposed.
Prasanna Gururaj
Raghavendrarao
Chapter 2
Background
2-1
Introduction
This chapter begins with an overview on the LTE network architecture which explains
the functions of the various elements present in the LTE network. The QoS concept
in LTE is presented in section 2.3. The QoS concept in LTE is based on bearers that
uniquely define the type of treatment for the packet flows between the mobile and the
gateway nodes in the network. Hence this section provides the necessary information
required for a better understanding of the QoS concept in LTE. The voice over LTE
solution also requires an IMS core network that performs the necessary signalling and
media related functions for providing voice services. The IMS core network architecture
is presented in Section 2.4 to provide an overview on the functions of the key elements
in IMS domain. In Section 2.5, the Differentiated Services architecture is explained
which will be used for packet classification in the transport network. In Section 2.6,
the scheduling strategies that will be used to analyse the performance of VoIP are
explained.
2-2
LTE Network Architecture
The LTE network architecture is shown in the Figure 2.1. The network architecture
called the Evolved Packet System (EPS) has a flat IP based architecture and is divided
into the Evolved Universal Terrestrial Radio Access Network E-UTRAN and Evolved
Packet Core (EPC). The overall architecture consists of five elements which are explained as follows.
E-UTRAN
The radio network called the E-UTRAN comprises of the E-Node Bs that are interconnected to each other over the X2 interface and connected to the core network elements
Prasanna Gururaj
Raghavendrarao
Background
Figure 2-1: LTE Network Architecture
over the S1 interface. The E-Node Bs are responsible in scheduling and allocation of
the radio resources for the users in the LTE network. The E-Node B terminates the
control plane signalling messages as well as the user plane data with the EPC over the
S1 interface.
EPC
The EPC is the core network comprising of four elements which are Mobility Management Entity (MME), Serving gateway, Packet Data Network (PDN) gateway, Proxy and
Charging Rules Function (PCRF) and Home Subscriber Server (HSS).
MME
MME is the most important element in the EPC as it terminates the control plane
signalling from the user. Some of the functions performed by MME include authentication, mobility management, security and retrieval of subscription information from
the HSS.
Serving gateway
Serving gateway is responsible for forwarding the user plane packets from the mobile
towards the PDN Gateway. It is also responsible for tunnelling the user plane IP packets using the GPRS tunnelling protocol (GTP) when the user moves across different E
Node Bs and serves as a mobility anchor for the user plane packets in the LTE network.
PDN Gateway
Packet data network gateway is the end node in the LTE network. It acts as an edge
router and routes the user plane IP packets from the mobile nodes to other networks
like Internet, IMS etc. It is also responsible for allocation of IP address to the user.
PCRF
PCRF is responsible for enforcing various operator policies on the network like guarPrasanna Gururaj
Raghavendrarao
2-3 QoS Architecture in LTE
anteed QoS, maximum bit rate provisioned for a user etc. It communicates with the
PDN-gateway in enforcing these policies for various users in the LTE network.
HSS
HSS is the master database containing all the subscription information of the user
along with the subscription for various services that are offered by the operator. It also
comprises of the authentication centre which stores all the keys required for ensuring
the encryption and integrity of the data in the network.
2-3
QoS Architecture in LTE
In LTE, the QoS is provided by means of a bearer which uniquely identifies the packet
flow between the user and the PDN-GW and is responsible for the priority that is given
to a packet flow across the LTE network. Bearers are established after the successful
authentication and registration of the user in the LTE network. The LTE bearer
architecture is shown in the Figure 2.2.
Figure 2-2: QoS Architecture in LTE [8]
Each bearer is associated with a Traffic Flow Template (TFT) which is used to differentiate the types of packets that flow through it. The TFT does this classification
based on one of the following parameters:
Port numbers
ToS/DSCP Values
Source/Destination address
Protocol (TCP/UDP)
Prasanna Gururaj
Raghavendrarao
10
Background
QCI
Resource Type
Priority
Packet Delay Budget
1
2
GBR
GBR
2
4
100
150
Packet
Error Loss
Rate
102
103
GBR
50
103
GBR
300
106
5
6
Non-GBR
Non-GBR
1
6
100
300
106
106
Non-GBR
100
103
Non-GBR
300
106
Non-GBR
300
106
Services
Voice.
Voice Conversation (Real Time
Streaming).
Real Time Gaming.
Non
Conversational
Video
(buffered video).
IMS Signalling.
Video (Buffered
Streaming).
Interactive Gaming.
Video (Buffered
Streaming).
Video (Buffered
Streaming).
Table 2-1: EPS QOS Bearer Definitions [9]
The bearers are classified as two types namely the default and dedicated bearers. Default bearers are established during the allocation of IP address to the user by the PDN
Gateway. Default bearers provides the basic IP connectivity to the LTE network and
does not provide any guaranteed QoS for the packets that are transmitted across this
bearer. Dedicated bearers are used for specific services like voice, video streaming etc
and are established based on the subscription profile of the user.
The bearers are also classified as Guaranteed Bit Rate (GBR) and Non Guaranteed
Bit Rate (N-GBR). As the name indicates the GBR bearers provide guaranteed QoS to
the packets that flows through this bearer and is less likely to be affected during heavy
congestion at the network. On the other hand, the N-GBR bearers are used for services
that do not have strict QoS constraints. As shown in the Figure 2.2, each bearer in
LTE is characterized by a QoS Class Identifier (QCI), Allocation and Retention Priority
(ARP), packet delay budget and maximum bit rates. The QCI uniquely identifies the
type of bearer that is provisioned for the user at the radio and core networks. It is used
in determining the type of treatment a packet flow experiences at each of the nodes in
the LTE network. The ARP is used to decide whether a bearer can be admitted and is
also used to release the bearers based on priority levels when the network is congested.
The Table 2.1 [9] summarizes the values for each type of bearer.
Prasanna Gururaj
Raghavendrarao
2-4 IMS Network Architecture
2-4
11
IMS Network Architecture
IP Multimedia Subsystem is a core network architecture standardized by the 3GPP [10]

to provide multimedia services like voice, streaming services like video on demand etc
over an IP backbone independent of the underlying access network through which the
user registers with it . The most important service provided by an IMS network is the
Multimedia Telephony Service (MMTel) which is the basic voice over IP service but
with guaranteed QoS. IMS is also capable of interworking the circuit switched 2G/3G
network with packet switched networks like LTE. Hence IMS based voice is envisioned as
the ultimate target solution for supporting voice in advanced next generation networks
like LTE.
The Figure 2.3 presents the key elements of an IMS network. The main elements
of IMS core network are the Proxy Call Session Control Function(P-CSCF), Serving
Call Session Control Function (S-CSCF), Interrogating Call Session Control Function
(I-CSCF), Breakout Gateway Control Function (BGCF), Media Gateway Control Function (MGCF) and Media Resource Function (MRF). The main functions of these entities
are explained as follows:
Figure 2-3: IMS Network Architecture
P-CSCF : P-CSCF is a SIP proxy server in the IMS domain which is the first
point of contact for the user within the IMS domain. All the requests of the user
to the elements in the IMS domain as well as to the application servers are routed
through the P-CSCF. In addition, the P-CSCF performs functions like subscriber
Prasanna Gururaj
Raghavendrarao
12
Background
authentication and establishment of security association with the mobile. It may

also authorize QoS resources for the voice bearer by way of a policy decision
function.
S-CSCF : S-CSCF is the main element in the IMS domain which performs important functions like subscriber registration, authorization for using specific application servers, DNS lookup to retrieve the address of the destination etc. It
downloads the user profiles from the HSS for performing authorization of the
subscriber.
I-CSCF : I-CSCF is a SIP server that acts as a last point of contact in the IMS
domain i.e. it is at the edge of the IMS domain and all requests from other IMS
domains as well as requests from remote application servers are routed through
the I-CSCF. During initial registration, the I-CSCF queries the HSS to assign a
S-CSCF for the specific user.
BGCF : BGCF performs breakout to other domains when routing of the request
based on ENUM lookup is failed at the S-CSCF. It is mainly used when the
destination user is a PSTN user and the call needs to be transferred to the CS
domain.
MGCF : MGCF is used to translate SIP signalling into ISUP signalling for communication towards PSTN and other CS networks. It also controls the media
gateway which translates the RTP into CS media stream.
MRF : MRF is used in transcoding between different codecs and provides media
related functions like mixing of media streams and playing tones etc. The MRF is
subdivided into Media resource function controller (MRFC) and Media resource
function processor (MRFP) which perform the media translation activities in the
control and user plane respectively.
2-5
Differentiated Services Architecture
The Differentiated services (Diffserv) architecture defined in [11] is used by the E-Node
B and the PDN gateway to map the QCI to a DSCP value in uplink and downlink
respectively. This mapping at the E-Node B and the PDN gateway allows for the classification of the packets in the underlying transport network. The Diffserv architecture
consists of various Per Hop Behaviours (PHB) that are used to identify and classify
the packets and apply appropriate QoS treatment at the transport network. The PHB
classes are broadly classified into three classes namely Expedited Forwarding(EF), Assured Forwarding(AF) and Best Effort(BE).
The EF class has the highest priority and is generally used for delay critical services
like signalling, voice etc. The AF class consists of several sub classes with different
levels of drop precedences as shown in the Table 2.2. The drop precedence enables the
Prasanna Gururaj
Raghavendrarao
2-6 Scheduling Strategies
Drop Precedence
Level 1
Level 2
Level 3
13
AF 4X
AF41
(DSCP 34)
AF42
(DSCP 36)
AF43
(DSCP 38)
Af 3X
AF31
(DSCP 32)
AF32
(DSCP 30)
AF33
(DSCP 28)
AF 2X
AF21
(DSCP 26)
AF22
(DSCP 24)
AF23
(DSCP 22)
AF 1X
AF11
(DSCP 20)
AF12
(DSCP 18)
AF13
(DSCP 16)
Table 2-2: Assured Forwarding Drop Precedence Classification
operator to provide various levels of QoS for different types of services. The Best Effort
class is the default PHB and has the least priority among the three classes. The AF
class and BE class employ the Weighted Random Early Detection technique to detect
congestion of queues based on pre defined thresholds. When the number of packets
in the queue exceeds a minimum threshold, the WRED technique starts dropping of
packets based on the weight assigned to each queue. If the link is heavily congested and
the number of packets in the queue exceeds the maximum threshold then all incoming
packets to the queue are dropped.
2-6
Scheduling Strategies
The transport network consists of a scheduler which assigns the available network bandwidth based on certain priorities and weights. The scheduler uses the classification of
the packets based on DSCP to form these strategies for allocation of resources in the
transport network. In this work, three different scheduling strategies are evaluated and
their performance is compared for delivering high quality voice service in LTE network.
The following section gives an overview on the different scheduling strategies that are
implemented in the transport network.
2-6-1
Strict Priority Scheduling
In this type of scheduling, the packets are grouped into four levels of priority namely
low, normal, medium and high. Packets which are very sensitive to delay like voice are
given a high priority and services like streaming which have tolerable delay budgets
are given medium priority. TCP based services like HTTP and FTP are mapped to
normal and low priorities respectively as they have less constraint on the delay budgets.
The scheduler always processes the high priority packets before servicing packets in
other queues. This scheduling is especially useful for services like VoIP which have
stringent delay requirements. The major drawback of this scheduling is, when the
network is congested with high priority traffic like voice, the low priority data traffic
will completely devoid of resources and hence the overall throughput of the network is
reduced.
Prasanna Gururaj
Raghavendrarao
14
2-6-2
Background
Weighted Round Robin (WRR) Scheduling
This type of scheduling is based on the classical round robin scheduling where different
types of services are served in a round robin manner. The only addition in WRR is
the presence of weight which determines the number of packets that are removed from
the queue. The packets are grouped into various queues and each queue is assigned a
weight. Based on the weight, the scheduler calculates the bandwidth for each queue
and corresponding to this bandwidth, number of packets in the queue are removed at a
time before moving to the next queue. Hence WRR does packet by packet scheduling
in a round robin manner.
2-6-3
Weighted Fair Scheduling
In Weighted Fair scheduling defined in [12], the packets are grouped into various queues
and each queue is assigned a weight which determines the fraction of the total bandwidth available to the queue. In our case, there are different PHBs such as EF, AF
and BE are assigned weights based on the priority of the traffic. The bandwidth for
each queue is based on the weights and is expresses as
BWk =
Wk
BW
W
(2-1)
The Weighted Fair scheduling assigns the bandwidth for each service based on the
weight assigned to each queue and not based on the number of packets. Hence when
various types of traffic like VoIP, FTP, HTTP are flowing in the network, the bandwidth
for each service is proportional to its weight and independent of the size of the packet
in the queue. The main difference between Weighted Round Robin and Weighted Fair
is that the former does packet by packet scheduling in each turn whereas the latter
does bit by bit scheduling. Weighted Fair hence has an advantage in the fact that it is
aware of the true size of the packets in each queue while performing scheduling whereas
Weighted Round Robin is not aware of the same.
Prasanna Gururaj
Raghavendrarao
Chapter 3
Simulation Model
3-1
Introduction
This chapter begins with an explanation on the details of the OPNET simulation environment used for modelling the LTE network. In section 3.3, the issues with the LTE
model in OPNET are presented briefly and the modifications that were performed to
resolve these issues are highlighted. The configuration parameters of the LTE network
and the simulation settings for the VoIP and FTP process models are provided in the
subsequent sections.
3-2
Introduction to OPNET Modeller
The OPNET simulation environment [13] is a discrete event simulation tool that is
used in analysing the performance of various networks like LTE, WiMAX, Wi-Fi and
Zigbee. The models library in OPNET consists of a large number of models supporting
variety of protocols like TCP, UDP, SIP and is capable of simulating applications like
voice, video, FTP etc. In this thesis, the LTE model in OPNET is used along with
application models like voice and FTP. The details of the simulation environment are
presented in the following sections.
3-2-1
Overview of LTE Model in OPNET
The OPNET modeller has a hierarchical environment consisting of the network model,
node model and process model. All the three models need to be configured to perform
the simulation. The LTE network model in OPNET is shown in Figure 3.1. The model
consists of mobile nodes , an E-Node B and an EPC. The LTE core network consisting
of the MME, serving gateway and PDN-gateway is modelled by a single device represented as the EPC in the Figure 3.1. The LTE attribute definition node is used to
Prasanna Gururaj
Raghavendrarao
16
Simulation Model
Figure 3-1: LTE Network Model
define various configuration parameters like DL and UL frequencies, bandwidth and

the various bearers that will be configured on the mobile nodes. The LTE model implements most of the features that are standardized by the 3GPP. However, it has some
limitations in the establishment of bearers and hence significant changes are required in
the model to perform our analysis. The Figure 3.2 gives the data flow in LTE network.
Figure 3-2: Data Flow in LTE Network [13]
It is seen that for each bearer in the radio network, there is a corresponding S1 bearer
in the transport network. This S1 bearer uses the GPRS Tunnelling Protocol (GTP).
Prasanna Gururaj
Raghavendrarao
3-2 Introduction to OPNET Modeller
17
Hence for each bearer that is established between an user and EPC, there is a separate
GTP tunnel established for control plane signalling as well as user plane data. The
signalling GTP tunnel is used for transmitting all the signalling information related to
the establishment of the bearer. The data GTP tunnel is used in forwarding all the
user plane IP packets from the user to EPC and vice versa.
The Figure 3.3 gives the complete protocol architecture across various nodes in the
LTE network. The GTP-U layer in the E-Node B represents the tunnels that are created between the E-Node B and EPC. In the uplink when the E-Node B receives IP
packets from the mobile, the GTP layer encapsulates the received IP packet and copies
the contents of the inner IP header to the outer IP header. The same process is repeated
in the downlink direction when the EPC node receives an IP packet from outside the
LTE network. The encapsulated GTP packet structure is shown in the Figure 3.4.
Figure 3-3: Protocol Architecture
Figure 3-4: GTP Encapsulated Packet
Prasanna Gururaj
Raghavendrarao
18
Simulation Model
3-2-2
Issues in LTE model
In the LTE model present in OPNET, the process of GTP encapsulation is not implemented in accordance to the QoS type of the bearer . The packets entering the E-Node
B in the uplink are encapsulated into an IP packet without any classification based on
the type of bearer (DSCP mapped to BE by default). The same issue is there in the
downlink when the packets entering EPC are encapsulated without proper classification. Due to this problem, when there are different types of services like voice, FTP,
HTTP, video streaming etc., there is no proper classification of packets at the IP level
in the transport network.
As explained in the problem definition in Chapter 1, the intermediate nodes in the
transport network between E-Node B and EPC are not aware of the classification based
on bearers. The type of scheduling strategies used in the transport network also has no
meaning, if all the packets are classified with same priority. So, it is very important for
the E-Node B and EPC to perform packet level classification by mapping the bearer
type to DSCP. Hence changes are required in the process models in the E-Node B and
EPC. The following sections illustrate the changes that were performed in the E-Node
B and EPC to achieve this objective.
3-3
Changes in LTE model
As mentioned in the previous section, there is no packet level classification among the
bearers in the transport network which needs to be implemented. This section presents
an overview on the changes that were done in the E-Node B and EPC nodes in the
OPNET.
3-3-1
LTE S1 process model In E-Node B
The node model for the E-Node B is shown in Figure 3.5. The node model gives an
overview on the various layers of the 3GPP LTE stack that are implemented in E-Node
B. In this node model, there are two processes lte_ s1 and gtp (highlighted in Figure
3.5) that are to be modified. The lte _ s1 process model at the E-Node B is shown in
the Figure 3.6. This process is run every time when a dedicated bearer is created in
the LTE network. The s1 _ msg _ rcvd represents the state in which the E-Node B
has received a new bearer request message from the mobile. The state runs a dedicated
bearer setup function and commands the GTP layer in the E-Node B to create a tunnel
for this dedicated bearer towards the EPC in the uplink direction. So, the change that
needs to be performed in this process is to map the QoS type of the tunnel created to
the type of the bearer that is received in the request. This is done as follows:
Each bearer configured in the mobile node is mapped to a bearer ID which
uniquely identifies the bearer in the network.
Prasanna Gururaj
Raghavendrarao
3-3 Changes in LTE model
19
Figure 3-5: E-Node B Node Model
Figure 3-6: lte _ s1 Process model in E-Node B
The bearer is also configured with a TFT as explained in Section 2.3 which is used
to perform the mapping between the bearer level QoS and DSCP value in the IP
header. So there is an indirect mapping between the bearer ID and DSCP value.
By using this mapping, the functions in the process model are changed such that
during the creation of GTP tunnel between the E-Node B and EPC, the DSCP
value corresponding to the bearer ID is also taken into consideration.
Prasanna Gururaj
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20
3-3-2
Simulation Model
LTE S1 NAS Process model in EPC
The node model of the EPC is shown in Figure 3.7. Similar to the E-Node B node
model, the EPC also has the lte _ s1 _ nas and gtp process models that are responsible
for bearer creation and tunnel creation respectively. Hence these two processes are to be
changed to overcome the issues explained in the previous section. The Figure 3.8 shows
Figure 3-7: Node Model in EPC
the lte _ s1 _ nas process model. This process model is used to setup the S1 bearer
that carries the data between the E-Node B and EPC for the mobile in the downlink
direction. In the Figure 3.8, the state s1 _ msg _ rcvd represents the state in the
EPC corresponding to the one explained in the previous subsection at the E-Node B.
This state acts as a trigger towards the state nas _ msg _ rcvd which indeed actually
contains the functions responsible for setting up the S1 bearer towards the E-Node B.
The mapping procedure used in the previous section for E-Node B is again followed
here in the EPC.
3-3-3
GTP Process model in E-Node B and EPC
The GTP process model that runs in both the E-Node B and EPC nodes is shown in
Figure 3.9. The GTP-U block performs the encapsulation of the user plane IP packet
received at the E-Node B and delivers it to the UDP layer for transport towards the
EPC. The GTP-U block consists of four states which are idle, tunnel search, gtpencap
and to UDP. When a packet arrives at the E-Node B, the process goes from the idle
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Raghavendrarao
3-3 Changes in LTE model
21
Figure 3-8: lte _ s1 _ nas Process model in EPC
state to the tunnel search state. If the tunnel corresponding to the bearer is found,
the process goes to the gtpencap state, where the packet is encapsulated and sent to
the UDP module. Else, the process goes to the tunnel management state, where the
tunnel creation function is executed before the encapsulation of packet is performed.
The same process is repeated at EPC in the downlink direction when a packet arrives
from outside the LTE network.
Hence the gtpencap state is where the actual process of mapping between the bearer
ID and the DSCP takes place and the contents of the inner IP header are copied to the
outer IP header. The various functions that were used to perform this mapping in the
GTP layer were modified to overcome the limitations that were mentioned in Section
3.1.2.
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22
Simulation Model
Figure 3-9: GTP Process Model
3-4
Simulation Environment
This section lists all the configuration details of the various nodes that were used in
the analysis. The network topology used for performing this simulation is shown in the
Figure 3.10. In the network topology, there are two cells represented by E-Node B 1
and E-Node B 2 connected to the EPC via an Edge router. Each cell consists of 30
LTE users. The EPC node is connected to an IMS network via an edge router. The
Ethernet links between the E Node B and the EPC are 5 Mbps. All other links in the
core network are of 10 Mbps capacity. There are two FTP servers connected to the
EPC node which are used by the mobile nodes for establishing FTP sessions in the
network. The configuration details of each of the nodes are explained below.
3-4-1
Mobile Node
The LTE mobile nodes are configured to run VoIP and FTP services. Each mobile
node is configured to run one type of application at a time. The Figure 3.11 shows the
important configuration details of the mobile nodes in the network. The EPS bearer
configuration attribute defines four bearers namely Platinum, Gold, Silver and Bronze.
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3-4 Simulation Environment
23
Figure 3-10: LTE Simulation Network
Each of the bearer is assigned to a TFT packet filter which in our case is the DSCP
value. The Table 3.1 shows the mapping between the bearer type and DSCP value.
This mapping is used by the mobiles to identify the type of bearer for different types of
Bearer Type
Platinum
Gold
Silver
Bronze
DSCP
EF
AF 11
AF 43
BE
Table 3-1: EPS Bearer to DSCP Mapping
services like voice, FTP, etc. The mobility feature in the mobile nodes is set to disabled
as we assume that all the mobiles are stationary in the area around the E-Node B.
3-4-2
E-Node B
The E-Node B in the network is configured with 3 MHz bandwidth. The total capacity
of each cell is limited to 10 Mbps. The channel between the mobile nodes and E-Node
B is configured to be an error free channel as the primary objective of this analysis is to
investigate the impact of congestion in the core network. Hence various physical layer
effects like multipath and interference effects are not modelled in these simulations.
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24
Simulation Model
Figure 3-11: Mobile Configuration
The MAC scheduler implemented in the OPNET E-Node B module uses a priority
scheduling among the guaranteed and non guaranteed bit rate bearers which implies
the guaranteed bit rate bearers are always allocated radio resources ahead of the non
guaranteed bit rate bearers. To avoid the scenario of packets getting dropped due to
non availability of resources in the radio network, the peak usage of each cell is limited
to 50 percent of the total capacity. The summary of the configuration parameters of
the E-Node B is listed in the Table 3.2.
Parameter Name
Bandwidth
UL Frequency
DL Frequency
Channel Characteristics
No. of Transmit/Receive Antennas
Value
3MHz
1920 MHz
2110 MHz
Error Free
2
Table 3-2: E-Node B Configuration Parameters
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3-4 Simulation Environment
3-4-3
25
IMS Model
The IMS model used in this simulation environment is used from the contributed models
section available in [13]. It consists of proxy, serving and interrogating call session
Figure 3-12: IMS Proxy session control function configuration attribute
control functions (P/I/S-CSCF) which are used in signalling procedures for the VoIP
calls between the different users in the network. The IMS signalling flow in the LTE
network requires the highest priority as it is the first procedure that is invoked towards
the establishment of the VoIP call between the users. Hence all the IMS signalling
packets are marked with the highest priority in both the radio and core networks.
The Figure 3.12 gives the configuration attribute of the P-CSCF in the IMS network.The domain name and area configured in these servers are also configured in the
mobiles and using these attributes, each mobile registers with the IMS network. The
three call session control functions are used to route the signalling between two VoIP
users before the establishment of the media path. The SIP signalling procedure defined
in [1] is followed for establishment of the VoIP calls between the users in the network.
The IMS model is used in our simulation only to emulate the real world scenario as the
main focus of our study is on the user plane voice bearer and not on the control plane
signalling data.
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26
3-4-4
Simulation Model
Application Configuration
We use two traffic models namely voice and FTP for performing this analysis. The
details of the traffic models are explained below.
VoIP model
We use the commonly used G.711 voice codec for all the simulations. The codec has
a bit rate of 64 Kbps with 20 milliseconds frame size and 1 frame per packet. Hence,
there are 50 packets that are transmitted per second. The RTP/UDP/IP layers add
headers to each packet and hence the overall bandwidth is around 90 kbps. In our
simulations, silence suppression is used and is modelled as an exponential distribution
with talk spurt length of 1.2 seconds(mean) and silence length of 0.8 seconds(mean).
A summary of the configuration details is given in Table 3.3.
Parameter Name
Codec
Frame Size
Voice Activity Factor
Silence Suppression
Value
G.711 (64 Kbps)
20 ms
0.6
Enabled
Table 3-3: VoIP Configuration Parameters
FTP model
The FTP server is configured to send a file of size 1 MB upon request by each mobile.
The inter repetition time between requests is 30 seconds. There is a separate TCP
connection established for each request between the mobile and the server.
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Chapter 4
Results
4-1
Introduction
In this chapter, the performance of the various scheduling scheduling strategies that
were explained in Chapter 2 are investigated. The chapter begins with an introduction
on the various QoS parameters that were used for performing the analysis. In all the
simulations, only voice and FTP services are used. The details of the traffic models for
voice and FTP are as explained in Chapter 3.
4-2
QoS parameters for Voice
The following are the parameters that were used to determine the QoS of the VoIP call
in the LTE network.
Packet End to End delay : This parameter gives the total voice packet delay i.e.
the mouth to ear delay between the users. In all simulations, the mean end to
end delay is shown for the all the users in the network.
Packet Delay Variation (PDV) : This parameter gives the variance in the end to
end delay among all the packets received at the user. The mean of this PDV is
shown for all the users in the network.
S1 Delay : S1 delay is the one way delay between the E-Node B and the EPC.
This parameter gives the mean time taken for a packet to traverse between the
E-Node B and EPC. The S1 delay is measured at the E-Node B.
Packet Loss Rate (PLR): The packet loss rate gives the number of voice packets
that are lost in the network due to congestion. The packet loss rate is measure at
the EPC node, since the congestion is in the core network. The mean of the PLR
for all the users in the network is shown for all the simulations.
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Results
In addition to the above QoS parameters, the Mean Opinion Score (MOS) is also presented for all the simulations. The MOS is a measure of the Quality of Experience for
the VoIP users in the network. The E-Model defined in [14] is used to calculate the
MOS based on the R-factor. The R-factor called the rating factor is used to measure
the quality of the VoIP call based on various parameters like packet end end delay,
packet loss etc. The R-factor is expressed as follows [14]
R = 94.2 Id Ie
(4-1)
where Id is the impairments caused due to the mouth to ear delay and Ie is the impairment caused due to packet losses in the network. The R-factor is mapped to a MOS
score using the following mapping defined in [14]:
M OS = 1 + 0.035R + 7 106 R(R 60)(100 R), 0 R 100
(4-2)
M OS = 1, R 0
(4-3)
M OS = 4.5, R > 100
(4-4)
The OPNET software uses the above model to calculate R factor and is mapped to the
MOS using the above equation. The MOS score is mapped to the level of satisfaction
of the users based on Table 4.1 [14].
MOS score
4.3 - 5
4 - 4.3
3.6 - 4
3.1 - 3.6
2.6 - 3.1
Less than 2.6
Quality of VoIP call experienced by the user

Very much satisfied
Satisfied
Many users satisfied
Many users dissatisfied
Nearly all users dissatisfied
Not recommended
Table 4-1: MOS satisfaction level
In all the simulations, the average of the MOS for all VoIP users in the network is
presented.
4-3
Scenario 1
This scenario is used to illustrate the significance of QoS in the transport network by
mapping both VoIP and FTP users with the same priority in the transport network.
4-3-1
Description of the Scenario
Case 1: In this case, only voice traffic is generated in the network. The number
of voice users in the network is periodically increased from 20 to 100. There are
totally 25 LTE mobiles running VoIP application in each cell and each user is
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4-3 Scenario 1
29
configured to establish multiple VoIP sessions simultaneously. The VoIP session

is carried over a Guaranteed Bit Rate (GBR) bearer (QCI 1 in Table 2.1) and is
mapped to default best effort (DSCP-BE) QoS in the transport network.
Case 2: In this case, both voice and FTP traffic are generated in the network.
The number of voice users are same as the previous case and there are totally
10 FTP users (5 in each cell). VoIP users are mapped to GBR bearer as in case
1 and FTP users are mapped to Non GBR bearer (QCI 9 in Table 2.1) in the
downlink. Both the voice and FTP are mapped to the same best effort QoS in
the transport network. So, the packets entering the nodes EPC, Edge Router 1
and E-Node B 1 & 2 are served with First In First Out Scheduling (FIFO). Hence,
this case analyses the performance of best effort VoIP service when the network
is congested with data service.
4-3-2
Analysis of results
The Figures 4.1 and 4.2 show the mean end to end delay and mean packet delay variation for VoIP users in the network. For case 1, the delay is constant at 80 ms whereas
when there is an ongoing FTP session in case 2, there is a significant increase in the
end to end delay for VoIP users. In case 1, since there are only VoIP users present in
Figure 4-1: Voice Packet End to End Delay vs No. of VoIP users
the network, the total traffic in the link between the Edge Router 1 and the EPC is still
within the total bandwidth even when the number of VoIP users is large. At 100 VoIP
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30
Results
Figure 4-2: Packet Delay Variation Vs No. of VoIP users
users the peak traffic that can be expected is maximum at 9 Mbps. This value is arrived by assuming that all the users are sending voice packets at the same time and the
bandwidth required for a single voice call is 90 Kbps after adding the RTP /UDP/ IP
headers to the actual voice payload. The peak traffic will never be reached as each user
has an exponential distribution on the talk spurts and silence periods. So, the mean end
to end delay for all VoIP users in the network remains constant. This also explains the
Figure 4.2, which shows no variation in delay among the packets received at the mobile.
In case 2, since there is no priority among voice and FTP, the smaller VoIP packets are getting queued in the core network and the edge router till the larger FTP
packets are processed in each node. This causes a larger variation among the packets
received at the mobile as shown in Figure 4.2. The minimum mean delay is 150 ms,
when the number of VoIP users is 20 and is much higher than the acceptable limit of
100 ms.
From the scenario, it is evident that to achieve an acceptable QoS for VoIP in LTE,
there needs to be proper classification in the transport network.
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4-4 Scenario 2
4-4
31
Scenario 2
In Scenario 2, the voice users are accorded the highest priority and the FTP users are
mapped to the lowest priority. Each VoIP user is assigned to a Platinum bearer (QCI
1 in Table 2.1) and mapped to the highest EF QoS class in the transport network. The
FTP users are assigned to a Bronze bearer (QCI 9 in Table 2.1) and mapped to the
BE QoS class in the transport network.
4-4-1
This scenario evaluates the performance of the scheduling algorithms explained in Chapter 2. The scenario is subdivided into three cases as follows:
Case 1: In this case, the Weighted Fair scheduling algorithm explained in Chapter
2 is analysed. The high priority VoIP traffic is assigned a weight of 7 and the FTP
traffic is assigned a weight of 3. Hence, the total bandwidth assigned for voice
users is 7 Mbps and the total bandwidth for FTP users is 3 Mbps.
Case 2: In this case, the Strict Priority scheduling algorithm explained in Chapter
2 is analysed. In terms of priority as explained earlier the VoIP users are mapped
to high priority and FTP users are mapped to low priority.
Case 3: In this case, the Weighted Round Robin scheduling algorithm explained
in Chapter 2 is analysed. The weights for the voice and FTP services are same as
those assigned in Case 1.
4-4-2
Analysis of results
This section presents an analysis on the various parameters that were explained in
Section 4.1.
Packet end to end Delay and PDV
In Figure 4.3, the mean end to end delay for the voice packets is shown. In case 1 and
case 3, the Weighted Fair and Weighted Round Robin scheduling algorithms a definite
bandwidth is assigned to the VoIP users. Hence till this bandwidth limit is reached,
the mean end to end packet delay shown in Figure 4.3 is constant at 80 ms. When the
number of VoIP users is 80, the mean end to end delay is around 100 ms for both the
cases which is still within the acceptable limit. The increase in the end to end delay is
attributed to the fact that the peak bandwidth for 80 VoIP users is around 7.2 Mbps.
This bandwidth is higher than the provisioned bandwidth for VoIP users based on the
weight which is calculated to be around 7 Mbps. The delay falls within the 100 ms
threshold as the peak bandwidth for voice will not be reached due to the exponential
distribution of the talk spurts between the users.
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Results
Figure 4-3: End to End Delay Vs No. of VoIP Users
Figure 4-4: S1 Delay Vs No. of VoIP Users
The Figure 4.4 shows the mean one way S1 delay i.e. the time taken for the voice
packets to reach the E-Node B from the EPC.
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4-4 Scenario 2
33
Figure 4-5: PDV vs No. of VoIP users
When the number of VoIP users is 80, the S1 delay shows a substantial increase
which explains the overall increase in the end to end delay at the mobile node.
When the number of VoIP users reaches 100, there is a significant increase in the
bandwidth demand of the VoIP users, leading to more waiting time in the queues
at the core network as shown in Figure 4.4. The mean end to end delay at this
point is around 120 ms which is beyond the tolerable limit.
The PDV shown in Figure 4.5, follows a similar pattern like the mean end to end delay
with the variation increasing to 0.3 ms. The PDV is small compared to the No QoS
case explained in Scenario 1. This is because of of the separate bandwidth provisioning
for VoIP and FTP users in these two scheduling algorithms.
The mean end to end delay and delay variation for Strict Priority scheduling is also
shown in Figures 4.3 and 4.5. The delay remains constant at 80 ms and packet delay
variation is negligible. As explained in the section 2.6, the Strict Priority scheduling
always performs better for VoIP users which are assigned a higher priority compared
to the FTP users.
PLR and MOS
In Figures 4.6 and 4.7, the PLR for the VoIP users in the network and the corresponding
MOS is shown. From the Figure 4.6, it is seen that for case 1 and case 3, the packet loss
rate is almost negligible till number of VoIP users is equal to 80 when the packet loss
rate reached the threshold of 2 percent. Beyond this, with an increase in the number of
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34
Results
Figure 4-6: Packet Loss Rate vs No. of VoIP users
Figure 4-7: Mean Opinion Score vs No. of VoIP users
VoIP calls, the packet loss rate also increases significantly. The value beyond this point
is of no significance, as more than 2 percent drop in the number of packets implies that
the VoIP calls are dropped. The high packet loss rate beyond this point attributes to
the sharp decrease in the value of the Mean Opinion Score shown in Figure 4.7.
In case 1, the minimum value of MOS is 2.8 as shown in Figure 4.7 whereas in case 3,
the minimum MOS value is 3. In both cases, the lower value of MOS implies that all
the users are dissatisfied and hence beyond 80 users, there is no possibility of having
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4-4 Scenario 2
35
more number of VoIP users for case 1 and case 3. In case 2, the PLR is negligible as
shown in Figure 4.6 and hence there is no impact on the MOS values for case 2.
4-4-3
Impact on FTP traffic
The increase in the number of VoIP users will have a direct impact on the file transfer
time for the FTP users in the network. The Figure 4.8 shows the FTP transfer time
for the three cases. The transfer time is increased by almost twice in case 1 and case
Figure 4-8: FTP Transfer time vs No. of VoIP users
3 when the number of VoIP users reaches 100. The increase is mainly due to the the
congestion in the link between the EPC and the Edge router, thereby leading to more
queuing of packets in the EPC. The transfer time for case 2 shows a large increase
when the number of VoIP users in the network increases beyond 80. This leads to an
undesirable situation where there are more TCP re transmissions in the network leading
to increased congestion. The Figure 4.9, shows the total FTP traffic received by all the
users in the network. It is seen from the Figure 4.9, that there is significant drop in
the FTP throughput when the number of VoIP users increase in the network for all the
three cases. For case 1 this drop in throughput is around 30 percent whereas for case 3
the drop in throughput is around 50 percent. Weighted Fair scheduling performs better
than Weighted Round Robin due to the fact that it performs bit by bit scheduling and
hence is aware of the actual packet size of FTP packets before scheduling of resources.
So, there is less delay for servicing large FTP packets in Weighted Fair compared to
Weighted Round Robin which explains the behaviour in Figure 4.9. For case 3 there is
almost a 75 percent drop in throughput when the number of VoIP users in the network
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Results
Figure 4-9: FTP Throughput vs No. of VoIP users
increases from 80 to 100. Hence the overall QoS for FTP traffic is severely degraded
due to the increase in the number of voice users for case 2 when compared to case 1
and case 3.
4-4-4
Summary of the Results for Scenario 2
In this scenario, the performance of VoIP when mapped to Platinum bearer is analysed
for three scheduling algorithms. Within the acceptable QoS thresholds (100 ms end
to end delay and 2 percent packet loss), the number of VoIP users for case 1 and case
3 is 80 Users. The case 2 has a higher capacity of 100 users within the acceptable
limits but it comes at a cost as Strict Priority scheduling totally starves the resources
for low priority traffic when the network is congested with high priority traffic. Hence,
the FTP transfer time shows more than 50 percent increase compared to case 1 and 3
which is not acceptable in environments with mixed traffic.
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4-5 Scenario 3
4-5
37
Scenario 3
In Scenario 3, each VoIP user is assigned to a Gold bearer (QCI 7) in both directions
and mapped to the AF 11 QoS class in the transport network. The FTP users are
assigned to the Bronze bearer as in Scenario 2 and mapped to the BE QoS class in the
transport network.
4-5-1
This scenario evaluates the performance of VoIP when assigned to a normal priority
in the radio and core networks. The scenario is subdivided into three cases as earlier
described in Scenario 2.
Case 1: In this case, the Weighted Fair algorithm is analysed as in Scenario 2.
The VoIP traffic is assigned a weight of 5 and the FTP traffic is assigned a weight
of 3 in the transport network. Hence, the total bandwidth assigned for voice users
is around 6 Mbps and the total bandwidth for FTP users is around 4 Mbps.
Case 2: In this case, VoIP users are assigned Normal priority and FTP users are
assigned low priority.
Case 3: The weights for the Weighted Round Robin algorithm are same as those
in case 1.
4-5-2
Analysis of results
Packet end to end delay and PDV
In Figure 4.10, the mean end to end delay for the VoIP users are shown. The mean
end to end delay from the Figure 4.10 for case 1 and case 3 shows a significant increase
when the number of VoIP users is beyond 60. At 70 users, the delay crosses the threshold of 100 ms. This behaviour is due to the fact that when the number of VoIP users
crosses 70, the maximum peak bandwidth for voice reaches around 6.3 Mbps which is
slightly more than the provisioned bandwidth for voice which is 6 Mbps. The increase
in the end to end delay is marginal until the number of voice users reaches 80 when
the peak bandwidth is around 7.2 Mbps. At this point, more number of VoIP packets
are buffered in the interface of EPC node thereby increasing the end to end delay. The
Figure 4.11 exactly proves this point as the mean one way delay between the EPC and
E- Node B increases significantly when the number of VoIP users crosses 70.
The case 2 in this scenario follows the same behaviour as in scenario 1. This is mainly
due to the fact that merely changing the priority to normal does not affect the VoIP
quality as still VoIP packets are served first by the scheduler before the FTP users are
served. The PDV is shown in Figure 4.12 for all the three cases and as explained for
scenario 1, the packet delay variation has no significant impact when the number of
VoIP users are increased.
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Results
Figure 4-10: End to End Delay vs No. of VoIP users
Figure 4-11: S1 delay vs No. of VoIP users
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4-5 Scenario 3
39
Figure 4-12: Packet delay variation vs No. of VoIP users
PLR and MOS
The Figure 4.13 shows the PLR for this scenario. The PLR for case 1 and case 3 exceeds
the threshold value of 2 percent when the number of VoIP users is around 65 users.
Beyond this point, there is a significant increase in the packet loss rate which implies
that beyond this point the calls will get dropped. In case 2, there is no packet loss as
it follows the same behaviour as explained in scenario 2. The Figure 4.14 shows the
average MOS for all the users in the network. It is seen that the quality of experience
degrades for case 1 more than for case 3 when the number of users is increased beyond
60.
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40
Results
Figure 4-13: Packet Loss Rate vs No. of VoIP users
Figure 4-14: Mean Opinion Score vs No. of VoIP users
4-5-3
In this scenario, the FTP traffic is not impacted much when the number of VoIP users
is increased. This can be observed in the Figure 4.15 which shows the mean file transfer
time of all the FTP users. When the number of VoIP users are increased beyond 80, the
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4-5 Scenario 3
41
mean FTP transfer time for case 1 almost remains constant and for case 3 it increases
marginally by 3 percent. In comparison with scenario 2, we observe that the mean FTP
transfer time is almost reduced by 50 percent when the number of VoIP users is 100
for scenario 3. This is largely due to the fact that a higher percentage of bandwidth is
assigned to the FTP users in this scenario.
Figure 4-15: Mean FTP transfer time vs No. of VoIP users
The throughput for the FTP traffic in downlink is shown in Figure 4.16. In this
scenario, there is no change in the FTP throughput when the number of VoIP users
is increased from 80 to 100 for case 1 whereas there is a slight decrease in case 3.
This behaviour is due to the same concept explained in Scenario 2. For Strict Priority scheduling, there is a significant drop as in scenario 2 and hence there is a severe
degradation in FTP throughput when compared to other two scheduling strategies.
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Results
Figure 4-16: FTP Throughput vs No. of VoIP users
4-5-4
Summary of results for Scenario 3
In this scenario, the performance of VoIP when mapped to Gold bearer is analysed for
the three different scheduling algorithms. Within the acceptable QoS thresholds (100
ms end to end delay and 2 percent packet loss), the number of VoIP users for case 1
and case 3 is 65 Users whereas for case 2 it remains the same as in previous scenario at
100 users. The main reason behind the drop in the capacity of VoIP users is due to the
fact that the VoIP users are mapped to AF bearer in the transport network which has
a lower bandwidth limit compared to the previous scenario where the EF class had an
higher bandwidth limit. The FTP throughput achieved in this scenario is much better
compared to the previous scenario when the network is congested with VoIP users for
case 1 and case 3.
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4-6 Scenario 4
4-6
43
Scenario 4
In this scenario, the VoIP users are mapped into both Platinum and Gold bearers i.e.
the VoIP users are split into two groups mapped into high priority Platinum bearer
and normal priority Gold bearer. The description of the scenario is as follows.
4-6-1
Description of the scenario
In the transport network the high priority VoIP users are mapped to EF class and
normal priority VoIP users are mapped to AF 11 class. The FTP users are mapped
into Bronze bearer (QCI 9). The scheduling strategy is used such that for high priority
VoIP users, Strict Priority scheduling is used. The remaining available bandwidth is
shared between the normal priority VoIP users and FTP users using the Weighted Fair
scheduling algorithm. The scenario is divided into four cases according to the number
of high priority and normal priority VoIP users. In all the cases, the weights for normal
priority VoIP users and FTP users are set to 6 and 3 respectively. The Table 4.1 shown
below gives the details of each of the four cases.
Application Type
VoIP (High Priority)
VoIP (Normal Priority)
Case 1
30
40
Case 2
20
60
Case 3
30
60
Case 4
20
80
Table 4-2: Scenario Description
4-6-2
Analysis of Results
Packet end to end delay and PDV
The Figures 4.17 and 4.18 shows the packet end to end delay for the premium and
normal VoIP user. In Figure 4.17, we see that there is no significant change in the end
to end delay for the four cases. This is mainly due to the fact that for premium users,
we use strict priority scheduling and hence they are always served first even during the
time of congestion. There is also no significant PDV due to the same reason and hence
PDV is not plotted for the premium VoIP users.
In Figure 4.18, the packet end to end delay for the normal VoIP user is shown. The
end to end delay for case 1 is around 80 ms. The peak bandwidth for normal users
when all of them send packet simultaneously is around 3.6 Mbps which is less than the
provisioned bandwidth of 4.8 Mbps. Hence there is very less waiting time in the core
network for the VoIP packets which explains the less packet end to end delay. Due
to the same reason, there is no variation in packet delay as seen in Figure 4.19. In
case 2 and case 3, the end to end delay is around 100 ms. This is a significant change
compared to the case 1 but still the value is within the acceptable limits of end to end
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Results
Figure 4-17: Packet end to end delay for high priority VoIP users
Figure 4-18: Packet end to end delay for normal priority VoIP users
delay. For both the cases, the peak bandwidth is around 5.4 Mbps which is higher than
the provisioned bandwidth. Hence the packers are buffered in the EPC node, which
leads to an increase in the end to end delay. For case 4, the delay is 140 ms which
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4-6 Scenario 4
45
is beyond the acceptable value. The peak bandwidth in this case is around 7.2 Mbps
which is well beyond the provisioned bandwidth. This leads to a congestion in the EPC
node leading to more waiting times. The values of PDV for cases 2 and 3 are around
0.5 ms whereas for case 4 the PDV is around 2 ms as shown in Figure 4.19.
Figure 4-19: Packet delay variation
PLR and MOS
The packet loss rate for the premium VoIP users is null since they are served using
strict priority. As there is no packet loss in this case and the delay is within the limits,
the value of MOS is greater then 4.3 which implies there is very high quality of experience for premium users. The PLR and MOS for premium VoIP users follow the same
pattern as in Figures 4.6 and 4.7.
The Figure 4.21 shows the PLR for normal VoIP users. There is no packet loss for
case 1 as the congestion scenario is not yet reached and the bandwidth is within the
limits. For case 2 and case 3, there is a packet loss of around 1.5 percent which is
still within the acceptable value of 2 percent. For case 4, there is a large packet loss
of around 10 percent which implies that the calls are dropped. The Figure 4.22 shows
the MOS for the normal VoIP users. It is clear from the Figure 4.22 that the MOS for
cases 2 and 3 are less compared to case 1 but still within the acceptable value.
Prasanna Gururaj
Raghavendrarao
46
Results
Figure 4-20: Packet Loss Rate
Figure 4-21: Mean Opinion Score
The Figure 4.23 shows the FTP transfer time for all the four cases. It is seen that
for case 1 the transfer time is 24 seconds and for case 2 the transfer time is around
27 seconds. The total number of VoIP users in the network is 70 users (premium +
normal) for case 1. In Figure 4.15, at the same point, it is seen that the transfer time
Prasanna Gururaj
Raghavendrarao
4-6 Scenario 4
47
is around 26 seconds in Scenario 3 . Similarly the total number of VoIP users in case 2
is 80 and at the same point the transfer time seen in Figure 4.15 is 30 seconds. Hence
in these two cases for the same number of VoIP users, there is a marginal decrease in
the FTP transfer time compared to scenario 3.
In Figure 4.23, the FTP transfer time for case 3 and case 4 are 50 and 65 respectively.
In comparison with Figure 4.15, the transfer time is significantly higher for both the
cases. This is due to the fact that in case 3 in Figure 4.23, there are more number of
VoIP users (premium + normal = 90) which are within the acceptable limits of QoS
compared to 4.15 and hence the VoIP capacity is increased at the expense of the FTP
throughput. In case 4, the number of normal VoIP users are very high which leads
to more congestion in the core network and hence the throughput for both the VoIP
and FTP users are significantly affected leading to a poor performance for both the
services.
Figure 4-22: FTP Transfer time
Prasanna Gururaj
Raghavendrarao
48
Results
4-7
Comparison of Scenarios
In this section, a comparison of the scenarios 2, 3 and 4 is done for a better understanding of the results. The table 4.2 gives the number of satisfied VoIP users in each
scenario which are within the acceptable limits of 100 ms delay and 2 percent packet
loss. The corresponding FTP transfer time at this point is also shown in Table 4.2.
The number of voice users in Scenario 4 shown in the Table 4.3 is the total number of
voice users (premium+normal).
Scenario Name
Scenario 2
Scenario 3
Scenario 4
case 1
80
65
70
No. of VoIP users

case 2 case 3 case 4
100
80
X
100
65
X
80
90
20
Corresponding FTP Transfer time

case 1 case 2 case 3
case 4
38
130
40
X
24
130
24
X
24
26
50
65
Table 4-3: Number of satisfied VoIP users
From the table 4.2, it is seen that the Scenario 4 has a better capacity in terms of
number of VoIP users compared to the other two scenarios. This is explained as follows.
In Scenario 2, the emphasis is more on increasing the VoIP capacity at the cost of increase in transfer time for FTP users when there is congestion in the network. Though
there no delay guarantees for FTP users in this scenario, there should not be total
degradation of throughput for FTP as in case 2. Hence in Scenario 2, the maximum
number of VoIP users that can be supported with acceptable QoS limits for voice is
80. Comparing this with Scenario 4, there are 90 VoIP users that can be supported
within the acceptable QoS limits. Hence there is about 10 percent increase in the VoIP
capacity at a cost of increase transfer time in Scenario 4 when compared to Scenario 2.
The Scenario 3 has strict bounds on the QoS of data traffic i.e. the mean FTP transfer
time is not increased by more than 20 percent when there is congestion in the network.
Hence in Scenario 3, the maximum number of VoIP users that can be supported while
ensuring that the increase in delay for data traffic is within the bounds is 65. Comparing the Scenario 3 with Scenario 4, we see that for the same criteria i.e increase in
delay nor more than 20 percent the number of VoIP users that can be supported is 80.
Hence there is a 20 percent increase in the VoIP capacity in Scenario 4 when compared
to Scenario 3.
Hence by grouping of VoIP users into different levels of priority an increase in capacity
is achieved when compared to mapping them to a single specific service class.
Prasanna Gururaj
Raghavendrarao
Chapter 5
Technical Details of Voice over LTE
via IMS based solution - An Operator
Perspective
5-1
Introduction
This chapter gives an overview on the various technical impacts of VoLTE solution
on the existing CS and PS networks. The VoLTE solution will introduce the voice
functionality in the LTE network using the new IMS framework which is widely being
accepted as the long term solution for supporting voice in LTE network. IMS based
voice is widely seen as the better solution in the current scenario capable of delivering
voice in the LTE network. Hence, operators worldwide or considering the deployment
of an IMS based solution. This chapter explains the technical details of VoLTE solution
from an industry perspective.
5-2
VoLTE Architecture
The Figure 5.1 shows the important elements in the VoLTE architecture. The architecture shows a scenario where the LTE network is deployed as a separate PS network.
The IMS network is deployed as an overlay to the LTE network and it provides the
basic call origination/termination functionalities as well as value added services like
Presence, Instant messaging etc. The user after obtaining an IP address from the LTE
network performs a registration operation with the IMS network which enables the
users to get access to the basic services like voice and also other value added services
based on subscription. The Table 5.1 gives the relevant protocols and interfaces for
VoLTE solution [15].
Prasanna Gururaj
Raghavendrarao
50
Technical Details of Voice over LTE via IMS based solution - An Operator Perspective
Figure 5-1: VoLTE Architecture
Nodes
MME HSS
PCRF P-CSCF
I/S-CSCF HSS
I/S-CSCF AS
P-CSCF I/S-CSCF
Interfaces
S6a
Rx
Cx
ISC
Mw
Protocols
Diameter
Diameter
Diameter
SIP
SIP
Table 5-1: VoLTE Relevant Interfaces and Protocols
In the above architecture, LTE is deployed as a standalone network and there is no

integration with the 2G/3G networks. During the initial roll out of LTE, the coverage
will be minimum and hence there should be some way of integrating the LTE network
with the 2G/3G network. This integration is quite challenging owing to the fact that
LTE has a completely packet switched architecture. As stated earlier, the voice in LTE
is carried as VOIP packets. When the user is roaming outside of LTE coverage i.e. in
2G/3G domains the voice call needs to be switched from VoIP based to legacy TDM
Prasanna Gururaj
Raghavendrarao
5-3 Options for integrating LTE with existing CS/PS networks
51
based call. Hence there are a few possible solutions for integrating LTE and 2G/3G
networks which are explained in the following sections.
5-3
Options for integrating LTE with existing CS/PS networks
During initial LTE deployments, the coverage is going to be limited and hence it is
required to integrate the LTE network with the existing 2G/3G network. When loss of
LTE coverage is detected, the user should be able to attach to 2G/3G network. If there
is an ongoing voice call in the LTE network, then a handover needs to be performed to
the 2G/3G network without interruption of voice call. There are two architectures for
integration of LTE with 2G/3G networks which are seen as a possible approaches for
achieving voice call continuity between LTE and 2G/3G networks. They are as follows:
Independent PS based solution.
Enhanced Single Radio Voice call continuity(SRVCC) / IMS Centralized Services(ICS).
5-3-1
Independent PS based solution
In PS based solution, voice over IMS is implemented in both the LTE and 3G networks. During loss of LTE coverage, a PS handover is performed towards the 3G
network thereby providing seamless mobility between LTE and 3G networks. Thus
both the voice and data sessions that are active in the LTE network are simultaneously
transferred to the 3G network, there by preventing loss of voice/data during the loss of
LTE coverage. The Figure 5.2 shows the architecture of PS based solution. The handover procedure is defined in [16]. The overview of the procedure is briefly summarized
below:
The user is initially attached to the LTE network and a voice call is established
via IMS in the LTE network.
When loss of LTE coverage is detected, the E-Node B in the LTE network initiates
an handover towards the MME which then forwards the same to the SGSN. The
MME also separates the voice bearers from the non voice bearers and performs a
mapping between the LTE bearer and 3G PDP context.
The target SGSN reserves the necessary resources in the 3G network and also
creates a session request towards the serving gateway. The SGSN reverts back to
the MME on successful completion of the reservation procedure.
The MME in the LTE network then performs the handover execution procedure
by sending handover command towards the E-Node B
The E-Node B then sends a handover command to the UE containing the radio
access network parameters of the target 3G network.
Prasanna Gururaj
Raghavendrarao
52
Figure 5-2: LTE-3G Integrated architecture
The UE can continue the voice session after successful completion of the handover
procedure.
Advantages and Limitations
The major advantage is, it is seen as the simplest solution for integrating LTE with
3G network as it involves minimum changes in terms of network architecture. The
existing PS network for 3G can be reused easily without any major upgrades. 3G
network has PS based capabilities and hence handover of voice from LTE to 3G can be
accomplished easily via IMS without significant interruption.
The independent PS based solution cannot be taken as a target solution as it requires

complete coverage of 3G network. Since both voice and data are carried in the 3G PS
network, higher bandwidth is required. Hence the advanced release of UMTS which is
HSPA+ is needed to support high data rates for carrying both data and voice in the
network simultaneously. The existing CS network is not reused in this scenario which
can be a major factor in the future when the legacy networks like 2G become obsolete.
Prasanna Gururaj
Raghavendrarao
5-3 Options for integrating LTE with existing CS/PS networks
5-3-2
53
Enhanced Single Radio Voice call continuity (SRVCC) / IMS Centralized

Services (ICS)
The integration of the LTE network with the 2G/3G network based on Enhanced Single
Radio Voice Call Continuity (SR-VCC)/IMS Centralized Services architecture is shown
in Figure 5.4
Figure 5-3: SRVCC/ICS Architecture
Enhanced SRVCC
In the Enhanced SR-VCC based approach defined in [16], the call control of the LTE
network lies within IMS network. The Service Control and Centralization Application
server (SCC AS) in the IMS network is the responsible element for anchoring the call
in IMS. The SIP signalling messages from the user attached to the LTE network and
the destination user is relayed via the SCC AS. The mobile is also assigned a Session
Transfer Number for SRVCC (STN-SR) by the SCC AS during the initial registration
and is used during handover of the call from the LTE to 2G/3G network. The handover
procedure defined in [17] for enhanced SRVCC is briefly explained as follows:
The E-Node B initiates a handover procedure towards the MME when a loss of
LTE coverage is detected based on the measurement reports from the UE.
The MME splits the voice and data bearers and initiates a handover procedure
towards the Enhanced MSC server in the CS domain.
Prasanna Gururaj
Raghavendrarao
54
The Enhanced MSC server in the CS domain is responsible for reservation of

bearers in the CS domain. This is done by forwarding the handover request
message to the target MSC server to which the LTE user will be registered in the
CS domain.
In addition, the enhanced MSC server initiates the transfer of the call in the
IMS domain, by using the STN-SR. The SCC AS in the IMS domain executes
the session transfer procedure in IMS domain and the media bearer is switched
towards the CS domain.
After a successful completion of the access transfer procedure in the CS domain,
the Enhanced MSC server indicates the successful completion of the procedure to
the MME.
The MME sends a handover command to the UE via the E-Node B and the UE
attaches to the CS domain by following the CS domain attach procedure and the
call flow is switched to the MSC Server/Media Gateway in the CS domain.
IMS Centralized Services
IMS Centralized services (ICS) defined in [17] is an extension of the Enhanced - SRVCC
and leads towards complete integration of 2G/3G networks with LTE network. The
ICS architecture is same as shown in Figure 5.4. The difference lies in the fact that
in ICS based approach calls that are originating in 2G/3G network i.e. calls from
legacy mobiles are also anchored at IMS network. Since, the call control for both the
CS (2G/3G) and PS (LTE) domains is within the IMS, seamless handover of users
between the 2G/3G and LTE networks can be facilitated easily.
Advantages and Limitations
The main advantage of using the SRVCC/ICS based approach is it facilitates the
integration of the legacy CS networks with the LTE network. During the initial roll
out of LTE when the coverage is minimum, the SRVCC based approach enables easy
roaming between LTE and 2G/3G domains. It also enables the users to experience
the same services independent of the access network to which the user is connected.
Lastly, IMS Centralized Services approach is the way towards the future when the
legacy networks like 2G/3G become obsolete and a common infrastructure would save
lot of costs in operating these networks.
The drawback of SRVCC/ICS based approach is, it requires significant upgrades
in the existing CS networks. In the CS domain, elements like MSC server are to
be upgraded which involve significant costs. The other major limitation is the long
handover time when a user moves from LTE to 2G domain which causes a significant
disruption when there is an ongoing call in the LTE network.
Prasanna Gururaj
Raghavendrarao
Chapter 6
Conclusion
6-1
Conclusion
In this thesis, the performance of voice over LTE is analysed when the transport network
is congested with data traffic. The analysis was carried out using OPNET simulation
tool. The LTE model in OPNET had significant limitations in the classification of bearers in the transport network. This led to a situation where there was no prioritization
of the bearers in the transport network. Hence to being with, various functions in the
process models of the E-Node B and EPC were modified to achieve proper classification
of bearers in the transport network. The modification was performed such that each
bearer in the LTE network will be mapped to a specific DSCP in the IP header. This
enabled us to do classification of IP packets for different services like voice and FTP in
the transport network.
The importance of classification of voice and data traffic in the transport network
is realized from the results of Section 4.2. Without proper classification, we see that
there is a 50 percent increase in the packet end to end delay for voice even when
there is no congestion in the transport network. In Sections 4.3 and 4.4, the role of
scheduling algorithms on the performance of voice and data was analysed. We see that
the capacity for voice users is higher when there is absolute priority for voice in the
transport network. But there is a drawback of using Strict Priority for voice, as there
is a significant degradation in the performance of data traffic at times of congestion.
The Weighted Fair and Weighted Round Robin algorithms were used to overcome
this drawback. A comparative analysis was carried out to understand the performance
of voice when these scheduling algorithms are implemented in the transport network.
We see that the capacity of voice users in the network is reduced when the voice bearers
are mapped to a AF service class in the transport network. This is mainly due to the
Prasanna Gururaj
Raghavendrarao
56
Conclusion
reduced bandwidth allocated for voice users in the AF class than in the EF class. There
is a trade-off between the performance of voice and data traffic, depending on the type
of classification in the transport network.
Finally, we present a different approach where voice users are mapped into two priority levels and mapped to both EF and AF classes in the transport network. With this
approach, we see that we can add more voice users in the network within the acceptable
QoS levels than mapping voice into a single EF or AF service class in the transport network. At times of high congestion, there is a significant reduction in the performance
of voice users belonging to the AF service but in a more controlled manner. Such an
approach enables the operators to offer different levels of service quality for voice users.
It also enables the operators to drop calls belonging to normal service class when there
is heavy congestion the core network.
6-2
Future Work
In our thesis, we have used only the VoIP and FTP traffic models to analyse the performance of VoIP in the network. LTE supports very high data rates and hence services
like Video streaming, Interactive gaming can also be used by the mobile users. In the
current LTE specifications, there are different bearers that have defined for each service
as seen in Table 2.1. But the mapping of these bearers to IP based QoS is also important for classification in the transport network. In the future, when there are no CS
based networks like GSM and voice is carried entirely over PS based networks like LTE,
there will be a significant impact on the delivery of voice when multiple services are
present in the network. In such scenarios, when there is multiple level of classification
for different types of services, each type of service has different QoS requirements and
mapping them to IP based QoS in the transport network needs to be done carefully.
This will be an interesting area to investigate as the role of scheduling algorithms in
the transport network become much more important owing to the fact that over provisioning of bandwidth for one type of service has a direct impact on the performance
of other service.
The usage of admission control is another area that needs to be investigated. Most
of the studies that have been done in this area are focussed on the radio network i.e.
the use of admission control is studied when there is congestion in the radio network.
But when admission control takes into account the availability of resources in both the
radio and core networks, efficient link usage in the transport network can be achieved
without increasing the link capacity in the transport network.
Prasanna Gururaj
Raghavendrarao
Bibliography
[1] 3GPP Technical Specification 23.272, "Circuit Switched (CS) fallback in Evolved
Packet System (EPS)", Stage 2 (Release 10); http://www.3gpp.org, 2011.
[2] Alcatel-Lucent Strategic White Paper, "Options for Providing Voice over LTE and
Their Impact on the GSM/UMTS Network"; www.alcatel-lucent.com, August 2009.
[3] Siomina, I.; Wanstedt, S.; , "The impact of QoS support on the end user satisfaction in LTE networks with mixed traffic," IEEE 19th International Symposium on
Personal, Indoor and Mobile Radio Communications, pp.1-5, 15-18 Sept. 2008.
[4] Zaki, Y.; Weerawardane, T.; Gorg, C.; Timm-Giel, A., "Multi-QoS-Aware Fair
Scheduling for LTE," IEEE 73rd Vehicular Technology Conference (VTC Spring)
vol., no., pp.1-5, 15-18 May 2011.
[5] Puttonen, J.; Henttonen, T.; Kolehmainen, N.; Aschan, K.; Moisio, M.; Kela, P.; ,
"Voice-Over-IP Performance in UTRA Long Term Evolution Downlink," IEEE Vehicular Technology Conference, vol., no., pp.2502-2506, 11-14 May 2008.
[6] Yasir Zaki, Nokila Zahariev, Thushara Weerawardane, Carmelita Grg and Andreas
Timm-Giel, "Optimized Service Aware LTE MAC Scheduler: Design, Implementation
and Performance Evaluation", OPNET workshop, Washington, D.C., August 29September 1, 2011.
[7] Li, X.; Toseef, U.; Weerawardane, T.; Bigos, W.; Dulas, D.; Goerg, C.; Timm-Giel,
A.; Klug, A.; , "Dimensioning of the LTE S1 interface," Third Joint IFIP Wireless
and Mobile Networking Conference (WMNC), vol., no., pp.1-6, 13-15 Oct. 2010.
[8] Ekstrom, H.; , "QoS control in the 3GPP evolved packet system," IEEE Communications Magazine , vol.47, no.2, pp.76-83, February 2009.
[9] 3GPP Technical Specification 23.203, "Policy and charging control architecture (Release 11)", www.3gpp.org, 2012
Prasanna Gururaj
Raghavendrarao
58
Bibliography
[10] 3GPP Technical Specification 23.228, "IP Multimedia Subsystem (IMS); Stage 2
(Release 11) http://www.3gpp.org, 2012.
[11] S. Blake, D. Black, M. Carlson, E. Davies, Z. Wang, W. Weiss, "An architecture
for Differentiated Services", "Request for Comments 2475, Internet Engineering Task
Force", December 1998.
[12] A. Demers, S. Keshav, and S. Shenker "Analysis and simulation of a fair queueing
algorithm", In Symposium proceedings on Communications architectures and protocols ", ACM, New York, NY, USA, 1989.
[13] OPNET Modeller, www.opnet.com accessed on December 2011.
[14] ITU-T Recommendation G.107, "The E-Model, a computational model for use in
transmission planning", 2011.
[15] 3GPP Technical Specification 23.401, "General Packet Radio Service (GPRS) enhancements for Evolved Universal Terrestrial Radio Access Network (E-UTRAN)
access " Stage 2 (Release 10), http://www.3gpp.org, 2011.
[16] 3GPP Technical Specification 23.216, " Enhanced Single Radio Voice Call Continuity (SRVCC),Stage 2(Release 11) " http://www.3gpp.org;, 2012.
[17] 3GPP Technical Specification 23.292, " IMS Centralized Services Stage 2(Release
11) " http://www.3gpp.org, 2012.
Prasanna Gururaj
Raghavendrarao
Glossary
List of Acronyms
3GPP
Third Generation Partnership Project
ARP
Allocation and Retention Priority
BGCF
Breakout Gateway Control Function
CS
Circuit Switched
Diffserv
Differentiated Services
E-UTRAN
(Evolved Universal Terrestrial Radio Access Network)
EPC
Evolved Packet Core
FIFO
First In First Out
GTP
GPRS Tunnelling Protocol
HSS
Home Subscriber Server
HSPA
High Speed Packet Access
I-CSCF
Interrogating Call Session Control Function
ICS
IMS Centralized Services
IMS
IP Multimedia Subsystem
LTE
Long Term Evolution
MOS
Mean Opinion Score
MSC
Mobile Switching Centre
MIMO
Multiple Input Multiple Output
MGCF
Media Gateway Control Function
Prasanna Gururaj
Raghavendrarao
60
Glossary
MME
Mobility Management Entity
MRF
Media Resource Function
PDN
Packet Data Network
PCRF
Proxy and Charging Rules Function
P-CSCF
Proxy Call Session Control Function
PDV
Packet Delay Variation
PS
Packet Switched
PLR
Packet Loss Rate
QCI
QoS Class Identifier
QoE
Quality of Experience
RCS
Rich Communication Suite
RTP
Real Time Protocol
S-CSCF
Serving Call Session Control Function
SCC AS
Service Control and Centralization Application Server
TFT
Traffic Flow Template
VoLTE
Voice over LTE via IP Multimedia Subsystem
WRR
Weighted Round Robin
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Raghavendrarao

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Voice over LTE

Master of Science Thesis

Wireless and Mobile Communications Group

Master of Science Thesis

Voice over LTE

For the degree of Master of Science in

Faculty of Electrical Engineering, Mathematics and Computer Science

Master of Science Thesis

Master of Science Thesis

Delft University of Technology

ir. Gerard Fossung (KPN)

dr. R.R. Venkatesha Prasad

dr.ir. Bert Jan Kooij

Master of Science Thesis

Master of Science Thesis

Circuit Switch(CS) fallback . . . . . . . . . . . . . . . . . . . . .

Voice over LTE via IP Multimedia Subsystem (VoLTE) . . . . . . .

Weighted Round Robin (WRR) Scheduling . . . . . . . . . . . . .

Weighted Fair Scheduling . . . . . . . . . . . . . . . . . . . . . .

Motivation for the Thesis

Master of Science Thesis

3-2-2 Issues in LTE model . . . . . . . . .

Independent PS based solution . . . . . . . . . . . . . . . . . . .

Master of Science Thesis

Master of Science Thesis

Master of Science Thesis

1-1 Circuit Switch Fallback [2] . . . . . . . . . . . . . . . . . . . . . . . . . .

1-2 VoLTE [2] . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

2-1 LTE Network Architecture . . . . . . . . . . . . . . . . . . . . . . . . . .

2-3 IMS Network Architecture . . . . . . . . . . . . . . . . . . . . . . . . . .

3-1 LTE Network Model . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

Voice Packet End to End Delay vs No. of VoIP

Master of Science Thesis

5-1 VoLTE Architecture . . . . . . . . . . . . . . . . . . . . . . . . . . . . .

5-3 SRVCC/ICS Architecture . . . . . . . . . . . . . . . . . . . . . . . . . .

Mean Opinion Score vs No. of VoIP users . . . . . . .

Master of Science Thesis

2-1 EPS QOS Bearer Definitions [9] . . . . . . . . . . . . . . . . . . . . . . .

2-2 Assured Forwarding Drop Precedence Classification . . . . . . . . . . . . .

3-1 EPS Bearer to DSCP Mapping . . . . . . . . . . . . . . . . . . . . . . .

4-1 MOS satisfaction level . . . . . . . . . . . . . . . . . . . . . . . . . . . .

5-1 VoLTE Relevant Interfaces and Protocols . . . . . . . . . . . . . . . . . .

Master of Science Thesis

Master of Science Thesis

Master of Science Thesis

Prasanna Gururaj Raghavendrarao

Master of Science Thesis

Long Term Evolution (LTE) is a fourth generation technology which is standardized

Solutions for Supporting Voice over LTE

Circuit Switch(CS) fallback

Figure 1-1: Circuit Switch Fallback [2]

Voice over LTE via IP Multimedia Subsystem (VoLTE)

In this solution, voice functionality is provided by the IP Multimedia Subsystem (IMS).

Master of Science Thesis

1-2 Motivation for the Thesis

Figure 1-2: VoLTE [2]

Motivation for the Thesis

The Differentiated services architecture (Diffserv) which is commonly used in IP based

Master of Science Thesis

1-4 Related Work

Organization of the Thesis

The thesis is organized as follows.

Master of Science Thesis

LTE Network Architecture