masters thesis voice over lte.pdf

Voice over LTE

PRASANNA GURURAJRAGHAVENDRARAO

Mastero

fScie

nceTh

esis

Wireless and Mobile Communications Group

Department of Telecommunications

Faculty of Electrical Engineering, Mathematics andComputer Science

Delft University of Technology

II

Prasanna GururajRaghavendrarao

Master of Science Thesis

III

Voice over LTE


For the degree of Master of Science inWireless and Mobile Communications Group (WMC)

at Department of Telecommunicationsat Delft University of Technology


29.6.2012

Faculty of Electrical Engineering, Mathematics and Computer ScienceDelft University of Technology

Delft, The Netherlands

Master of Science Thesis Prasanna GururajRaghavendrarao

IV



Delft University of TechnologyDepartment of

Telecommunications

The undersigned hereby certify that they have read and recommend to the Faculty ofElectrical Engineering, Mathematics and Computer Science for acceptance a thesis

entitledVoice over LTE

byPrasanna GururajRaghavendrarao

in partial fulfillment of the requirements for the degree ofMaster 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 thatis gaining widespread attention due to its high data rates and improved Quality ofService (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 supportof voice over LTE has lot of challenges owing to the fact that both voice and data trafficare to be carried over the same radio and core networks. The optimum usage of re-sources in the radio network is of high importance as there is a growing need to improvethe capacity at reduced cost. The transport network is another key area that needsto be carefully planned according to the capacity of the radio network. Differentiationand scheduling of resources in the transport network plays a key role in guaranteeinggood end to end performance for both voice and data services.

In this thesis, the impact of differentiation and scheduling of resources in the trans-port network on the end to end performance of voice over LTE is investigated. Theresults indicate that without proper prioritization and scheduling of resources in thetransport network , the performance of voice is severely affected when the transportnetwork is congested with data traffic. To overcome this scenario, we prioritize voiceover data traffic and analyse its performance for different transport network schedulingalgorithms. From the results, it is clear that with proper classification and scheduling ofresources in the transport network, significant increase in voice capacity is observed. Onthe other hand, by totally prioritizing voice, performance of the data traffic is affectedto a large extent. Hence, to achieve a balance, voice users are classified into differentpriority 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 simulationtool.


ii



Table of Contents

Acknowledgements xi

1 Introduction 11-1 Solutions for Supporting Voice over LTE . . . . . . . . . . . . . . . . . . 1

1-1-1 Circuit Switch(CS) fallback . . . . . . . . . . . . . . . . . . . . . 11-1-2 Voice over LTE via IP Multimedia Subsystem (VoLTE) . . . . . . . 2

1-2 Motivation for the Thesis . . . . . . . . . . . . . . . . . . . . . . . . . . 31-3 Problem Definition . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 41-4 Related Work . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 51-5 Organization of the Thesis . . . . . . . . . . . . . . . . . . . . . . . . . . 5

2 Background 72-1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 72-2 LTE Network Architecture . . . . . . . . . . . . . . . . . . . . . . . . . . 72-3 QoS Architecture in LTE . . . . . . . . . . . . . . . . . . . . . . . . . . . 92-4 IMS Network Architecture . . . . . . . . . . . . . . . . . . . . . . . . . . 112-5 Differentiated Services Architecture . . . . . . . . . . . . . . . . . . . . . 122-6 Scheduling Strategies . . . . . . . . . . . . . . . . . . . . . . . . . . . . 13

2-6-1 Strict Priority Scheduling . . . . . . . . . . . . . . . . . . . . . . 132-6-2 Weighted Round Robin (WRR) Scheduling . . . . . . . . . . . . . 142-6-3 Weighted Fair Scheduling . . . . . . . . . . . . . . . . . . . . . . 14

3 Simulation Model 153-1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 153-2 Introduction to OPNET Modeller . . . . . . . . . . . . . . . . . . . . . . 15

3-2-1 Overview of LTE Model in OPNET . . . . . . . . . . . . . . . . . 15


iv Table of Contents

3-2-2 Issues in LTE model . . . . . . . . . . . . . . . . . . . . . . . . . 183-3 Changes in LTE model . . . . . . . . . . . . . . . . . . . . . . . . . . . . 18

3-3-1 LTE S1 process model In E-Node B . . . . . . . . . . . . . . . . . 183-3-2 LTE S1 NAS Process model in EPC . . . . . . . . . . . . . . . . . 203-3-3 GTP Process model in E-Node B and EPC . . . . . . . . . . . . . 20

3-4 Simulation Environment . . . . . . . . . . . . . . . . . . . . . . . . . . . 223-4-1 Mobile Node . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 223-4-2 E-Node B . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 233-4-3 IMS Model . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 253-4-4 Application Configuration . . . . . . . . . . . . . . . . . . . . . . 26

4 Results 274-1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 274-2 QoS parameters for Voice . . . . . . . . . . . . . . . . . . . . . . . . . . 274-3 Scenario 1 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 28

4-3-1 Description of the Scenario . . . . . . . . . . . . . . . . . . . . . 284-3-2 Analysis of results . . . . . . . . . . . . . . . . . . . . . . . . . . 29

4-4 Scenario 2 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 314-4-1 Description of the Scenario . . . . . . . . . . . . . . . . . . . . . 314-4-2 Analysis of results . . . . . . . . . . . . . . . . . . . . . . . . . . 314-4-3 Impact on FTP traffic . . . . . . . . . . . . . . . . . . . . . . . . 354-4-4 Summary of the Results for Scenario 2 . . . . . . . . . . . . . . . 36

4-5 Scenario 3 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 374-5-1 Description of the Scenario . . . . . . . . . . . . . . . . . . . . . 374-5-2 Analysis of results . . . . . . . . . . . . . . . . . . . . . . . . . . 374-5-3 Impact on FTP traffic . . . . . . . . . . . . . . . . . . . . . . . . 404-5-4 Summary of results for Scenario 3 . . . . . . . . . . . . . . . . . . 42

4-6 Scenario 4 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 434-6-1 Description of the scenario . . . . . . . . . . . . . . . . . . . . . . 434-6-2 Analysis of Results . . . . . . . . . . . . . . . . . . . . . . . . . . 43

4-7 Comparison of Scenarios . . . . . . . . . . . . . . . . . . . . . . . . . . . 48

5 Technical Details of Voice over LTE via IMS based solution - An OperatorPerspective 495-1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 495-2 VoLTE Architecture . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 495-3 Options for integrating LTE with existing CS/PS networks . . . . . . . . . 51

5-3-1 Independent PS based solution . . . . . . . . . . . . . . . . . . . 515-3-2 Enhanced Single Radio Voice call continuity (SRVCC) / IMS Cen-

tralized Services (ICS) . . . . . . . . . . . . . . . . . . . . . . . . 53



Table of Contents v

6 Conclusion 556-1 Conclusion . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 556-2 Future Work . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 56

Glossary 59List of Acronyms . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 59List of Symbols . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 60


vi Table of Contents



List of Figures

1-1 Circuit Switch Fallback [2] . . . . . . . . . . . . . . . . . . . . . . . . . . 21-2 VoLTE [2] . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3

2-1 LTE Network Architecture . . . . . . . . . . . . . . . . . . . . . . . . . . 82-2 QoS Architecture in LTE [8] . . . . . . . . . . . . . . . . . . . . . . . . . 92-3 IMS Network Architecture . . . . . . . . . . . . . . . . . . . . . . . . . . 11

3-1 LTE Network Model . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 163-2 Data Flow in LTE Network [13] . . . . . . . . . . . . . . . . . . . . . . . 163-3 Protocol Architecture . . . . . . . . . . . . . . . . . . . . . . . . . . . . 173-4 GTP Encapsulated Packet . . . . . . . . . . . . . . . . . . . . . . . . . . 173-5 E-Node B Node Model . . . . . . . . . . . . . . . . . . . . . . . . . . . 193-6 lte _ s1 Process model in E-Node B . . . . . . . . . . . . . . . . . . . . 193-7 Node Model in EPC . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 203-8 lte _ s1 _ nas Process model in EPC . . . . . . . . . . . . . . . . . . . . 213-9 GTP Process Model . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 223-10 LTE Simulation Network . . . . . . . . . . . . . . . . . . . . . . . . . . . 233-11 Mobile Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 243-12 IMS Proxy session control function configuration attribute . . . . . . . . . 25

4-1 Voice Packet End to End Delay vs No. of VoIP users . . . . . . . . . . . . 294-2 Packet Delay Variation Vs No. of VoIP users . . . . . . . . . . . . . . . . 304-3 End to End Delay Vs No. of VoIP Users . . . . . . . . . . . . . . . . . . 324-4 S1 Delay Vs No. of VoIP Users . . . . . . . . . . . . . . . . . . . . . . . 324-5 PDV vs No. of VoIP users . . . . . . . . . . . . . . . . . . . . . . . . . . 334-6 Packet Loss Rate vs No. of VoIP users . . . . . . . . . . . . . . . . . . . 34


viii List of Figures

4-7 Mean Opinion Score vs No. of VoIP users . . . . . . . . . . . . . . . . . . 344-8 FTP Transfer time vs No. of VoIP users . . . . . . . . . . . . . . . . . . 354-9 FTP Throughput vs No. of VoIP users . . . . . . . . . . . . . . . . . . . 364-10 End to End Delay vs No. of VoIP users . . . . . . . . . . . . . . . . . . . 384-11 S1 delay vs No. of VoIP users . . . . . . . . . . . . . . . . . . . . . . . . 384-12 Packet delay variation vs No. of VoIP users . . . . . . . . . . . . . . . . . 394-13 Packet Loss Rate vs No. of VoIP users . . . . . . . . . . . . . . . . . . . 404-14 Mean Opinion Score vs No. of VoIP users . . . . . . . . . . . . . . . . . . 404-15 Mean FTP transfer time vs No. of VoIP users . . . . . . . . . . . . . . . 414-16 FTP Throughput vs No. of VoIP users . . . . . . . . . . . . . . . . . . . 424-17 Packet end to end delay for high priority VoIP users . . . . . . . . . . . . 444-18 Packet end to end delay for normal priority VoIP users . . . . . . . . . . . 444-19 Packet delay variation . . . . . . . . . . . . . . . . . . . . . . . . . . . . 454-20 Packet Loss Rate . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 464-21 Mean Opinion Score . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 464-22 FTP Transfer time . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 47

5-1 VoLTE Architecture . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 505-2 LTE-3G Integrated architecture . . . . . . . . . . . . . . . . . . . . . . . 525-3 SRVCC/ICS Architecture . . . . . . . . . . . . . . . . . . . . . . . . . . 53



List of Tables

2-1 EPS QOS Bearer Definitions [9] . . . . . . . . . . . . . . . . . . . . . . . 102-2 Assured Forwarding Drop Precedence Classification . . . . . . . . . . . . . 13

3-1 EPS Bearer to DSCP Mapping . . . . . . . . . . . . . . . . . . . . . . . 233-2 E-Node B Configuration Parameters . . . . . . . . . . . . . . . . . . . . . 243-3 VoIP Configuration Parameters . . . . . . . . . . . . . . . . . . . . . . . 26

4-1 MOS satisfaction level . . . . . . . . . . . . . . . . . . . . . . . . . . . . 284-2 Scenario Description . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 434-3 Number of satisfied VoIP users . . . . . . . . . . . . . . . . . . . . . . . 48

5-1 VoLTE Relevant Interfaces and Protocols . . . . . . . . . . . . . . . . . . 50


x List of Tables



Acknowledgements

First, I would like to thanks my supervisors at KPN Dr.ir. Jos Adema and ir. GerrardFossung for their valuable inputs and suggestions during the writing of this thesis. Iwould also like to thank Perry Jackson for giving me an opportunity to do this thesis atKPN and the Mobile Innovation Voice Team for their kind support and encouragementduring 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 thewriting of this thesis. Last but not the least, I would like to thank my family andfriends for their continuous love and support.

Delft Prasanna Gururaj Raghavendrarao29.6.2012


xii Acknowledgements



Chapter 1

Introduction

Long Term Evolution (LTE) is a fourth generation technology which is standardizedin 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 predecessorUMTS which is capable of supporting both the Circuit Switched (CS) as well as PacketSwitched (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 standardizationbodies 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 asfollows:

1-1-1 Circuit Switch(CS) fallback

The Circuit Switch fallback solution defined in [1], provides a convenient way in reusingthe existing GSM/UMTS network to support voice in LTE network. This solution isstandardized in [1] and provides the operators with flexibility to roll out LTE as a dataonly 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/UMTSnetwork during the initial registration procedure. This combined registration is facili-tated by the Mobility Management Entity(MME) in the LTE network which performsthe registration in the 2G/3G network on behalf of the user. During the initiation of


2 Introduction

Figure 1-1: Circuit Switch Fallback [2]

the voice call by the user, the MME redirects the request towards the MSC server inthe CS domain. On successful reservation of the resources in the CS domain for thecall, the MSC server shall respond to the MME on the status of the request. TheMME then instructs the E-Node B to request the user to perform a handover to theGSM/UMTS network. The ongoing data session for the user in the LTE network issuspended if the destination network is a GSM network. If the destination network isan UMTS network, then a separate handover of the existing data bearers from LTE toUMTS 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 thehandover procedure and disruption of data transmission throughout the duration of thevoice call when the user falls back to a GSM network. This solution can be used duringthe initial roll out when LTE is more used for high speed data and voice is completelyhandled by legacy circuit switched networks. Hence CS fallback is being seen only asa 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 shownin Figure 1.2. The IMS network is mainly used to provide all the basic services for voice



1-2 Motivation for the Thesis 3

that are provided by the existing CS networks. In addition, it also provides enhancedmultimedia 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 ratherthan relying on the existing CS networks for supporting voice feature. The IMS networkis also capable of integrating with the legacy 2G/3G networks and thus can supportvoice call continuity even when the subscriber moves out of LTE coverage. Hence, thesubscriber 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 providingenhanced services to the LTE network and also supports integration with the existing2G/3G networks.

1-2 Motivation for the Thesis

The VoLTE solution mentioned in the above section is widely being considered fordeployment by operators across the world as it provides simultaneous support of bothvoice and data in the LTE network. In VoLTE, voice is carried in the LTE network asVoice over IP (VoIP) packets. Hence the VoLTE architecture is significantly differentfrom the 2G/3G networks which have distinct CS capabilities for voice. IP basednetworks are mainly designed for best effort services which do not provide any strictguarantees on the quality of service demands of the various services that are offeredto the users. In legacy networks like GSM/UMTS, IP based networks were mainlyused for carrying data services like FTP, HTTP etc. However, with the growth ofmobile broadband technologies like High Speed Packet Access (HSPA) and LTE, thereis 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 andmaintenance of the networks. It also enables the operators to introduce new IP basedservices like Rich Communication Suite (RCS), that can provide the users with improvedquality of experience at reduced costs. Hence the VoLTE solution should provide theusers with a better quality of experience at reduced costs than the existing CS networks.


4 Introduction

In VoIP based networks, the user perceived Quality of Experience (QoE) depends onvarious QoS parameters like delay, jitter, latency, packet loss etc. In addition, sinceboth the data and voice are carried over the same PS network in LTE, there needs to beproper classification among them for scheduling of network resources in the radio andcore network domains. During congestion periods, scheduling algorithms used in boththe radio and core networks for allocation of resources play a critical role in meetingthe stringent delay and packet loss requirements of VoIP service as well as the packetloss requirements of the data service. The capacity of LTE radio network is very highand it can provide a peak cell throughput of around 300 Mbps in the downlink in the4x4 MIMO configuration. This places a direct challenge on the transport network withrespect to the scheduling of the resources. The motivation of this thesis is to studythe effects of congestion in the transport network and to analyse its impact on theperformance of voice in LTE network.

1-3 Problem Definition

In mobile broadband networks like LTE, the high performance of the radio networkcan be realized with proper scheduling of resources for different types of services. Butproper scheduling of resources in the radio network alone is not sufficient to guaranteea good end to end performance. During periods of high congestion, packet losses mightoccur in the transport network which can reduce the overall performance of the servicethat is offered to the user. Hence, the transport network between the radio and corenetworks is another area which needs proper dimensioning and scheduling of resourcesfor various types of services. The transport network is not aware of the QoS architectureof LTE. This implies that the various bearers that are used to classify the services inLTE domain needs to be mapped to IP based QoS techniques.

The Differentiated services architecture (Diffserv) which is commonly used in IP basednetworks 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 toguarantee good end to end performance. The scheduling of resources in the transportnetwork is another area which needs proper attention as the choice of scheduling algo-rithms is pivotal for optimum usage of resources. There are various scheduling strategieslike Weighted Fair scheduling, Strict Priority scheduling and Weighted Round Robinscheduling that are used to schedule the packets based on the priority of each typeof service. For real time traffic like VoIP, the role of the classification and schedulingstrategies is of paramount importance as they play a crucial role in guaranteeing theend to end quality of service to the users. During periods of congestion, real timeservices like VoIP can be severely impacted if there is a marginal increase in the endto 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 strategiesand to analyse their impacts on VoIP traffic during congestion periods. The analysisis done based on simulations using OPNET simulation tool.



1-4 Related Work 5

1-4 Related Work

The transport of voice over LTE has a lot of challenges with respect to QoS as mentionedearlier. In the literature, there are a number of studies which are focussed on theoptimum scheduling of resources for supporting VoIP service. In [3], Siomina et.al. haveanalysed the impact of prioritizing VoIP over other services in the radio network. Theperformance of prioritized VoIP is compared with Best Effort VoIP and the advantagein terms of increase in capacity is explained. In [4], Zaki, et.al. have studied the impactof dynamic packet scheduling on the performance of VoIP in LTE. Puttonen, J [5] andYasir Zaki [6], have studied the impacts of MAC scheduling algorithm for differenttypes of services. Most of these studies are focussed on the scheduling of resources inthe LTE radio network. To the best of my knowledge there are very few studies thathave 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 studiedthe impact of dimensioning in the transport network. In this study, analytical modelshave been proposed for dimensioning the transport network for real time and non realtime 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 ofLTE and IMS networks is introduced followed by a brief explanation on the QOSconcepts in LTE. The chapter also gives an overview on the Diffserv architecturethat will be used in the transport network for classification of various serviceslike voice, FTP etc. The chapter concludes with the explanation on the variousscheduling strategies that will be used in the transport network.

• In chapter 3, the details of the OPNET modeller are presented. The limitationsof the LTE model in OPNET and the changes that were done on the variousprocess models are explained. The configuration details of the various nodes inthe LTE network and the final OPNET simulation environment that will be usedfor performing the analysis is presented at the end of this chapter.

• In chapter 4, the results of the simulation are presented. The chapter begins witha brief introduction of the various metrics that were used to perform the analysisfollowed 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 followedby various scenarios that are being considered for integration of VoLTE withthe existing 2G/3G networks. The idea behind this study is to get an industryperspective on the technical impacts of VoLTE solution.


6 Introduction

• In chapter 6, the main results of the thesis are summarized and topics for furtherresearch have been proposed.



Chapter 2

Background

2-1 Introduction

This chapter begins with an overview on the LTE network architecture which explainsthe functions of the various elements present in the LTE network. The QoS conceptin LTE is presented in section 2.3. The QoS concept in LTE is based on bearers thatuniquely define the type of treatment for the packet flows between the mobile and thegateway nodes in the network. Hence this section provides the necessary informationrequired for a better understanding of the QoS concept in LTE. The voice over LTEsolution also requires an IMS core network that performs the necessary signalling andmedia related functions for providing voice services. The IMS core network architectureis presented in Section 2.4 to provide an overview on the functions of the key elementsin IMS domain. In Section 2.5, the Differentiated Services architecture is explainedwhich 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 areexplained.

2-2 LTE Network Architecture

The LTE network architecture is shown in the Figure 2.1. The network architecturecalled the Evolved Packet System (EPS) has a flat IP based architecture and is dividedinto the Evolved Universal Terrestrial Radio Access Network E-UTRAN and EvolvedPacket Core (EPC). The overall architecture consists of five elements which are ex-plained as follows.

E-UTRANThe radio network called the E-UTRAN comprises of the E-Node B’s that are intercon-nected to each other over the X2 interface and connected to the core network elements


8 Background

Figure 2-1: LTE Network Architecture

over the S1 interface. The E-Node B’s are responsible in scheduling and allocation ofthe radio resources for the users in the LTE network. The E-Node B terminates thecontrol plane signalling messages as well as the user plane data with the EPC over theS1 interface.

EPCThe EPC is the core network comprising of four elements which are Mobility Manage-ment Entity (MME), Serving gateway, Packet Data Network (PDN) gateway, Proxy andCharging Rules Function (PCRF) and Home Subscriber Server (HSS).

MMEMME is the most important element in the EPC as it terminates the control planesignalling from the user. Some of the functions performed by MME include authenti-cation, mobility management, security and retrieval of subscription information fromthe HSS.

Serving gatewayServing gateway is responsible for forwarding the user plane packets from the mobiletowards the PDN Gateway. It is also responsible for tunnelling the user plane IP pack-ets using the GPRS tunnelling protocol (GTP) when the user moves across different ENode B’s and serves as a mobility anchor for the user plane packets in the LTE network.

PDN GatewayPacket data network gateway is the end node in the LTE network. It acts as an edgerouter and routes the user plane IP packets from the mobile nodes to other networkslike Internet, IMS etc. It is also responsible for allocation of IP address to the user.

PCRFPCRF is responsible for enforcing various operator policies on the network like guar-



2-3 QoS Architecture in LTE 9

anteed QoS, maximum bit rate provisioned for a user etc. It communicates with thePDN-gateway in enforcing these policies for various users in the LTE network.

HSSHSS is the master database containing all the subscription information of the useralong with the subscription for various services that are offered by the operator. It alsocomprises of the authentication centre which stores all the keys required for ensuringthe 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 packetflow between the user and the PDN-GW and is responsible for the priority that is givento a packet flow across the LTE network. Bearers are established after the successfulauthentication and registration of the user in the LTE network. The LTE bearerarchitecture 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 dif-ferentiate the types of packets that flow through it. The TFT does this classificationbased on one of the following parameters:

• Port numbers

• ToS/DSCP Values

• Source/Destination address

• Protocol (TCP/UDP)


10 Background

QCI Resource Type Priority Packet Delay Budget PacketError LossRate

Services

1 GBR 2 100 10−2 Voice.2 GBR 4 150 10−3 Voice Conversa-

tion (Real TimeStreaming).

3 GBR 3 50 10−3 Real Time Gam-ing.

4 GBR 5 300 10−6 Non Conver-sational Video(buffered video).

5 Non-GBR 1 100 10−6 IMS Signalling.6 Non-GBR 6 300 10−6 Video (Buffered

Streaming).7 Non-GBR 7 100 10−3 Interactive Gam-

ing.8 Non-GBR 8 300 10−6 Video (Buffered

Streaming).9 Non-GBR 9 300 10−6 Video (Buffered

Streaming).

Table 2-1: EPS QOS Bearer Definitions [9]

The bearers are classified as two types namely the default and dedicated bearers. De-fault bearers are established during the allocation of IP address to the user by the PDNGateway. Default bearers provides the basic IP connectivity to the LTE network anddoes not provide any guaranteed QoS for the packets that are transmitted across thisbearer. Dedicated bearers are used for specific services like voice, video streaming etcand are established based on the subscription profile of the user.

The bearers are also classified as Guaranteed Bit Rate (GBR) and Non GuaranteedBit Rate (N-GBR). As the name indicates the GBR bearers provide guaranteed QoS tothe packets that flows through this bearer and is less likely to be affected during heavycongestion at the network. On the other hand, the N-GBR bearers are used for servicesthat do not have strict QoS constraints. As shown in the Figure 2.2, each bearer inLTE is characterized by a QoS Class Identifier (QCI), Allocation and Retention Priority(ARP), packet delay budget and maximum bit rates. The QCI uniquely identifies thetype of bearer that is provisioned for the user at the radio and core networks. It is usedin determining the type of treatment a packet flow experiences at each of the nodes inthe LTE network. The ARP is used to decide whether a bearer can be admitted and isalso 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.



2-4 IMS Network Architecture 11

2-4 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 etcover an IP backbone independent of the underlying access network through which theuser registers with it . The most important service provided by an IMS network is theMultimedia Telephony Service (MMTel) which is the basic voice over IP service butwith guaranteed QoS. IMS is also capable of interworking the circuit switched 2G/3Gnetwork with packet switched networks like LTE. Hence IMS based voice is envisioned asthe ultimate target solution for supporting voice in advanced next generation networkslike LTE.

The Figure 2.3 presents the key elements of an IMS network. The main elementsof IMS core network are the Proxy Call Session Control Function(P-CSCF), ServingCall Session Control Function (S-CSCF), Interrogating Call Session Control Function(I-CSCF), Breakout Gateway Control Function (BGCF), Media Gateway Control Func-tion (MGCF) and Media Resource Function (MRF). The main functions of these entitiesare 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 firstpoint of contact for the user within the IMS domain. All the requests of the userto the elements in the IMS domain as well as to the application servers are routedthrough the P-CSCF. In addition, the P-CSCF performs functions like subscriber


12 Background

authentication and establishment of security association with the mobile. It mayalso authorize QoS resources for the voice bearer by way of a policy decisionfunction.

• S-CSCF : S-CSCF is the main element in the IMS domain which performs im-portant functions like subscriber registration, authorization for using specific ap-plication servers, DNS lookup to retrieve the address of the destination etc. Itdownloads the user profiles from the HSS for performing authorization of thesubscriber.

• I-CSCF : I-CSCF is a SIP server that acts as a last point of contact in the IMSdomain i.e. it is at the edge of the IMS domain and all requests from other IMSdomains as well as requests from remote application servers are routed throughthe I-CSCF. During initial registration, the I-CSCF queries the HSS to assign aS-CSCF for the specific user.

• BGCF : BGCF performs breakout to other domains when routing of the requestbased on ENUM lookup is failed at the S-CSCF. It is mainly used when thedestination user is a PSTN user and the call needs to be transferred to the CSdomain.

• MGCF : MGCF is used to translate SIP signalling into ISUP signalling for com-munication towards PSTN and other CS networks. It also controls the mediagateway which translates the RTP into CS media stream.

• MRF : MRF is used in transcoding between different codecs and provides mediarelated functions like mixing of media streams and playing tones etc. The MRF issubdivided into Media resource function controller (MRFC) and Media resourcefunction processor (MRFP) which perform the media translation activities in thecontrol and user plane respectively.

2-5 Differentiated Services Architecture

The Differentiated services (Diffserv) architecture defined in [11] is used by the E-NodeB and the PDN gateway to map the QCI to a DSCP value in uplink and downlinkrespectively. This mapping at the E-Node B and the PDN gateway allows for the clas-sification of the packets in the underlying transport network. The Diffserv architectureconsists of various Per Hop Behaviours (PHB) that are used to identify and classifythe packets and apply appropriate QoS treatment at the transport network. The PHBclasses are broadly classified into three classes namely Expedited Forwarding(EF), As-sured Forwarding(AF) and Best Effort(BE).

The EF class has the highest priority and is generally used for delay critical serviceslike signalling, voice etc. The AF class consists of several sub classes with differentlevels of drop precedences as shown in the Table 2.2. The drop precedence enables the



2-6 Scheduling Strategies 13

Drop Precedence AF 4X Af 3X AF 2X AF 1XLevel 1 AF41

(DSCP 34)AF31(DSCP 32)

AF21(DSCP 26)

AF11(DSCP 20)

Level 2 AF42(DSCP 36)

AF32(DSCP 30)

AF22(DSCP 24)

AF12(DSCP 18)

Level 3 AF43(DSCP 38)

AF33(DSCP 28)

AF23(DSCP 22)

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 Effortclass is the default PHB and has the least priority among the three classes. The AFclass and BE class employ the Weighted Random Early Detection technique to detectcongestion of queues based on pre defined thresholds. When the number of packetsin the queue exceeds a minimum threshold, the WRED technique starts dropping ofpackets based on the weight assigned to each queue. If the link is heavily congested andthe number of packets in the queue exceeds the maximum threshold then all incomingpackets to the queue are dropped.

2-6 Scheduling Strategies

The transport network consists of a scheduler which assigns the available network band-width based on certain priorities and weights. The scheduler uses the classification ofthe packets based on DSCP to form these strategies for allocation of resources in thetransport network. In this work, three different scheduling strategies are evaluated andtheir performance is compared for delivering high quality voice service in LTE network.The following section gives an overview on the different scheduling strategies that areimplemented in the transport network.

2-6-1 Strict Priority Scheduling

In this type of scheduling, the packets are grouped into four levels of priority namelylow, normal, medium and high. Packets which are very sensitive to delay like voice aregiven a high priority and services like streaming which have tolerable delay budgetsare given medium priority. TCP based services like HTTP and FTP are mapped tonormal and low priorities respectively as they have less constraint on the delay budgets.The scheduler always processes the high priority packets before servicing packets inother queues. This scheduling is especially useful for services like VoIP which havestringent delay requirements. The major drawback of this scheduling is, when thenetwork is congested with high priority traffic like voice, the low priority data trafficwill completely devoid of resources and hence the overall throughput of the network isreduced.


14 Background

2-6-2 Weighted Round Robin (WRR) Scheduling

This type of scheduling is based on the classical round robin scheduling where differenttypes of services are served in a round robin manner. The only addition in WRR isthe presence of weight which determines the number of packets that are removed fromthe queue. The packets are grouped into various queues and each queue is assigned aweight. Based on the weight, the scheduler calculates the bandwidth for each queueand corresponding to this bandwidth, number of packets in the queue are removed at atime before moving to the next queue. Hence WRR does packet by packet schedulingin a round robin manner.

2-6-3 Weighted Fair Scheduling

In Weighted Fair scheduling defined in [12], the packets are grouped into various queuesand each queue is assigned a weight which determines the fraction of the total band-width available to the queue. In our case, there are different PHB’s such as EF, AFand BE are assigned weights based on the priority of the traffic. The bandwidth foreach queue is based on the weights and is expresses as

BWk = Wk

W∗BW (2-1)

The Weighted Fair scheduling assigns the bandwidth for each service based on theweight assigned to each queue and not based on the number of packets. Hence whenvarious types of traffic like VoIP, FTP, HTTP are flowing in the network, the bandwidthfor each service is proportional to its weight and independent of the size of the packetin the queue. The main difference between Weighted Round Robin and Weighted Fairis that the former does packet by packet scheduling in each turn whereas the latterdoes bit by bit scheduling. Weighted Fair hence has an advantage in the fact that it isaware of the true size of the packets in each queue while performing scheduling whereasWeighted Round Robin is not aware of the same.



Chapter 3

Simulation Model

3-1 Introduction

This chapter begins with an explanation on the details of the OPNET simulation envi-ronment used for modelling the LTE network. In section 3.3, the issues with the LTEmodel in OPNET are presented briefly and the modifications that were performed toresolve these issues are highlighted. The configuration parameters of the LTE networkand the simulation settings for the VoIP and FTP process models are provided in thesubsequent sections.

3-2 Introduction to OPNET Modeller

The OPNET simulation environment [13] is a discrete event simulation tool that isused in analysing the performance of various networks like LTE, WiMAX, Wi-Fi andZigbee. The models library in OPNET consists of a large number of models supportingvariety of protocols like TCP, UDP, SIP and is capable of simulating applications likevoice, video, FTP etc. In this thesis, the LTE model in OPNET is used along withapplication models like voice and FTP. The details of the simulation environment arepresented 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 performthe simulation. The LTE network model in OPNET is shown in Figure 3.1. The modelconsists of mobile nodes , an E-Node B and an EPC. The LTE core network consistingof the MME, serving gateway and PDN-gateway is modelled by a single device rep-resented as the EPC in the Figure 3.1. The LTE attribute definition node is used to


16 Simulation Model

Figure 3-1: LTE Network Model

define various configuration parameters like DL and UL frequencies, bandwidth andthe various bearers that will be configured on the mobile nodes. The LTE model im-plements most of the features that are standardized by the 3GPP. However, it has somelimitations in the establishment of bearers and hence significant changes are required inthe 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 bearerin the transport network. This S1 bearer uses the GPRS Tunnelling Protocol (GTP).



3-2 Introduction to OPNET Modeller 17

Hence for each bearer that is established between an user and EPC, there is a separateGTP tunnel established for control plane signalling as well as user plane data. Thesignalling GTP tunnel is used for transmitting all the signalling information related tothe establishment of the bearer. The data GTP tunnel is used in forwarding all theuser plane IP packets from the user to EPC and vice versa.

The Figure 3.3 gives the complete protocol architecture across various nodes in theLTE network. The GTP-U layer in the E-Node B represents the tunnels that are cre-ated between the E-Node B and EPC. In the uplink when the E-Node B receives IPpackets from the mobile, the GTP layer encapsulates the received IP packet and copiesthe contents of the inner IP header to the outer IP header. The same process is repeatedin the downlink direction when the EPC node receives an IP packet from outside theLTE network. The encapsulated GTP packet structure is shown in the Figure 3.4.

Figure 3-3: Protocol Architecture

Figure 3-4: GTP Encapsulated Packet


18 Simulation Model

3-2-2 Issues in LTE model

In the LTE model present in OPNET, the process of GTP encapsulation is not imple-mented in accordance to the QoS type of the bearer . The packets entering the E-NodeB in the uplink are encapsulated into an IP packet without any classification based onthe type of bearer (DSCP mapped to BE by default). The same issue is there in thedownlink when the packets entering EPC are encapsulated without proper classifica-tion. 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 levelin the transport network.

As explained in the problem definition in Chapter 1, the intermediate nodes in thetransport network between E-Node B and EPC are not aware of the classification basedon bearers. The type of scheduling strategies used in the transport network also has nomeaning, if all the packets are classified with same priority. So, it is very important forthe E-Node B and EPC to perform packet level classification by mapping the bearertype to DSCP. Hence changes are required in the process models in the E-Node B andEPC. The following sections illustrate the changes that were performed in the E-NodeB 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 thebearers in the transport network which needs to be implemented. This section presentsan overview on the changes that were done in the E-Node B and EPC nodes in theOPNET.

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 anoverview on the various layers of the 3GPP LTE stack that are implemented in E-NodeB. In this node model, there are two processes lte_ s1 and gtp (highlighted in Figure3.5) that are to be modified. The lte _ s1 process model at the E-Node B is shown inthe Figure 3.6. This process is run every time when a dedicated bearer is created inthe LTE network. The s1 _ msg _ rcvd represents the state in which the E-Node Bhas received a new bearer request message from the mobile. The state runs a dedicatedbearer setup function and commands the GTP layer in the E-Node B to create a tunnelfor this dedicated bearer towards the EPC in the uplink direction. So, the change thatneeds to be performed in this process is to map the QoS type of the tunnel created tothe 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 whichuniquely identifies the bearer in the network.



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 usedto perform the mapping between the bearer level QoS and DSCP value in the IPheader. 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 thatduring the creation of GTP tunnel between the E-Node B and EPC, the DSCPvalue corresponding to the bearer ID is also taken into consideration.


20 Simulation Model

3-3-2 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 nodemodel, the EPC also has the lte _ s1 _ nas and gtp process models that are responsiblefor bearer creation and tunnel creation respectively. Hence these two processes are to bechanged 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 bearerthat carries the data between the E-Node B and EPC for the mobile in the downlinkdirection. In the Figure 3.8, the state s1 _ msg _ rcvd represents the state in theEPC 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 actuallycontains 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 followedhere 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 inFigure 3.9. The GTP-U block performs the encapsulation of the user plane IP packetreceived at the E-Node B and delivers it to the UDP layer for transport towards theEPC. The GTP-U block consists of four states which are idle, tunnel search, gtpencapand to UDP. When a packet arrives at the E-Node B, the process goes from the idle



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 tothe UDP module. Else, the process goes to the tunnel management state, where thetunnel 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 arrivesfrom outside the LTE network.

Hence the gtpencap state is where the actual process of mapping between the bearerID and the DSCP takes place and the contents of the inner IP header are copied to theouter IP header. The various functions that were used to perform this mapping in theGTP layer were modified to overcome the limitations that were mentioned in Section3.1.2.


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 inthe analysis. The network topology used for performing this simulation is shown in theFigure 3.10. In the network topology, there are two cells represented by E-Node B 1and E-Node B 2 connected to the EPC via an Edge router. Each cell consists of 30LTE users. The EPC node is connected to an IMS network via an edge router. TheEthernet links between the E Node B and the EPC are 5 Mbps. All other links in thecore network are of 10 Mbps capacity. There are two FTP servers connected to theEPC node which are used by the mobile nodes for establishing FTP sessions in thenetwork. 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 mobilenode is configured to run one type of application at a time. The Figure 3.11 shows theimportant configuration details of the mobile nodes in the network. The EPS bearerconfiguration attribute defines four bearers namely Platinum, Gold, Silver and Bronze.



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 DSCPvalue. 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 DSCPPlatinum EFGold AF 11Silver AF 43Bronze BE

Table 3-1: EPS Bearer to DSCP Mapping

services like voice, FTP, etc. The mobility feature in the mobile nodes is set to disabledas 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 capacityof each cell is limited to 10 Mbps. The channel between the mobile nodes and E-NodeB is configured to be an error free channel as the primary objective of this analysis is toinvestigate the impact of congestion in the core network. Hence various physical layereffects like multipath and interference effects are not modelled in these simulations.


24 Simulation Model

Figure 3-11: Mobile Configuration

The MAC scheduler implemented in the OPNET E-Node B module uses a priorityscheduling among the guaranteed and non guaranteed bit rate bearers which impliesthe guaranteed bit rate bearers are always allocated radio resources ahead of the nonguaranteed bit rate bearers. To avoid the scenario of packets getting dropped due tonon availability of resources in the radio network, the peak usage of each cell is limitedto 50 percent of the total capacity. The summary of the configuration parameters ofthe E-Node B is listed in the Table 3.2.

Parameter Name ValueBandwidth 3MHz

UL Frequency 1920 MHzDL Frequency 2110 MHz

Channel Characteristics Error FreeNo. of Transmit/Receive Antennas 2

Table 3-2: E-Node B Configuration Parameters



3-4 Simulation Environment 25

3-4-3 IMS Model

The IMS model used in this simulation environment is used from the contributed modelssection 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 VoIPcalls between the different users in the network. The IMS signalling flow in the LTEnetwork requires the highest priority as it is the first procedure that is invoked towardsthe establishment of the VoIP call between the users. Hence all the IMS signallingpackets 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 net-work.The domain name and area configured in these servers are also configured in themobiles and using these attributes, each mobile registers with the IMS network. Thethree call session control functions are used to route the signalling between two VoIPusers before the establishment of the media path. The SIP signalling procedure definedin [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 themain focus of our study is on the user plane voice bearer and not on the control planesignalling data.


26 Simulation Model

3-4-4 Application Configuration

We use two traffic models namely voice and FTP for performing this analysis. Thedetails of the traffic models are explained below.

VoIP model

We use the commonly used G.711 voice codec for all the simulations. The codec hasa 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 addheaders to each packet and hence the overall bandwidth is around 90 kbps. In oursimulations, silence suppression is used and is modelled as an exponential distributionwith 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 ValueCodec G.711 (64 Kbps)

Frame Size 20 msVoice Activity Factor 0.6Silence Suppression 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 TCPconnection established for each request between the mobile and the server.



Chapter 4

Results

4-1 Introduction

In this chapter, the performance of the various scheduling scheduling strategies thatwere explained in Chapter 2 are investigated. The chapter begins with an introductionon the various QoS parameters that were used for performing the analysis. In all thesimulations, only voice and FTP services are used. The details of the traffic models forvoice 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 callin 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 toend delay is shown for the all the users in the network.

• Packet Delay Variation (PDV) : This parameter gives the variance in the end toend delay among all the packets received at the user. The mean of this PDV isshown 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 theE-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 packetsthat are lost in the network due to congestion. The packet loss rate is measure atthe EPC node, since the congestion is in the core network. The mean of the PLRfor all the users in the network is shown for all the simulations.


28 Results

In addition to the above QoS parameters, the Mean Opinion Score (MOS) is also pre-sented for all the simulations. The MOS is a measure of the Quality of Experience forthe VoIP users in the network. The E-Model defined in [14] is used to calculate theMOS based on the R-factor. The R-factor called the rating factor is used to measurethe 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 impair-ment caused due to packet losses in the network. The R-factor is mapped to a MOSscore using the following mapping defined in [14]:

MOS = 1 + 0.035R + 7 ∗ 10−6 ∗R(R− 60)(100−R), 0 ≤ R ≤ 100 (4-2)

MOS = 1, R ≤ 0 (4-3)MOS = 4.5, R > 100 (4-4)

The OPNET software uses the above model to calculate R factor and is mapped to theMOS using the above equation. The MOS score is mapped to the level of satisfactionof the users based on Table 4.1 [14].

MOS score Quality of VoIP call experienced by the user4.3 - 5 Very much satisfied4 - 4.3 Satisfied3.6 - 4 Many users satisfied3.1 - 3.6 Many users dissatisfied2.6 - 3.1 Nearly all users dissatisfied

Less than 2.6 Not recommended

Table 4-1: MOS satisfaction level

In all the simulations, the average of the MOS for all VoIP users in the network ispresented.

4-3 Scenario 1

This scenario is used to illustrate the significance of QoS in the transport network bymapping 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 numberof voice users in the network is periodically increased from 20 to 100. There aretotally 25 LTE mobiles running VoIP application in each cell and each user is



4-3 Scenario 1 29

configured to establish multiple VoIP sessions simultaneously. The VoIP sessionis carried over a Guaranteed Bit Rate (GBR) bearer (QCI 1 in Table 2.1) and ismapped 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 totally10 FTP users (5 in each cell). VoIP users are mapped to GBR bearer as in case1 and FTP users are mapped to Non GBR bearer (QCI 9 in Table 2.1) in thedownlink. Both the voice and FTP are mapped to the same best effort QoS inthe transport network. So, the packets entering the nodes EPC, Edge Router 1and 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 networkis 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 varia-tion for VoIP users in the network. For case 1, the delay is constant at 80 ms whereaswhen there is an ongoing FTP session in case 2, there is a significant increase in theend 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 stillwithin the total bandwidth even when the number of VoIP users is large. At 100 VoIP


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 ar-rived by assuming that all the users are sending voice packets at the same time and thebandwidth required for a single voice call is 90 Kbps after adding the RTP /UDP/ IPheaders to the actual voice payload. The peak traffic will never be reached as each userhas an exponential distribution on the talk spurts and silence periods. So, the mean endto end delay for all VoIP users in the network remains constant. This also explains theFigure 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 pack-ets are getting queued in the core network and the edge router till the larger FTPpackets are processed in each node. This causes a larger variation among the packetsreceived 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 of100 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.



4-4 Scenario 2 31

4-4 Scenario 2

In Scenario 2, the voice users are accorded the highest priority and the FTP users aremapped to the lowest priority. Each VoIP user is assigned to a Platinum bearer (QCI1 in Table 2.1) and mapped to the highest EF QoS class in the transport network. TheFTP users are assigned to a Bronze bearer (QCI 9 in Table 2.1) and mapped to theBE QoS class in the transport network.


This scenario evaluates the performance of the scheduling algorithms explained in Chap-ter 2. The scenario is subdivided into three cases as follows:

• Case 1: In this case, the Weighted Fair scheduling algorithm explained in Chapter2 is analysed. The high priority VoIP traffic is assigned a weight of 7 and the FTPtraffic is assigned a weight of 3. Hence, the total bandwidth assigned for voiceusers 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 Chapter2 is analysed. In terms of priority as explained earlier the VoIP users are mappedto high priority and FTP users are mapped to low priority.

• Case 3: In this case, the Weighted Round Robin scheduling algorithm explainedin Chapter 2 is analysed. The weights for the voice and FTP services are same asthose assigned in Case 1.


This section presents an analysis on the various parameters that were explained inSection 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 andcase 3, the Weighted Fair and Weighted Round Robin scheduling algorithms a definitebandwidth 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 thenumber of VoIP users is 80, the mean end to end delay is around 100 ms for both thecases which is still within the acceptable limit. The increase in the end to end delay isattributed 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 theweight which is calculated to be around 7 Mbps. The delay falls within the 100 msthreshold as the peak bandwidth for voice will not be reached due to the exponentialdistribution of the talk spurts between the users.


32 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 voicepackets to reach the E-Node B from the EPC.



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 increasewhich 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 thebandwidth demand of the VoIP users, leading to more waiting time in the queuesat the core network as shown in Figure 4.4. The mean end to end delay at thispoint 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 delaywith the variation increasing to 0.3 ms. The PDV is small compared to the No QoScase explained in Scenario 1. This is because of of the separate bandwidth provisioningfor VoIP and FTP users in these two scheduling algorithms.The mean end to end delay and delay variation for Strict Priority scheduling is alsoshown in Figures 4.3 and 4.5. The delay remains constant at 80 ms and packet delayvariation is negligible. As explained in the section 2.6, the Strict Priority schedulingalways performs better for VoIP users which are assigned a higher priority comparedto the FTP users.

PLR and MOS

In Figures 4.6 and 4.7, the PLR for the VoIP users in the network and the correspondingMOS is shown. From the Figure 4.6, it is seen that for case 1 and case 3, the packet lossrate is almost negligible till number of VoIP users is equal to 80 when the packet lossrate reached the threshold of 2 percent. Beyond this, with an increase in the number of


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 pointis of no significance, as more than 2 percent drop in the number of packets implies thatthe VoIP calls are dropped. The high packet loss rate beyond this point attributes tothe 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 allthe users are dissatisfied and hence beyond 80 users, there is no possibility of having



4-4 Scenario 2 35

more number of VoIP users for case 1 and case 3. In case 2, the PLR is negligible asshown 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 transfertime for the FTP users in the network. The Figure 4.8 shows the FTP transfer timefor 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 thecongestion in the link between the EPC and the Edge router, thereby leading to morequeuing of packets in the EPC. The transfer time for case 2 shows a large increasewhen the number of VoIP users in the network increases beyond 80. This leads to anundesirable situation where there are more TCP re transmissions in the network leadingto increased congestion. The Figure 4.9, shows the total FTP traffic received by all theusers in the network. It is seen from the Figure 4.9, that there is significant drop inthe FTP throughput when the number of VoIP users increase in the network for all thethree cases. For case 1 this drop in throughput is around 30 percent whereas for case 3the drop in throughput is around 50 percent. Weighted Fair scheduling performs betterthan Weighted Round Robin due to the fact that it performs bit by bit scheduling andhence 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 toWeighted Round Robin which explains the behaviour in Figure 4.9. For case 3 there isalmost a 75 percent drop in throughput when the number of VoIP users in the network


36 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 degradeddue to the increase in the number of voice users for case 2 when compared to case 1and 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 analysedfor three scheduling algorithms. Within the acceptable QoS thresholds (100 ms endto end delay and 2 percent packet loss), the number of VoIP users for case 1 and case3 is 80 Users. The case 2 has a higher capacity of 100 users within the acceptablelimits but it comes at a cost as Strict Priority scheduling totally starves the resourcesfor 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 3which is not acceptable in environments with mixed traffic.



4-5 Scenario 3 37

4-5 Scenario 3

In Scenario 3, each VoIP user is assigned to a Gold bearer (QCI 7) in both directionsand mapped to the AF 11 QoS class in the transport network. The FTP users areassigned to the Bronze bearer as in Scenario 2 and mapped to the BE QoS class in thetransport network.


This scenario evaluates the performance of VoIP when assigned to a normal priorityin the radio and core networks. The scenario is subdivided into three cases as earlierdescribed 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 weightof 3 in the transport network. Hence, the total bandwidth assigned for voice usersis 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 areassigned low priority.

• Case 3: The weights for the Weighted Round Robin algorithm are same as thosein case 1.


Packet end to end delay and PDV

In Figure 4.10, the mean end to end delay for the VoIP users are shown. The meanend to end delay from the Figure 4.10 for case 1 and case 3 shows a significant increasewhen the number of VoIP users is beyond 60. At 70 users, the delay crosses the thresh-old of 100 ms. This behaviour is due to the fact that when the number of VoIP userscrosses 70, the maximum peak bandwidth for voice reaches around 6.3 Mbps which isslightly more than the provisioned bandwidth for voice which is 6 Mbps. The increasein the end to end delay is marginal until the number of voice users reaches 80 whenthe peak bandwidth is around 7.2 Mbps. At this point, more number of VoIP packetsare buffered in the interface of EPC node thereby increasing the end to end delay. TheFigure 4.11 exactly proves this point as the mean one way delay between the EPC andE- 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 mainlydue to the fact that merely changing the priority to normal does not affect the VoIPquality as still VoIP packets are served first by the scheduler before the FTP users areserved. The PDV is shown in Figure 4.12 for all the three cases and as explained forscenario 1, the packet delay variation has no significant impact when the number ofVoIP users are increased.


38 Results

Figure 4-10: End to End Delay vs No. of VoIP users

Figure 4-11: S1 delay vs No. of VoIP users



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 exceedsthe 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 impliesthat beyond this point the calls will get dropped. In case 2, there is no packet loss asit follows the same behaviour as explained in scenario 2. The Figure 4.14 shows theaverage MOS for all the users in the network. It is seen that the quality of experiencedegrades for case 1 more than for case 3 when the number of users is increased beyond60.


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 Impact on FTP traffic

In this scenario, the FTP traffic is not impacted much when the number of VoIP usersis increased. This can be observed in the Figure 4.15 which shows the mean file transfertime of all the FTP users. When the number of VoIP users are increased beyond 80, the



4-5 Scenario 3 41

mean FTP transfer time for case 1 almost remains constant and for case 3 it increasesmarginally by 3 percent. In comparison with scenario 2, we observe that the mean FTPtransfer time is almost reduced by 50 percent when the number of VoIP users is 100for scenario 3. This is largely due to the fact that a higher percentage of bandwidth isassigned 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 thisscenario, there is no change in the FTP throughput when the number of VoIP usersis 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 Prior-ity scheduling, there is a significant drop as in scenario 2 and hence there is a severedegradation in FTP throughput when compared to other two scheduling strategies.


42 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 forthe three different scheduling algorithms. Within the acceptable QoS thresholds (100ms end to end delay and 2 percent packet loss), the number of VoIP users for case 1and case 3 is 65 Users whereas for case 2 it remains the same as in previous scenario at100 users. The main reason behind the drop in the capacity of VoIP users is due to thefact that the VoIP users are mapped to AF bearer in the transport network which hasa lower bandwidth limit compared to the previous scenario where the EF class had anhigher bandwidth limit. The FTP throughput achieved in this scenario is much bettercompared to the previous scenario when the network is congested with VoIP users forcase 1 and case 3.



4-6 Scenario 4 43

4-6 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 bearerand 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 andnormal priority VoIP users are mapped to AF 11 class. The FTP users are mappedinto Bronze bearer (QCI 9). The scheduling strategy is used such that for high priorityVoIP users, Strict Priority scheduling is used. The remaining available bandwidth isshared between the normal priority VoIP users and FTP users using the Weighted Fairscheduling algorithm. The scenario is divided into four cases according to the numberof high priority and normal priority VoIP users. In all the cases, the weights for normalpriority VoIP users and FTP users are set to 6 and 3 respectively. The Table 4.1 shownbelow gives the details of each of the four cases.

Application Type Case 1 Case 2 Case 3 Case 4VoIP (High Priority) 30 20 30 20

VoIP (Normal Priority) 40 60 60 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 andnormal VoIP user. In Figure 4.17, we see that there is no significant change in the endto 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 thetime of congestion. There is also no significant PDV due to the same reason and hencePDV 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. Theend to end delay for case 1 is around 80 ms. The peak bandwidth for normal userswhen all of them send packet simultaneously is around 3.6 Mbps which is less than theprovisioned bandwidth of 4.8 Mbps. Hence there is very less waiting time in the corenetwork for the VoIP packets which explains the less packet end to end delay. Dueto the same reason, there is no variation in packet delay as seen in Figure 4.19. Incase 2 and case 3, the end to end delay is around 100 ms. This is a significant changecompared to the case 1 but still the value is within the acceptable limits of end to end


44 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 thanthe provisioned bandwidth. Hence the packers are buffered in the EPC node, whichleads to an increase in the end to end delay. For case 4, the delay is 140 ms which



4-6 Scenario 4 45

is beyond the acceptable value. The peak bandwidth in this case is around 7.2 Mbpswhich is well beyond the provisioned bandwidth. This leads to a congestion in the EPCnode leading to more waiting times. The values of PDV for cases 2 and 3 are around0.5 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 usingstrict 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 experi-ence for premium users. The PLR and MOS for premium VoIP users follow the samepattern as in Figures 4.6 and 4.7.

The Figure 4.21 shows the PLR for normal VoIP users. There is no packet loss forcase 1 as the congestion scenario is not yet reached and the bandwidth is within thelimits. For case 2 and case 3, there is a packet loss of around 1.5 percent which isstill within the acceptable value of 2 percent. For case 4, there is a large packet lossof around 10 percent which implies that the calls are dropped. The Figure 4.22 showsthe MOS for the normal VoIP users. It is clear from the Figure 4.22 that the MOS forcases 2 and 3 are less compared to case 1 but still within the acceptable value.


46 Results

Figure 4-20: Packet Loss Rate

Figure 4-21: Mean Opinion Score

Impact on FTP traffic

The Figure 4.23 shows the FTP transfer time for all the four cases. It is seen thatfor case 1 the transfer time is 24 seconds and for case 2 the transfer time is around27 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



4-6 Scenario 4 47

is around 26 seconds in Scenario 3 . Similarly the total number of VoIP users in case 2is 80 and at the same point the transfer time seen in Figure 4.15 is 30 seconds. Hencein these two cases for the same number of VoIP users, there is a marginal decrease inthe 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 thecases. This is due to the fact that in case 3 in Figure 4.23, there are more number ofVoIP users (premium + normal = 90) which are within the acceptable limits of QoScompared to 4.15 and hence the VoIP capacity is increased at the expense of the FTPthroughput. In case 4, the number of normal VoIP users are very high which leadsto more congestion in the core network and hence the throughput for both the VoIPand FTP users are significantly affected leading to a poor performance for both theservices.

Figure 4-22: FTP Transfer time


48 Results

4-7 Comparison of Scenarios

In this section, a comparison of the scenarios 2, 3 and 4 is done for a better under-standing of the results. The table 4.2 gives the number of satisfied VoIP users in eachscenario which are within the acceptable limits of 100 ms delay and 2 percent packetloss. 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 ofvoice users (premium+normal).

Scenario Name No. of VoIP users Corresponding FTP Transfer timecase 1 case 2 case 3 case 4 case 1 case 2 case 3 case 4

Scenario 2 80 100 80 X 38 130 40 XScenario 3 65 100 65 X 24 130 24 XScenario 4 70 80 90 20 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 ofnumber 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 in-crease in transfer time for FTP users when there is congestion in the network. Thoughthere no delay guarantees for FTP users in this scenario, there should not be totaldegradation of throughput for FTP as in case 2. Hence in Scenario 2, the maximumnumber of VoIP users that can be supported with acceptable QoS limits for voice is80. Comparing this with Scenario 4, there are 90 VoIP users that can be supportedwithin the acceptable QoS limits. Hence there is about 10 percent increase in the VoIPcapacity 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 transfertime 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 whileensuring that the increase in delay for data traffic is within the bounds is 65. Com-paring the Scenario 3 with Scenario 4, we see that for the same criteria i.e increase indelay 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 comparedto Scenario 3.

Hence by grouping of VoIP users into different levels of priority an increase in capacityis achieved when compared to mapping them to a single specific service class.



Chapter 5

Technical Details of Voice over LTEvia IMS based solution - An Operator

Perspective

5-1 Introduction

This chapter gives an overview on the various technical impacts of VoLTE solutionon the existing CS and PS networks. The VoLTE solution will introduce the voicefunctionality in the LTE network using the new IMS framework which is widely beingaccepted as the long term solution for supporting voice in LTE network. IMS basedvoice is widely seen as the better solution in the current scenario capable of deliveringvoice in the LTE network. Hence, operators worldwide or considering the deploymentof an IMS based solution. This chapter explains the technical details of VoLTE solutionfrom an industry perspective.

5-2 VoLTE Architecture

The Figure 5.1 shows the important elements in the VoLTE architecture. The archi-tecture 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 thebasic call origination/termination functionalities as well as value added services likePresence, Instant messaging etc. The user after obtaining an IP address from the LTEnetwork performs a registration operation with the IMS network which enables theusers to get access to the basic services like voice and also other value added servicesbased on subscription. The Table 5.1 gives the relevant protocols and interfaces forVoLTE solution [15].


50 Technical Details of Voice over LTE via IMS based solution - An Operator Perspective

Figure 5-1: VoLTE Architecture

Nodes Interfaces ProtocolsMME ↔ HSS S6a Diameter

PCRF ↔ P-CSCF Rx DiameterI/S-CSCF ↔ HSS Cx DiameterI/S-CSCF ↔ AS ISC SIP

P-CSCF ↔ I/S-CSCF Mw SIP

Table 5-1: VoLTE Relevant Interfaces and Protocols

In the above architecture, LTE is deployed as a standalone network and there is nointegration with the 2G/3G networks. During the initial roll out of LTE, the coveragewill be minimum and hence there should be some way of integrating the LTE networkwith the 2G/3G network. This integration is quite challenging owing to the fact thatLTE has a completely packet switched architecture. As stated earlier, the voice in LTEis carried as VOIP packets. When the user is roaming outside of LTE coverage i.e. in2G/3G domains the voice call needs to be switched from VoIP based to legacy TDM



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/3Gnetworks 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 isrequired to integrate the LTE network with the existing 2G/3G network. When loss ofLTE coverage is detected, the user should be able to attach to 2G/3G network. If thereis an ongoing voice call in the LTE network, then a handover needs to be performed tothe 2G/3G network without interruption of voice call. There are two architectures forintegration of LTE with 2G/3G networks which are seen as a possible approaches forachieving 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 Ser-vices(ICS).

5-3-1 Independent PS based solution

In PS based solution, voice over IMS is implemented in both the LTE and 3G net-works. During loss of LTE coverage, a PS handover is performed towards the 3Gnetwork thereby providing seamless mobility between LTE and 3G networks. Thusboth the voice and data sessions that are active in the LTE network are simultaneouslytransferred to the 3G network, there by preventing loss of voice/data during the loss ofLTE coverage. The Figure 5.2 shows the architecture of PS based solution. The han-dover procedure is defined in [16]. The overview of the procedure is briefly summarizedbelow:

• The user is initially attached to the LTE network and a voice call is establishedvia IMS in the LTE network.

• When loss of LTE coverage is detected, the E-Node B in the LTE network initiatesan handover towards the MME which then forwards the same to the SGSN. TheMME also separates the voice bearers from the non voice bearers and performs amapping between the LTE bearer and 3G PDP context.

• The target SGSN reserves the necessary resources in the 3G network and alsocreates a session request towards the serving gateway. The SGSN reverts back tothe MME on successful completion of the reservation procedure.

• The MME in the LTE network then performs the handover execution procedureby sending handover command towards the E-Node B

• The E-Node B then sends a handover command to the UE containing the radioaccess network parameters of the target 3G network.



Figure 5-2: LTE-3G Integrated architecture

• The UE can continue the voice session after successful completion of the handoverprocedure.

Advantages and Limitations

The major advantage is, it is seen as the simplest solution for integrating LTE with3G network as it involves minimum changes in terms of network architecture. Theexisting PS network for 3G can be reused easily without any major upgrades. 3Gnetwork has PS based capabilities and hence handover of voice from LTE to 3G can beaccomplished easily via IMS without significant interruption.

The independent PS based solution cannot be taken as a target solution as it requirescomplete coverage of 3G network. Since both voice and data are carried in the 3G PSnetwork, higher bandwidth is required. Hence the advanced release of UMTS which isHSPA+ is needed to support high data rates for carrying both data and voice in thenetwork simultaneously. The existing CS network is not reused in this scenario whichcan be a major factor in the future when the legacy networks like 2G become obsolete.



5-3 Options for integrating LTE with existing CS/PS networks 53

5-3-2 Enhanced Single Radio Voice call continuity (SRVCC) / IMS CentralizedServices (ICS)

The integration of the LTE network with the 2G/3G network based on Enhanced SingleRadio Voice Call Continuity (SR-VCC)/IMS Centralized Services architecture is shownin 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 LTEnetwork lies within IMS network. The Service Control and Centralization Applicationserver (SCC AS) in the IMS network is the responsible element for anchoring the callin IMS. The SIP signalling messages from the user attached to the LTE network andthe destination user is relayed via the SCC AS. The mobile is also assigned a SessionTransfer Number for SRVCC (STN-SR) by the SCC AS during the initial registrationand is used during handover of the call from the LTE to 2G/3G network. The handoverprocedure 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 ofLTE coverage is detected based on the measurement reports from the UE.

• The MME splits the voice and data bearers and initiates a handover proceduretowards the Enhanced MSC server in the CS domain.



• The Enhanced MSC server in the CS domain is responsible for reservation ofbearers in the CS domain. This is done by forwarding the handover requestmessage to the target MSC server to which the LTE user will be registered in theCS domain.

• In addition, the enhanced MSC server initiates the transfer of the call in theIMS domain, by using the STN-SR. The SCC AS in the IMS domain executesthe session transfer procedure in IMS domain and the media bearer is switchedtowards 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 tothe MME.

• The MME sends a handover command to the UE via the E-Node B and the UEattaches to the CS domain by following the CS domain attach procedure and thecall 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 - SRVCCand leads towards complete integration of 2G/3G networks with LTE network. TheICS architecture is same as shown in Figure 5.4. The difference lies in the fact thatin ICS based approach calls that are originating in 2G/3G network i.e. calls fromlegacy mobiles are also anchored at IMS network. Since, the call control for both theCS (2G/3G) and PS (LTE) domains is within the IMS, seamless handover of usersbetween 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 theintegration of the legacy CS networks with the LTE network. During the initial rollout of LTE when the coverage is minimum, the SRVCC based approach enables easyroaming between LTE and 2G/3G domains. It also enables the users to experiencethe 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 thelegacy networks like 2G/3G become obsolete and a common infrastructure would savelot of costs in operating these networks.

The drawback of SRVCC/ICS based approach is, it requires significant upgradesin the existing CS networks. In the CS domain, elements like MSC server are tobe upgraded which involve significant costs. The other major limitation is the longhandover time when a user moves from LTE to 2G domain which causes a significantdisruption when there is an ongoing call in the LTE network.



Chapter 6

Conclusion

6-1 Conclusion

In this thesis, the performance of voice over LTE is analysed when the transport networkis congested with data traffic. The analysis was carried out using OPNET simulationtool. The LTE model in OPNET had significant limitations in the classification of bear-ers in the transport network. This led to a situation where there was no prioritizationof the bearers in the transport network. Hence to being with, various functions in theprocess models of the E-Node B and EPC were modified to achieve proper classificationof bearers in the transport network. The modification was performed such that eachbearer in the LTE network will be mapped to a specific DSCP in the IP header. Thisenabled us to do classification of IP packets for different services like voice and FTP inthe transport network.

The importance of classification of voice and data traffic in the transport networkis realized from the results of Section 4.2. Without proper classification, we see thatthere is a 50 percent increase in the packet end to end delay for voice even whenthere is no congestion in the transport network. In Sections 4.3 and 4.4, the role ofscheduling algorithms on the performance of voice and data was analysed. We see thatthe capacity for voice users is higher when there is absolute priority for voice in thetransport network. But there is a drawback of using Strict Priority for voice, as thereis a significant degradation in the performance of data traffic at times of congestion.

The Weighted Fair and Weighted Round Robin algorithms were used to overcomethis drawback. A comparative analysis was carried out to understand the performanceof 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 bearersare mapped to a AF service class in the transport network. This is mainly due to the


56 Conclusion

reduced bandwidth allocated for voice users in the AF class than in the EF class. Thereis a trade-off between the performance of voice and data traffic, depending on the typeof classification in the transport network.

Finally, we present a different approach where voice users are mapped into two prior-ity levels and mapped to both EF and AF classes in the transport network. With thisapproach, we see that we can add more voice users in the network within the acceptableQoS levels than mapping voice into a single EF or AF service class in the transport net-work. At times of high congestion, there is a significant reduction in the performanceof voice users belonging to the AF service but in a more controlled manner. Such anapproach 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 thereis 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 perfor-mance of VoIP in the network. LTE supports very high data rates and hence serviceslike Video streaming, Interactive gaming can also be used by the mobile users. In thecurrent LTE specifications, there are different bearers that have defined for each serviceas seen in Table 2.1. But the mapping of these bearers to IP based QoS is also impor-tant for classification in the transport network. In the future, when there are no CSbased 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 arepresent in the network. In such scenarios, when there is multiple level of classificationfor different types of services, each type of service has different QoS requirements andmapping 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 inthe transport network become much more important owing to the fact that over pro-visioning of bandwidth for one type of service has a direct impact on the performanceof other service.

The usage of admission control is another area that needs to be investigated. Mostof 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 theradio and core networks, efficient link usage in the transport network can be achievedwithout increasing the link capacity in the transport network.



Bibliography

[1] 3GPP Technical Specification 23.272, "Circuit Switched (CS) fallback in EvolvedPacket System (EPS)", Stage 2 (Release 10); http://www.3gpp.org, 2011.

[2] Alcatel-Lucent Strategic White Paper, "Options for Providing Voice over LTE andTheir 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 satisfac-tion in LTE networks with mixed traffic," IEEE 19th International Symposium onPersonal, 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 FairScheduling 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 Ve-hicular Technology Conference, vol., no., pp.2502-2506, 11-14 May 2008.

[6] Yasir Zaki, Nokila Zahariev, Thushara Weerawardane, Carmelita Görg and AndreasTimm-Giel, "Optimized Service Aware LTEMAC Scheduler: Design, Implementationand Performance Evaluation", OPNET workshop, Washington, D.C., August 29-September 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 Wirelessand 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 Communi-cations Magazine , vol.47, no.2, pp.76-83, February 2009.

[9] 3GPP Technical Specification 23.203, "Policy and charging control architecture (Re-lease 11)", www.3gpp.org, 2012


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 architecturefor Differentiated Services", "Request for Comments 2475, Internet Engineering TaskForce", December 1998.

[12] A. Demers, S. Keshav, and S. Shenker "Analysis and simulation of a fair queueingalgorithm", In Symposium proceedings on Communications architectures and proto-cols ", 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 intransmission planning", 2011.

[15] 3GPP Technical Specification 23.401, "General Packet Radio Service (GPRS) en-hancements 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 Conti-nuity (SRVCC),Stage 2(Release 11) " http://www.3gpp.org;, 2012.

[17] 3GPP Technical Specification 23.292, " IMS Centralized Services Stage 2(Release11) " http://www.3gpp.org, 2012.



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


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



masters thesis voice over lte.pdf

Documents

long term evolution

prasanna gururaj raghavendrarao

weighted round robin

round robin manner

ip multimedia subsystem

evolved packet system

long term solution

existing cs ps networks