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WiMAX AND WLAN NETWORKS FOR VOICE OVER IP APPLICATION

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AAA Authentication, Authorization and Accounting
ADPCM Adaptive Differential Pulse Coded Modulation
AES-CCM Advanced Encryption Standard Counter with
CBC MAC
AMC Adaptive Modulation and Coding
AP Access point
ARQ Automatic Repeat Request
ASN Access Service Network
AWGN Adaptive White Gaussian Noise
BE Best Effort Service
BPSK Binary Phase Shift Keying
BSS Base Service Set
BWA Broadband Wireless access
CBR Constant Bit Rate
CID Connection Identifier
CS Convergence Sub-layer
CS-ACELP Conjugate Structure Algebraic-Code Excited
Linear Prediction
CSMA/CA Carrier Sense Multiple Access/ Collision
Avoidance
CSN Connectivity Service Network
CCA Clear Channel Assessment
DBPSK Differential Binary Phase Shift Keying
DCF Distributed Coordination Function
DCME Digital Circuit Multiplication Equipment
DHCP Dynamic Host Control Protocol
DL Downlink
DLC Data Link Control Layer
DL-MAP Downlink Map
DOCSIS Data over cable service interface specification
DQPSK Differential Quadrature Phase Shift Keying
DSL Digital Subscriber Line
DSSS Direct Sequence Spread Spectrum
EAP Extensible Authentication Protocol
ertPS extended real time Polling Service
ESS Extended Service Set
FDD Frequency Division Duplexing
FHSS Frequency Hop Spread Spectrum
FTP File Transfer Protocol
GFSK Gaussian Frequency Shift Keying
GRE Generic Routing Encapsulation
IETF-EAP Internet Engineering Task Force-Extensible
Authentication Protocol
IEEE Institute of Electrical and Electronic Engineers
IP Internet Protocol
IR Infra Red
ISI Inter Symbol Interference
ISM Industrial, Scientific and Medical
ITU-T Telecommunication Standardization Sector of the
International Telecommunications Union
LAN Local Area Network
LD-CELP Low-Delay Code Excited Linear Prediction
LLC Logical Link Control
LOS Line Of Sight
MAC Medium Access Control
MAC CPS MAC Common Part Sub-layer
MN Mobile Node
MOS Mean Opinion Score
MS Mobile Station
nrtPS non-real time Polling Service
NSP Network Service Provider
NWG Network Working Group
OFDMA Orthogonal Frequency Division Multiple Access
PC Point coordinator
PCF Point Coordination Function
PHY Physical layer
PLCP Physical Layer Convergence Protocol
PMD Physical Medium Dependent
PMKv2 Privacy and Key Management Protocol version 2
PPP Point to Point Protocol
PSTN Public Switched Telephone Network
PTM Point To Multipoint
PTP Point To Point
QAM Quadrature Amplitude Modulation
QoS Quality of Service
QPSK Quadrature Phase Shift Keying
RLC Radio Link Control
rtPS real time Polling Service
RTS/CTS Request-To-Send/ Clear-To-Send
SDU Service Data Units
SIP Session Initiation Protocol
SISO Single Input Single Output
SONET Synchronous Optical Network
SS Subscriber Station
TDD Time Division Duplexing

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WiMAX AND WLAN NETWORKS FOR VOICE OVER IP APPLICATION

  1. 1. WWW.ThesisScientist.com PERFORMANCE EVALUATION OF INTEGRATED WiMAX AND WLAN NETWORKS FOR VOICE OVER IP APPLICATION
  2. 2. WWW.ThesisScientist.com Abbreviations AAA Authentication, Authorization and Accounting ADPCM Adaptive Differential Pulse Coded Modulation AES-CCM Advanced Encryption Standard Counter with CBC MAC AMC Adaptive Modulation and Coding AP Access point ARQ Automatic Repeat Request ASN Access Service Network AWGN Adaptive White Gaussian Noise BE Best Effort Service BPSK Binary Phase Shift Keying BSS Base Service Set BWA Broadband Wireless access CBR Constant Bit Rate CID Connection Identifier CS Convergence Sub-layer CS-ACELP Conjugate Structure Algebraic-Code Excited Linear Prediction CSMA/CA Carrier Sense Multiple Access/ Collision Avoidance CSN Connectivity Service Network CCA Clear Channel Assessment DBPSK Differential Binary Phase Shift Keying DCF Distributed Coordination Function DCME Digital Circuit Multiplication Equipment DHCP Dynamic Host Control Protocol DL Downlink DLC Data Link Control Layer
  3. 3. WWW.ThesisScientist.com DL-MAP Downlink Map DOCSIS Data over cable service interface specification DQPSK Differential Quadrature Phase Shift Keying DSL Digital Subscriber Line DSSS Direct Sequence Spread Spectrum EAP Extensible Authentication Protocol ertPS extended real time Polling Service ESS Extended Service Set FDD Frequency Division Duplexing FHSS Frequency Hop Spread Spectrum FTP File Transfer Protocol GFSK Gaussian Frequency Shift Keying GRE Generic Routing Encapsulation IETF-EAP Internet Engineering Task Force-Extensible Authentication Protocol IEEE Institute of Electrical and Electronic Engineers IP Internet Protocol IR Infra Red ISI Inter Symbol Interference ISM Industrial, Scientific and Medical ITU-T Telecommunication Standardization Sector of the International Telecommunications Union LAN Local Area Network LD-CELP Low-Delay Code Excited Linear Prediction LLC Logical Link Control LOS Line Of Sight MAC Medium Access Control MAC CPS MAC Common Part Sub-layer MN Mobile Node
  4. 4. WWW.ThesisScientist.com MOS Mean Opinion Score MS Mobile Station nrtPS non-real time Polling Service NSP Network Service Provider NWG Network Working Group OFDMA Orthogonal Frequency Division Multiple Access PC Point coordinator PCF Point Coordination Function PHY Physical layer PLCP Physical Layer Convergence Protocol PMD Physical Medium Dependent PMKv2 Privacy and Key Management Protocol version 2 PPP Point to Point Protocol PSTN Public Switched Telephone Network PTM Point To Multipoint PTP Point To Point QAM Quadrature Amplitude Modulation QoS Quality of Service QPSK Quadrature Phase Shift Keying RLC Radio Link Control rtPS real time Polling Service RTS/CTS Request-To-Send/ Clear-To-Send SDU Service Data Units SIP Session Initiation Protocol SISO Single Input Single Output SONET Synchronous Optical Network SS Subscriber Station TDD Time Division Duplexing
  5. 5. WWW.ThesisScientist.com TDM Time Division Multiplexed TDMA Time Division Multiple Access TEK Traffic Encryption Key UGS Unsolicited Grant Service UL Uplink UL-MAP Uplink Map VBR Variable Bit Rate VoIP Voice over IP Wi-Fi Wireless Fidelity WiMAX Worldwide Interoperability for Microwave Access Wireless MAN Wireless Metropolitan Area Network Wireless HUMAN Wireless High-Speed Unlicensed Metropolitan Area Networks WLAN Wireless Local Area Network WWAN Wireless Wide Area Network
  6. 6. WWW.ThesisScientist.com Contents Chapter 1: Introduction……………………………………………..1 1.1 Introduction………….…………………..……………………….2 1.2 Voice over IP (VoIP).……………………………………………4 1.3 Problem Statement….……………………………………………9 1.4 Related Work……….……………………..……………………10 1.5 Thesis outline……….…………………..………………………10 Chapter 2: Effect of Terrain and Pathloss on Wireless Networks.12 2.1 Introduction…………………………………………………..…13 2.2 Background………………………………………………..……13 2.2.1 Free Space Propagation Model……………………………….…...13 2.2.2 Erceg‟s Suburban Fixed Model………………………………........14 2.2.3 Outdoor-to-Indoor and pedestrian pathloss Model…………..……16 2.2.4. Vehicular environment…………………………………….…..….16 2.3. Simulation Setup…………………………………………….....16 2.4. Simulation Results and Discussion……………………….……18 2.4.1 Pathloss……………………………………………………..……..18 2.4.2 Average Throughput………………………………………………19 2.4.3 Average Packet end to end delay……………………………...….20 2.5 Conclusion…………………………………………………..….21 Chapter 3: VoIP over WiMAX Network…………………………..22 3.1 Introduction……………………………………………………..23 3.2 Background of IEEE 802.16 and WiMAX …………………….23
  7. 7. WWW.ThesisScientist.com 3.3 Overview of 802.16 MAC Layer………………….……………25 3.4 Overview of 802.16 PHY Layer………………………………..28 3.5 WiMAX Network Architecture…………………...……………29 3.6 Experimental Setup……………………………………………..31 3.6.1 Scenario 1: Simulation Setup and Results………………………....31 3.6.1.1 Average Jitter…………………………………………....32 3.6.1.2 Average Packet End to End Delay……………………...34 3.6.1.3 Average MOS…………………………………………..35 3.6.2 Scenario 2: Simulation Setup and Results…………………………35 3.6.2.1 Average Jitter and packet end to end delay……………..37 3.7 Conclusion…………………………………………...………….38 Chapter 4: Voice over WLAN……………………………...…...….39 4.1 Introduction……………………………………………...……...40 4.2 Background of IEEE 802.11 and WLAN……………….………40 4.3 Protocol Architecture…………………………………………...41 4.4 Overview of IEEE 802.11 MAC Layer…………………………43 4.4.1 IEEE 802.11 MAC: DCF and PCF………………………..………43 4.5 Overview of IEEE 802.11 PHY layer……………………...…...44 4.5.1 Frequency Hop Spread Spectrum (FHSS)…………………………44 4.5.2 Dynamic Sequence Spread Spectrum (DSSS)……………….……44 4.5.3 Infrared……………………………………………………….……45 4.6 WLAN System Architecture………………..……………….….45 4.6.1 Infrastructure mode……………………………………45 4.6.2 Adhoc mode…………………………………………...46 4.7 Experimental Setup……………………………………………..47
  8. 8. WWW.ThesisScientist.com 4.7.1 Scenario 1: Simulation Setup and Results…………………………47 4.7.1.1 Average Jitter……………………………………………48 4.7.1.2 Average Packet End to End Delay……………………...49 4.7.1.3 Average MOS…………………………………………..50 4.7.2 Scenario 2: Simulation Setup and Results…………………………51 4.7.2.1 Average Jitter……………………………………………52 4.7.2.2 Average Packet End to End Delay……………………...52 4.8 Conclusion………………………………………………………53 Chapter 5: Voice over WiMAX-WLAN Integrated Network……54 5.1 Introduction……………………………………………………..55 5.2 WLAN vs. WiMAX…………………………………………….56 5.3 Types of Integration…………………………………………….58 5.3.1 Loose Coupling……………………………………………………58 5.3.2 Tight Coupling………………………………………………….…58 5.4 Experimental Setup……………………………………………..59 5.4.1 Scenario 1: Simulation Setup and Results…………………………59 5.4.1.1 Average Jitter……………………………………………60 5.4.1.2 Average MOS …………………………………………..61 5.4.1.3 Average Packet End-to-End Delay……………………...62 5.4.2 Scenario 2: Simulation Setup and results………………………….63 5.4.2.1 Average jitter………………………………...………….63 5.4.2.2 Average Packet End-to-End Delay……………………...64 5.6 Conclusion………………………………………………………65 Chapter 6: Development of Scenarios in OPNET ……..…………66 6.1 Why OPNET?..............................................................................67
  9. 9. WWW.ThesisScientist.com 6.2 Development of WiMAX in OPNET…………………………...67 6.2.1 Static Scenario……………………………………………………..68 6.2.2 Mobile Scenario……………………………………………………76 6.3 Development of WLAN in OPNET…………………………….82 6.3.1 Static Scenario……………………………………………………..82 6.3.2 Mobile Scenario……………………………………………………87 6.4 Development of WLAN-WiMAX Integrated Scenario in OPNET……………………………………………………………...88 6.4.1 Static Scenario……………………………………………………..88 6.4.2 Mobile Scenario……………………………………………………90 Chapter 7: Comparative Analysis….………………………………92 7.1 Introduction……………………………………………………..93 7.2 Comparative Results……………………………………………93 7.2.1 Stationary network…………………………………………………93 7.2.2 Mobile network……………………………………………………97 7.3 Conclusion………………………………………………………99 Chapter 8: Conclusion and Future Work………………………..100 Bibliography
  10. 10. WWW.ThesisScientist.com Chapter 1 Introduction Chapter Outline:-  Introduction   Voice over IP (VoIP)   Problem Statement   Related Work   Thesis Outline 1
  11. 11. WWW.ThesisScientis t.com Introduction 1.1 Introduction Wireless networking has become an essential part in the modern telecommunication system. The demand of high speed data transfer with quality has led to the evolution of technologies like WiMAX and WLAN and is still increasing. Hence, new ways to enhance quality and speed of connectivity are being searched for. With step towards the fourth generation communication networks, integrated networks are coming into operation. Also voice over IP is expected to be a low cost communication medium. The voice codecs are big constraints which affect the quality of the voice in a network. Hence, before real time deployment of VoIP over a network it is necessary to evaluate the voice performance over varying networks for various codecs. WLANs [1] are mostly designed for private wired LANs and have been enormously successful for data traffic but voice traffic differs fundamentally from data traffic in its sensitivity to delay and loss [2]. Voice over WLAN is popular, but maintaining the speech quality is still one of many technical challenges of the VoIP system. VoIP is spreading rapidly and there is need to support multiple concurrent VoIP communications but WLAN support handful number of users [3] [4]. The IEEE 802.11 MAC specifies two different mechanisms, namely the contention-based Distributed Coordination Function (DCF) [1] and the polling-based Point Coordination Function (PCF) [1]. The DCF uses a carrier sense multiple access with collision avoidance (CSMA/CA) scheme for medium access and the optional four way handshaking request-to-send/clear-to-send [1] mechanism (RTS/CTS). Incapability of providing differentiation and prioritization based upon traffic type results in providing satisfactory performance for best-effort traffic only, but inferior support for QoS requirements posed by real time traffic. These requirements make the DCF scheme a less feasible option to support QoS for VoIP traffic. The PCF mode enables the polled stations to transmit data without contending for the channel. Studies on VoIP over WLAN in PCF mode [5] shows that the polling overhead is high with increased number of stations in a basic service set (BSS). This results in excessive delay and poor performance of VoIP 2
  12. 12. WWW.ThesisScientist.com under heavy load conditions. Thus, both DCF and PCF have limited support for real-time applications. Supporting VoIP over WLAN using DCF mode poses significant challenges, because the performance characteristics of their physical and MAC layers are much worse than their wired counterparts and hence considered in our system. WiMAX (Worldwide Interoperability for Microwave Access) [6][16] on the other hand is designed to deliver a metro area broadband wireless access (BWA) service. So, while wireless LAN supports transmission range of up to few hundred meters, WiMAX system ranges up to 30 miles [6]. Unlike a typical IEEE 802.11 WLAN with 11Mbps bandwidth which supports very limited VoIP connections [4], an IEEE 802.16 WiMAX with 70Mbps bandwidth [7] can support huge number of users. These motivations led to study and comparison of the VoIP quality of service in IEEE 802.11b WLAN and IEEE 802.16 WiMAX network. IEEE 802.16 support 5 types of service classes, namely UGS (Unsolicited Grant Service), rtPS (real time Polling Service), nrtPS (non-real time Polling Service), BE (Best Effort Service), ertPS (extended rtPS service) [8]. UGS supports fixed-size data packets at a constant bit rate (CBR). It supports real time applications like VoIP or streaming applications but wastes bandwidth during the off periods. rtPS supports variable bit rate(VBR) real-time service such as VoIP. Delay-tolerant data streams such as an FTP is designed to be supported by the nrtPS. This requires variable-size data grants at a minimum guaranteed rate. The nrtPS is similar to the rtPS but allows contention based polling. Data streams, such as Web browsing, that do not require a minimum service-level guarantee is supported by BE service. BE connections are never polled but receive resources through contention. ertPS was introduced to support VBR real-time services such as VoIP and video streaming. It has an advantage over UGS and rtPS for VoIP applications as it carries lower overhead than UGS and rtPS [9] and hence is modeled in the system. Since, WiMAX is expected to create the opportunity to successfully penetrate the commercial barrier by providing higher bandwidth, establishing wireless commons becomes an important factor. Also, bandwidth crunch and network integration are some of the major technical and social challenges regarding the future of the community-based Wi-Fi networks [10]. According to [10], the foundation of the WiMAX PTP commons is the process of hot-spot 3
  13. 13. WWW.ThesisScientist.com interconnection and integration. Instead of global Internet connectivity, many current applications and businesses are expected to be better utilized by using the localized Wi-Fi constellation. With a step towards the next generation, it is expected that an integrated network as shown in Figure 1.1, comprising of both the WiMAX and WLAN network and using mobile nodes with dual stack is expected to provide a better performance than a similar WiMAX or WLAN network. FIGURE 1.1. Wi-Fi integration using WiMAX [10]. VoIP has been widely accepted for its cost effectiveness and easy implementation. A VoIP system consists of three indispensable components, namely 1) codec, 2) packetizer, and 3) playout buffer. Analog voice signals are compressed, and encoded into digital voice streams by the codecs. The output digital voice streams are then packed into constant-bit-rate (CBR) voice packets by the packetizer. A two way conversation is very sensitive to packet delay jitter but can tolerate certain degree of packet loss. Hence a playout buffer is used at the receiver end to smooth the speech by removing the delay jitter. Quality of noise sensitive VoIP is usually measured in terms of jitter, MOS and packet end-to-end delay. Perceived voice with zero jitter, high MOS and low packet end-to-end delay is considered to be the best. With the two competing wireless networks namely WLAN and WiMAX, this paper analyses the perceived voice quality as measured using OPNET simulation environment. 1.2 Voice over IP (VoIP) Voice is analog and is converted to digital format before transmitting over Internet. This process is called encoding and the converse is called decoding and both are performed by voice codecs [11]. With bandwidth utilization becoming a huge concern, voice compression techniques are used [11] to reduce bandwidth consumption. Voice compression by a codec adds an additional overhead of 4
  14. 14. WWW.ThesisScientist.com algorithmic delay. Thus, a codec is expected to provide good voice quality even after compression, with minimum delay. Table 1.1 shows the bandwidth requirements of some common codecs. G.711 is the international standard for encoding telephone audio. It has a fixed bit rate of 64kbps. G.723 and G.729 are low bit rate codecs at the expense of high codec complexity. G.723 is one of the most efficient codecs with the highest compression ratio and is used in video conferencing applications. G.729 is an industry standard with high bandwidth utilization for toll-quality voice calls [12]. G.726 uses ADPCM speech codec standard, and transmits at rates of 16, 24, 32, and 40 kbps. G.728 officially codes speech at 16 Kbit/s using low-delay code excited linear prediction [13]. For example, during a call using G.711 as codec, the amount of data transfer for both uplink and downlink will be 87.2 x 2 = 174.4Kbps = 0.1703 Mbps = 10.21 MB per minute. So, G.711 uses 10.21 Mb/min per VoIP call where as G.729 uses 0.5MB/min per voice call in the same way. TABLE 1.1. Bandwidth Requirement of Some Common Codecs [11][14] Codecs Algorithm Bandwidth Ethernet (Kbps) Bandwidth Usage (Kbps) G.711 PCM(Pulse Code Modulation) 64 87.2 G.729 CS-ACELP (Conjugate Structure 8 31.2 Algebraic-Code Excited Linear Prediction) G.723.1 Multi Rate Coder 6.3 21.9 G.723.1 Multi Rate Coder 5.3 20.8 G.726 ADPCM(Adaptive Differential 32 55.2 Pulse Code Modulation) G.726 ADPCM(Adaptive Differential 24 47.2 Pulse Code Modulation) G.728 LD-CELP (Low-Delay Code 16 31.5 Excited Linear Prediction) 5
  15. 15. WWW.ThesisScientist.com Figure 1.2. Packet Creation based on voice activity. Moreover, recently voice codecs are developed to detect talk-spurt [15] and silence lengths [15] within a conversation. Silence in a communication period leads to packetization of the background noise and sending it over the network. This causes bandwidth wastage. Usually, during a conversation we talk 35% of the time and remain quiet rest of the time [15]. With silence suppression during the silence period, the codec does not send data as shown in Figure 1.2. This decreases channel utilisation and thereby saves bandwidth. Voice communication is noise sensitive. Noise causes the signal to reach the destination with a lead or lag in the time period. This deviation is called jitter. Lead causes negative jitter and lag causes positive jitter and both degrade the voice quality. The time taken by voice to be transmitted from the mouth of the sender to the ear of the receiver is called packet end-to-end delay. The packet end-to-end delay should be very less for voice communication. Perceived voice quality is typically estimated by the subjective mean opinion score (MOS), an arithmetic average of opinion score. MOS of a particular codec is the average mark given by a panel of auditors listening to several recorded samples. It ranges from 1(unacceptable) to 5 (excellent). It depends on delay and packet dropped by the network. The E-model, an analytical model defined in ITU-T recommendation, provides a framework for an objective on-line quality estimation based on network performance measurements like delay and loss and application level factors like low bit rate codecs. The result of the E-model is the calculation of the R-factor (best case 100 worst case 0) [5]. R = R0 – Is – Id – Ie + A. (1.1) Where R0 groups the effects of noise, Is includes the effects of the other impairments related to the quantisation of the voice signal, Id represents the impairment caused due to delay, Ie covers the impairments caused by the low bit 6
  16. 16. WWW.ThesisScientist.com rate codecs and packet losses. The advantage factor A compensates for the above impairments under various user conditions. A is 10 for mobile telephony but 0 for VoIP [5]. R0 is considered to be 94.77 and Is is considered to be 1.43 in OPNET 14.5.A. The relation between MOS and R-factor: MOS = 1 + 0.035R + 7.10-6R(R – 60) (100 – R) (1.2) Packet networks operate on packet switching principles; hence voice in an IP or WiMAX network would be transmitted to the destination as a collection of packets where each one might follow different routes, thus arrive at the destination with different delays. Typically originating and terminating parts would respectively be the in-house portions of an enterprise network. The part lying in Internet or some public WAN (i.e. shared medium) is termed the core network. The packet flow is assumed to be from left to right. For flows in the opposite direction, the terminology of originating and terminating is interchanged. Unless stated otherwise, in view of delay calculations, originating and termination parts are regarded to be identical. On the originating side, the analog voice signal is digitized into pulse code modulation (PCM) signals by the voice codecs. Then the PCM samples are compressed and converted into packet format, thus ready to be sent across the net. For some network configurations, the edge router may also perform codec and compression functions. In the subsections below, we attempt to describe various kinds of delays together with their associated formulation in units of milliseconds (ms). o Look Ahead Delay: The codecs conforming to different standards may use voice blocks of different sizes. The block size used by the G.729 A codec is 10 ms whereas; codecs of G.723.1 have block sizes of 30 ms [16]-[18]. The present packet is checked while the previous one is being compressed. Note that such an operation is not possible at the very beginning of transmission. Thus, a fixed delay of 5 ms, called look-ahead delay, occurs in this section of packet voice transmission [16]-[17]. o Encoding and Compression Delay: Generally the G.729 A type codec is used in VoIP having a mean opinion score (MOS) value of 4.2 which almost provides toll quality [19]. This type of codec has an 8 kbps modulation rate, hence producing a 10 ms delay for encoding the packet [17]-[24]. 7
  17. 17. WWW.ThesisScientist.com o Packetization Delay: The time spent for packetization of encoded and compressed data packets is called packetization delay. To support a good quality voice call, the packetization delay should be less than 30 ms [17]. The packetization process commences after storing the packets into a buffer. As long as the buffering time remains below 10 ms, the compression and buffering periods will overlap and there will be no additional delays introduced at the buffer [17]. Otherwise, the buffering time would exceed the compression time, and the remaining period would add to the total delay created during the buffering operation. o Serialization delay (SD): Serialization of the data/voice packets to the core network is going to be implemented at the ingress point of the originating network. This creates an extra delay, which may be calculated as [19] SD= (payload + 48) x 8 / Link Speed (1.3) The term “payload” corresponds to the average number of bytes stored in payload of the packet and 48 indicates the number of bytes added by real time protocol, user datagram protocol and internet protocol (RTP/UDP/IP) headers including some optimizations [17], [19]. o Network Delays: An IP packet suffers several other delays while traversing the whole network. The sources of these delays are explained below: o Switching Delay: The packets wait for 10 ms at each switch in the originating network and 1 ms in the core network [17]. The total switching delay becomes Switching delay = NRo x 10 + NRc x 1 (1.4) Where NRo represents the number of routers in the core network and the NRc represents the number of routers in the core network. In the simulation, the originating/terminating hop counts are both taken as 1. o Propagation Delay (PD): Assuming that a signal propagates with a velocity of two thirds of the speed of light, each km of propagation will contribute an extra 5 μs delay. The propagation delay may be written as PD = Distance (in km) x 5 x 0.001 (1.5) o Data Queuing Delay (DQD): The packets are serial, so they are served sequentially. The waiting time of each packet will be 8
  18. 18. WWW.ThesisScientist.com different, depending on the packet size. Also, these queuing delays will also vary depending on the transmitted IP packet size. Because of these complications, data queuing delay is not easy to formulate. In the simulation, IP is assumed to carry only data packets. Therefore, the voice packet will have the highest priority as there will be no video packets. In this way the data packets will therefore jump ahead of the other packets and wait only for the one that is currently being processed. Even with this assumption an accurate prediction of queuing delay is still impossible, but may be simplified to [19]. o Voice/Data Contention Delay (VDCD): Voice data contention delay is expressed as VDCD = (MTU + 48) / Link Speed (1.6) The term MTU (maximum transferable unit) appears, because in worst case assumption, the maximum payload size of the data packet will be equal to MTU. o Jitter Buffer Delay: Each packet serialized to the network will follow a different route according to the status of network and routers at that moment, so each packet will arrive to destination with a different delay which is proportional to the maximum jitter size. o Voice Decoding Delay: The decoding time for each packet at the receiver is given by [19]. Voice Decoding Delay = Encoding Delay (1.7) Since a G.729 codec imposes an encoding delay of 10 ms [17]-[23] our decoding delay will simply be 10 ms for each packet. The purpose of this modeling is to compare the performance parameters for the voice codecs considering both with and without silence suppression in WiMAX 802.16d, WLAN 802.11b and their integrated network and thereby show that the integration provides optimal network capacity and quality of service. 1.3 Problem Statement With voice over IP coming into existence, the maximum utilisation of the spectrum has also become a concern. To accommodate maximum number of users with 9
  19. 19. WWW.ThesisScientist.com considerably good quality of voice has become a concern of the researchers. Innovative methods and techniques are coming up each day to suffice the needs. With the next generation communication networks coming into being our target is to analyse of the performance of voice over IP over a WLAN-WiMAX integrated 4G network. WLAN is chosen because it is a widely used low cost technology and WiMAX is expected to outshine the conventional DSL cables for providing broadband wireless access. Hence, it is expected that an integrated WLAN-WiMAX 1.4 Related Work [1], [2], [4], [5], [7] and [13] discusses the performance and capacity of the WLAN networks while [8], [9], [16] discusses the capacity and performance of WiMAX network. The idea of the voice codecs has been discussed in [15]. It also discusses the concept of silence suppression in it. In [10] it is proposed that an integrated architecture of WiMAX and WLAN is expected to perform better than the conventional WiMAX or WLAN network. Among them [2], [7] and [13] discusses the capacity improvement of WLAN network for voice traffic while quality of the voice traffic has been taken care of in [4] and [5]. Papers [6] [8] [9] have discussed the performance of the WiMAX network for voice over IP and multimedia traffic. To the best of our knowledge, the performance of WLAN- WiMAX integrated network has not been evaluated yet. In this thesis we provide a comparative discussion of the performance of a WLAN-WiMAX integrated network with respect to the upcoming application voice over IP. 1.5 Thesis Outline The thesis is organised as follows:  Chapter 2 discusses the performance of WiMAX network for various pathloss models.   Chapter 3 discusses the performance of a WiMAX network in both static and mobile conditions for the application voice over IP   Chapter 4 discusses the performance of a conventional WLAN network in both static and mobile conditions for the application voice over IP. 10
  20. 20. WWW.ThesisScientist.com  Chapter 5 discusses the performance of the WLAN-WiMAX integrated network in both static and mobile conditions for the voice over IP application.   Chapter 6 discusses the simulation setup in OPNET to perform the simulations in the aforesaid chapters   Chapter 7 shows the comparative study of the network performances for VoIP in both static and moving conditions   Chapter 8 discusses the future work that can be done as an extension to this thesis and finally concludes the thesis. 11
  21. 21. WWW.ThesisScientist.com Chapter 2 Effect of Terrain and Pathloss on Wireless Networks Chapter Outline:-  Introduction   Background  Free Space Propagation Model Erceg’s Suburban Fixed Model  Outdoor-to-Indoor and pedestrian pathloss Model Vehicular environment   Simulation Setup   Simulation Results and Discussion Pathloss Average Throughput  Average Packet end to end delay   Conclusion 12
  22. 22. WWW.ThesisScientist.com Effect of Terrain and Pathloss on Wireless Networks 2.1. Introduction With various wireless technologies coming up, performance of the wireless networks has become a concern. Mobile WiMAX is expected to be the wireless technology of the next generation as it supports vehicular mobility with broad coverage area. Among various other parameters which affect network performance, the terrain feature on which the network is deployed and the path loss model affects the network performance substantially. Air being the communicating medium in wireless networks, any hindrance on the communication path like buildings, trees, etc., affects the wireless signal propagation. Hence, prior network deployment, analysis of the terrain is essential. Voice over IP as discussed in Chapter 1 is expected to be the communication medium of the next generation and File Transfer Protocol is one of the most popular data transfer protocol. Hence, in this chapter the behaviour of various propagation models in WiMAX network with respect to path loss, throughput and delay for various terrain models with voice and FTP applications is studied. 2.2. Background The common propagation models namely Free Space path loss model, Suburban Fixed (Erceg), Outdoor to Indoor and Pedestrian Environment and Vehicular Environment are discussed in this paper. The Suburban Fixed path loss model is further subdivided in three categories based on the building and tree density namely terrain type A, B and C. The models are briefly discussed in the next sections. 2.2.1. Free Space Propagation model The free space propagation model is mathematically given by: 13
  23. 23. WWW.ThesisScientist.com (2.1) Where, Prx is received power in watts and is a function of distance between transmitter and receiver, Ptx is the transmitted power in watts, Grx and Gtx are the gain of the receiving and transmitting antennas respectively, L is the system-loss factor and is not related to propagation. It is usually greater than 1 and λ is the wavelength in meters [25]. 2.2.2. Erceg’s Suburban Fixed Model The Erceg model is based on extensive experimental data collected at 1.9GHz in 95 macro cells of suburban areas across the United States [26]. This model is a slope intercept model given by [26]: (2.2) Where, PL is the instantaneous attenuation, H is the intercept and is given by free space path loss at the desired frequency over a distance of d0 = 100 meters: (2.3) Where, λ is the wavelength. The parameter is a Gaussian random variable over the population of macro cells within each terrain category. It can be written as [26] [27]. (2.4) where hb is the height of the base station antenna in meters, σγ is the standard deviation of , x is a zero-mean Gaussian variable of unity standard deviation N[0, 1], and e, g, k and σγ are all data-derived constants for each terrain category. The shadow fading components varies randomly from one terminal location to another within any given macro-cell. It is a zero-mean Gaussian variable and can be expressed as [26] [27]. (2.5) 14
  24. 24. WWW.ThesisScientist.com where y and z are the zero-mean Gaussian variables of unit standard deviation N[0, 1], σ is the standard deviation of s, µσ is the mean of σ, and σσ is the standard deviation of σ. µσ and σσ are both data-derived constants for each terrain category. The numerical values of the above parameters are given in Table 2.1. The correlation factors of the model for the operating frequency and for the MS antenna height are given in [27] as: for Terrain type A and for Terrain type C Where, f is the frequency in MHz and hr is the height is the MS antenna above ground in meters. The Terrain Type A is a hilly terrain with moderate to heavy tree density, representing rural environments and has highest path loss. Terrain Type B is characterized by either a mostly flat terrain with moderate to heavy tree density or a hilly terrain with light tree density. Terrain Type C is a flat terrain with light tree density and is associated with the lowest path loss for rural environments [27]. TABLE 2.1. NUMERICAL VALUES CONSIDERED FOR THE PARAMETERS [26] Parameters Terrain Terrain Terrain Type A Type B Type C e 4.6 4.0 3.6 g m 1 0.0075 0.0065 0.005 k m 12.6 17.1 20.0 0.57 0.75 0.59 10.6 9.6 8.2 2.3 3.0 1.6 15
  25. 25. WWW.ThesisScientist.com 2.2.3. Outdoor-to-Indoor and Pedestrian Pathloss Model This environment is characterized by small cells and low transmission power. Base stations with low antenna heights are located outdoors; pedestrian users are located on streets and inside buildings and residences. (2.9) The above equation describes the path loss in dB where R is the distance between the base station and the mobile station in kilometres and „f‟ is the carrier frequency of 2000 MHz for IMT-2000 band application [28]. 2.2.4. Vehicular environment This environment is characterized by larger cells and higher transmits power. The path loss in vehicular environment in dB is given by: hb) log10 R – 18 log10 hb – 21 log10 f + 80dB (2.1 where, R is the distance between the base station and the mobile station, and f is the carrier frequency of 2000 MHz and Δhb is the base station antenna height in meters measured from the average roof top level. 2.3. Simulation Setup The Wireless Deployment Wizard of OPNET is used to deploy a 7 celled WiMAX network, with multiple subscriber stations in the range of a base station as shown in Figure 2.1. The base stations are connected to the core network by an IP backbone. There is a server backbone containing the voice server which is configured as the SIP server. The IP backbone is connected to the server backbone via an ASN gateway. This node, configured as the ASN gateway supports the mobility in the WiMAX network. The IP backbone, server backbone and the ASN gateway together represent the service provider company network. The green bidirectional dotted lines represent the generic routing encapsulation (GRE) tunnels. The cell radius is set to 30 kilometres. There are 10 nodes under the base station 2. 5 of these nodes are communicating with 5 mobile nodes under base station 7 and other 5 are communicating with 5 mobile nodes under base station 3. Voice call of PSTN quality is configured between these mobile nodes. The nodes participating in the same session are connected by the blue bidirectional dotted lines. The mobile nodes under base station 2 are configured to move at a speed of 50 km/hr in the path as shown by the white lines in Figure 2.1. The remaining key 16 P L = 4 0 ( 1 – 4 X 1 0- 3
  26. 26. WWW.ThesisScientist.com network configuration parameters in OPNET are summarized as in Table 2.2 and the attributes of the network components are shown in Table 2.3. FIGURE 2.1. Network Model for WiMAX. TABLE 2.2. NETWORK CONFIGURATION DETAILS Network Cell Radius No. of Base Stations No. of Subscriber Stations per BS No. of Mobile nodes in the network Speed of the mobile nodes Simulation time Base Station Model Subscriber Station Model ASN Gateway Model IP Backbone Model Voice Server Model Link Model (BS-Backbone) Link Model (ASN - Backbone) 7 celled WiMAX network 30km 7 10 10 50 km/hr 600 sec wimax_bs_ethernet4_slip4_router wimax_ss_wkstn ethernet4_slip8_gtwy ip32_cloud ppp_server PPP_DS3 PPP_SONET_OC12 17
  27. 27. WWW.ThesisScientist.com TABLE 2.3. ATTRIBUTES OF THE NETWORK COMPONENTS Attributes Value Physical Layer Model OFDMA 20Mhz MAC Protocol IEEE 802.16e Multipath Channel Model ITU Vehicular A No. of Transmitter per BS SISO Traffic Type of Service Interactive Voice and Data Scheduling Type ertPS, nrtPS Application Voice (PSTN quality), FTP Voice Codec G 711 FTP Load High 2.4. Simulation Results and Discussion The performance metric used to analyse the network performance is path loss, average throughput of the WiMAX network and packet end to end delay. Using OPNET, the simulation is conducted for the common path loss models and the three types of terrains: Terrain Types A, B, C of erceg path loss model. We are interested to investigate the combined effect of terrain (buildings, trees, etc.) and vehicular mobility on the Mobile WiMAX performance in OPNET. To investigate the combined effect of terrain and vehicular mobility, the terrains are simulated by choosing the terrain type in OPNET which is thereby selected by the simulator, based on the actual location and surrounding terrain of the transmitter-receiver pair. 2.4.1. Pathloss Path loss models are broadly classified into four groups namely free space, Suburban fixed, Outdoor to indoor and pedestrian, vehicular environment. The path loss due to these models in a WiMAX network with mobile nodes moving at a speed of 50 km/hr is modeled in OPNET modeler and the result is shown in Figure 2.2. 18
  28. 28. WWW.ThesisScientist.com FIGURE 2.2. Path loss due to various pathloss models in decibel. From the graph it is observed that the path loss for outdoor to indoor and pedestrian is the highest and the same for free space is the lowest. This is due to the fact that the path loss value varies with the amount of reflection in the communicating path. As a mobile node moves from outdoor to indoor the number of reflections in the communicating path increases rapidly thereby causing huge path loss. Also, we see that the path loss for free space is the least. This is concluded in [27] that the path loss increases with the number of obstructions in the communication path. Free Space implies a terrain with no or very less obstruction. Hence, number of obstruction decreases and pathloss thereby is minimum. 2.4.2. Average Throughput This metric measure the amount of voice traffic and FTP traffic received in bits per second on an average for each connection. From Figure 2.3, it is observed that the throughput of the network with free space path loss model is the highest and the same for outdoor to indoor and pedestrian is the lowest for both voice and FTP traffic. This is due to the fact that as the density of obstacles increases, the Line of Sight (LOS) gets affected. Thus the number of times the signal gets obstructed and reflected is more. This results in increasing attenuation and diffraction due to the building structures, trees or mountains. As the Line of Sight (LOS) between the transmitting and receiving nodes decreases, it causes delay. This results in packet loss thereby causing fall in the average throughput. 19
  29. 29. WWW.ThesisScientist.com FIGURE 2.3. WiMAX Network Throughput in Bits per second 2.4.3. Average Packet end to end delay This metric measures the time taken by each voice packet to travel from the mouth of the transmitter to the ear of the receiver and each FTP packet from the application layer of the sender to the application layer of the receiver, on average for each connection. From Figure 2.4, it is observed that the packet end to end delay of the network with free space as the pathloss model is the lowest and the same for outdoor to indoor and pedestrian is the highest irrespective of the application. This is due to the fact that as the density of the building structures, trees or mountains increases, the number of times the signal gets obstructed and reflected is more thereby increasing attenuation and diffraction. Hence, the LOS between the transmitting and receiving nodes decreases. This causes the mobile node to get disconnected from the base stations and thereby perform re-registration [1] to get re-connected to a base station. The registration procedure is time consuming and hence adds up to the packet end to end delay. With flat terrain model and very less or nil obstructions, the mobile node remains connected to some or the other base stations and thus the network scanning and registration process is avoided thereby causing less delay as shown in Figure. 2.4. 20
  30. 30. WWW.ThesisScientist.com FIGURE 2.4. Packet end to end delay in seconds 2.5. Conclusion The target of this work was to check the variation of the network performance due to varying pathloss and terrain models. This work shows the variation in the WiMAX network performance with varying path loss models deployed over suburban areas of terrain types varying from hilly terrain with moderate to heavy tree density to flat terrain with light tree density for Voice over IP and File Transfer Protocol application with mobile nodes moving at a speed of 50 km/hr. The aim was to evaluate the effect of various path loss models over varying suburban terrains, on the basis of average throughput and packet end to end delay and path loss of Mobile WiMAX. The reduction in Line-Of-Sight due to the terrain model of any area influences the network throughput directly by increasing the attenuation and diffraction losses, and indirectly affects the packet end to end delay by causing nodes to initiate network re-registration more frequently under fluctuating cell coverage. This is already mentioned in [26] [27] [29]. We conclude that our results tally with the already taken measurements of the various path loss models in [26] [27] [29] and thus, free space pathloss model is chosen for the future works as it is a basic pathloss models with all other parameters related to terrain and building density set as constant. 21
  31. 31. WWW.ThesisScientist.com Chapter 3 VoIP over WiMAX Network Chapter Outline:-  Introduction   Background of IEEE 802.16 and WiMAX   Overview of 802.16 MAC Layer   Overview of 802.16 PHY Layer   WiMAX Network Architecture   Experimental Setup  Scenario 1: Simulation Setup and Results Average Jitter  Average Packet End to End Delay Average MOS  Scenario 2: Simulation Setup and Results Average Jitter  Average Packet End to End Delay   Conclusion 22
  32. 32. WWW.ThesisScientist.com VoIP over WiMAX Network 3.1. Introduction In recent years, internet access has moved to a new dimension. It is now not restricted to Web browsing and emailing. Multimedia services including Voice-over-IP (VOIP) and media streaming have become the expectation of the next generation. To provide users with such connectivity, Broadband Wireless Access (BWA) comes into the picture. It promises users to be provided with megabit internet access seamlessly. One of the many technologies under BWA is WiMAX (Worldwide Interoperability for Microwave Access). Based on IEEE 802.16 it has been designed to provide metro area broadband wireless access. With 70 Mbps speed and over 50 miles of coverage area [30], WiMAX supports mobility up to 70-80 miles/hr and is expected to be the replacement of cable and Digital Subscriber Line (DSL) [31]. Hence, with an overview of the IEEE 802.16 BWA network, a discussion of the quality of voice achieved by using a WiMAX network supporting both in static and moving nodes is done in this chapter. 3.2. Background of IEEE 802.16 and WiMAX The IEEE 802.16 group was formed in 1998 to develop an air interface standard for wireless broadband. The group‟s initial focus was the development of a LOS based point-to-multipoint wireless broadband system for operation in the 10-66 GHz millimeter waveband. The resulting standard – the original IEEE 802.16 standard completed in December 2001 – was based on a single carrier physical (PHY) layer with a burst time division multiplexed (TDM) MAC layer. Many of the concepts related to the MAC layer were adapted for wireless from the popular cable modem DOCSIS (data over cable service interface specification) standard. The IEEE 802.16 group subsequently produced 802.16a, an amendment to the standard, to include NLOS applications in the 2 GHz – 11 GHz band using orthogonal frequency division multiple access (OFDMA), were also included. Further revisions resulted in a new standard in 2004, called IEEE 802.16d – 2004 which replaced all prior versions and formed the basis for the first WiMAX 23
  33. 33. WWW.ThesisScientist.com solution. These early WiMAX solutions based on IEEE 802.16d – 2004 targeted fixed applications, and are referred to as Fixed WiMAX [32]. In December 2005, the IEEE Group completed and approved IEEE 802.16e – 2005, an amendment to the IEEE 802.16d – 2004 standard that added mobility support. The IEEE 802.16e – 2005 forms the basis for the WiMAX solution for nomadic and mobile applications and is often referred to as mobile WiMAX [33]. The basic characteristics of various IEEE 802.16 standards are summarized in Table 3.1. TABLE 3.1. Basic Characteristics of various IEEE 802.16 standards. 802.16 802.16d – 2004 802.16e – 2005 Completed Completed June Completed Status December December 2004 2001 2005 10GHz-66GHz for Frequency fixed; 10GHz-66GHz 2GHz-11GHz 2GHz-11GHz for Band mobile applications Application Fixed LOS Fixed NLOS Fixed and Mobile NLOS MAC Point-to- Point-to-Multipoint, Point-to- Multipoint, Architecture mesh Multipoint, mesh mesh Single carrier, 256 Single Carrier, 256 OFDM OR Transmission Single Carrier OFDM or 2048 Scalable OFDM Scheme only OFDM with 128, 512, 1024, or 2048 subcarriers Modulation QPSK,16QAM, QPSK, 16QAM, 64 QPSK, 16QAM, 64QAM QAM 64 QAM Gross Data 32Mbps- 1Mbps – 75Mbps 1Mbps – 75Mbps Rate 134.4Mbps 24
  34. 34. WWW.ThesisScientist.com Multiplexing Burst Burst TDM/ TDMA/ Burst TDM/ TDM/TDMA OFDMA TDMA/ OFDMA Duplexing TDD and FDD TDD and FDD TDD and FDD 1.75 MHz, 3.5 MHz,7 1.75 MHz, 3.5 20MHz, MHz, 14 MHz, 1.25 MHz,7 MHz, 14 Channel 25MHz, MHz, 5 MHz, 10 MHz, 1.25 MHz, Bandwidths 28MHz MHz, 15 MHz, 8.75 5 MHz, 10 MHz, MHz 15 MHz, 8.75 MHz WirelessMAN – SCa WirelessMAN – SCaWirelessMAN – – Air-interface WirelessMAN OFDM WirelessMAN OFDM designation -SC WirelessMAN– WirelessMAN – OFDMS OFDMA WirelessHUMAN WirelessHUMAN WiMAX None 256 – OFDM as Fixed Scalable OFDMA implementation WiMAX as Mobile WiMAX 3.3. Overview of 802.16 MAC Layer IEEE 802.16 Medium Access Control (MAC) generally follows point to multi-point (PMP) network topology with support for mesh topology. From the reference model as shown in Figure 3.1 there are three sub layers in the MAC: Service Specific Convergence Sub-Layer (CS): providing any transformation or mapping of external network data through CS SAP (CS Service Access Point) MAC Common Part Sub-layer (MAC CPS): classifying external network service data units (SDUs) and associating these SDUs to proper MAC service flow and connection identifier (CID). Multiple CS specifications are provided to interface with various protocols. Privacy (or Security) Sub-layer: supporting authentication, secure key 25
  35. 35. WWW.ThesisScientist.com exchange, and encryption. FIGURE 3.1. Reference Model of IEEE 802.16 Unlike the random access techniques of typical MACs of IEEE 802, IEEE 802.16 MAC is connection oriented and similar to time division multiple access (TDMA). As a subscriber station (SS) enters the network, it creates one or more connections with the base station (BS). It also performs link adaptation and automatic repeat request (ARQ) functions to maintain the target bit error rate. Further to support multimedia services, the IEEE 802.16 MAC may have to use radio resources and provide quality of service (QoS) differentiation in services which are not considered typical MAC functions. The five different types of services to support QoS differentiation for different applications provided by IEEE 802.16 are as follows [35]: o Unsolicited Grant Services (UGS): UGS is designed to support fixed-size data packets at a constant bit rate (CBR). It supports real time applications like T1/E1 emulations but wastes bandwidth during the off periods or silence periods. Hence, it cannot be used for Voice over IP without silence suppression. o Real-Time Polling Services (rtPS): It is designed to support variable sized data packets, periodically like MPEG video and Voice over IP with silence suppression. o Non-Real-Time Polling Services (nrtPS): nrtPS is designed to support non real time services that require data bursts of variable size on a regular basis like FTP. It is similar to rtPS except for the contention based polling scheme. 26
  36. 36. WWW.ThesisScientist.com o Best Effort (BE): Data streams that do not require a minimum service-level guarantee is supported by BE service. It counts typical data traffic like Internet web browsing and FTP file transfer. o Extended Real-Time Polling Service (ertPS): It is designed to support variable rate real-time services such as Voice over IP and video streaming. It has an advantageous over UGS and rtPS for VoIP applications because it carries lower overhead than UGS and rtPS [9] and hence is considered in our system. To support OFDMA PHY, the MAC layer is responsible for assigning frames to the proper zones and exchanging this structure information with the SSs in the DL and UL maps. IEEE 802.16 MAC is connection oriented. As BS controls the access to the medium, bandwidth is granted to the SSs on demand. At the beginning of each frame, the BS schedules the uplink and downlink grants to meet the negotiated QoS requirements. Each SS learns the boundaries of its allocation under current uplink and sub-frame via the UL-MAP message. The DL-MAP delivers the timetable of downlink grants in the downlink sub frame. There are different bandwidth request schemes in WiMAX. For UGS, a fixed amount of bandwidth is requested periodically at the set-up phase of uplink and no bandwidth is requested explicitly after that. The unicast poll allocates necessary bandwidth for a polled uplink connection. The broadcast polls are issued by the BS to all uplink connections, while a truncated exponential back-off algorithm is employed to resolve possible collisions in polling. Based on the bandwidth requested and granted, the BS uplink scheduler estimates the residential backlog at each uplink connection, and allocates future grants. An SS scheduler must be implemented with each SS MAC, in order to re-distribute the granted capacity to all its connections. However, note that IEEE 802.16 does not specify scheduling algorithms that are left to manufacturers. Similar to the concept of cellular layer - 2/3, IEEE 802.16 MAC has the radio link control (RLC) to control PHY transition from one burst profile to another, in addition to traditional power control and ranging. Another important sub-layer in the IEEE 802.16 MAC is the security sub- layer, and an improved version has been developed for the IEEE 802.16e. The Privacy and Key Management Protocol version 2 (PMKv2) is the basis of Mobile WiMAX security. Device and user authentication adopts IETF EAP protocol. The 27
  37. 37. WWW.ThesisScientist.com traffic encryption follows the IEEE 802.11i using AES-CCM to protect traffic data. The keys used to derive the cipher text are generated from the EAP authentication. To avoid further attacks and hostile analysis, a periodic key (TEK) refreshing mechanism enables improved protection. A three-way handshake scheme in Mobile WiMAX optimizes the re-authentication mechanism for fast handover by preventing man-in-the-middle-attacks. To deal with mobility in Mobile WiMAX, IEEE 802.16e the MAC specifies MAC layer handover procedure, while the exact handover decision algorithm is not specifically defined. Further details regarding the IEEE 802.16 MAC are discussed in [30] [32] [33] [35] [9] [36] [37]. 3.4. Overview of 802.16 PHY Layer The Figure 3.2 shows the schematic representation of WiMAX PHY layer. The diagram has basically three blocks which are o Transmitter, o Receiver and o Channel. FIGURE 3.2. Schematic Model of WiMAX PHY Layer The transmitter consists of channel encoding, digital modulations, serial to parallel conversion block followed by an OFDM modulation block and a parallel to serial conversion block has been shown that has the specific purpose of serializing the data bits. The channel encoding block has “block encoder”, “convolutional encoder” and “puncture vector block” associated with it. The redundancy added to the data sequence is the sole purpose of the encoder block. In WiMAX scenario, we can 28
  38. 38. WWW.ThesisScientist.com only opt for BPSK, QPSK, 16-QAM and 64-QAM modulation. WiMAX PHY uses Adaptive Modulation and Coding (AMC) which takes into account the channel SNR to dynamically select the proper modulation technique appropriate for that channel condition to deliver the maximum throughput. Serial to parallel conversion of the incoming bit becomes the critical set of operation that determines the parallel transmission of data bits. OFDM modulation being the key multiplexing technique in WiMAX helps the transmission of data bits at a very high rate at a negligible amount of Inter Symbol Interference (ISI) with minimal amount of packet loss and bit error. The data transmitted through wireless channel reaches the receiver. The channel which might be AWGN, Rayleigh etc. determines the effective channel impairment introduced in the receiver. The receiver section does exactly the opposite to the transmitter. 3.5. WiMAX Network Architecture The IEEE 802.16e – 2005 standard provides the air interface for WiMAX but does not define the full end-to-end WiMAX network. The WiMAX Forum's Network Working Group (NWG) is responsible for developing the end-to-end network requirements, architecture, and protocols for WiMAX, using IEEE 802.16e – 2005 as the air interface. The WiMAX NWG has developed a network reference model to serve as an architecture framework for WiMAX deployments and to ensure interoperability among various WiMAX equipment and operators. The network reference model envisions unified network architecture for supporting fixed, nomadic, and mobile deployments and is based on an IP service model. Below is simplified illustration of IP-based WiMAX network architecture. The overall network may be logically divided into three parts: o Mobile Stations (MS) used by the end user to access the network. o The Access Service Network (ASN), which comprises one or more base stations and one or more ASN gateways that form the radio access network at the edge. The ASN gateway typically acts as a layer 2 traffic aggregation points within an ASN. Additional functions that may be part of the ASN gateway include intra-ASN location management and paging, radio resource management and admission control, caching of subscriber profiles and encryption keys, AAA client functionality, establishment and 29
  39. 39. WWW.ThesisScientist.com management of mobility tunnel with base stations, QoS and policy enforcement, and foreign agent functionality for mobile IP, and routing to the selected CSN. o Connectivity Service Network (CSN), which provides connectivity to the Internet, ASP, other public networks, and corporate networks. The CSN is owned by the NSP and includes AAA servers that support authentication for the devices, users, and specific services. The CSN also provides per user policy management of QoS and security. The CSN is also responsible for IP address management, support for roaming between different NSPs, location management between ASNs, and mobility and roaming between ASNs. The network reference model developed by the WiMAX Forum NWG defines a number of functional entities and interfaces between those entities. Figure 3.3 below shows some of the more important functional entities. o Base station (BS): The BS is responsible for providing the air interface to the MS. Additional functions that may be part of the BS are micro mobility management functions, such as handoff triggering and tunnel establishment, radio resource management, QoS policy enforcement, traffic classification, DHCP (Dynamic Host Control Protocol) proxy, key management, session management, and multicast group management. FIGURE 3.3. WiMAX Network Architecture [39] 30
  40. 40. WWW.ThesisScientist.com 3.6. Experimental Setup To analyze VoIP in a network, it is necessary to study real life scenarios. Hence, OPNET 14.5.A is chosen as the simulation tool as it will reflect the actual deployment of the WiMAX network. 3.6.1. Scenario 1 – Static Nodes Figure 3.4 shows the simulation setup used for WiMAX network. Using the Wireless Deployment Wizard of OPNET a 7 celled WiMAX network, with multiple subscriber stations in the range of a base station is deployed. The base station is connected to the core network by a server backbone via an IP backbone. The server backbone is further connected to the voice server which is configured as the SIP server. The base station, IP Backbone, Server Backbone and the Voice Server together represent the service provider company network. The cell radius is set to be 30 km. The Base Station transmission power is set to 10W and the same for subscriber station is set to 0.5W based on [35] as shown in Figures 3.5 and 3.6 respectively. The number of subscribers in cells 2 and 3 are 10 and Voice over IP calls are setup between them in mesh using SIP. FIGURE 3.4. Network Model for WiMAX in OPNET 31
  41. 41. WWW.ThesisScientist.com FIGURE 3.5. Base Station Parameters FIGURE 3.6. Subscriber Station Parameters With the simulation setup as mentioned before, the voice codecs being used for the Voice over IP calls are varied and the corresponding variation in voice jitter, MOS (Mean Optimal Score) and Packet End-to-end delay are noted. 3.6.1.1. Average Jitter Figure 3.7 shows the variation of jitter for the WiMAX network without using silence suppression for various codecs. Perceived voice quality is best if the jitter is zero. As shown in the figure, average voice jitter is almost 0 for the voice codecs G 723.1 with both 5.3Kbps and 6.3Kbps and G 726 with 32Kbps implying very good quality of voice whereas all other codecs shows some deviation. A positive jitter of 0.000000925926 seconds is shown by G 711 while all others show negative jitter of about 0.000001841621 seconds. Since, the bit rate of G 723.1 is 6.3 or 5.3 Kbps; it results in generation of small packets. As stated in [15] modem and fax signals cannot be carried by G 723.1 and hence it can be used only for narrow band communications. Like G 711, G 726 has its roots in the PSTN network. Hence, it is expected to provide users with good quality of voice. It 32
  42. 42. WWW.ThesisScientist.com is primarily used for international trunks to save bandwidth. Unlike G 711, G 726 uses 32Kbps to provide nearly the same quality of voice. This is because 32 Kbps is its de facto standard [34]. To increase the number of users supported count, silence suppression technique is important. As mentioned in Chapter 1, human speech during one way conversation has 35% talk spurt length and 65% silence length. Silence suppression prevents the packetization of the silence length and thereby save bandwidth. But usage of silence suppression technique increases the positive jitter considerably compared to the other voice codecs. This is shown in Figure 3.8. Hence, G 726 cannot be used in cases where silence suppression technique is used irrespective of its performance in cases where silence suppression has not been used. Mean Jitter without silence suppression (sec) 0.0000015 0.000001 Time(sec) 0.0000005 0 G729 G711 G723_1_5_3 G723_1_6_3 G726_16 G726_24 G726_32 G726_40 G728_12_8 G728_16 -0.0000005 -0.000001 -0.0000015 -0.000002 Voice Codecs FIGURE 3.7. Average voice jitter without Silence Suppression Mean Jitter with Silence Suppression (sec) 0.03 0.025 Time(sec) 0.02 0.015 0.01 0.005 0 -0.005 G729 G711 G723_1_5_3 G723_1_6_3 G726_16 G726_24 G726_32 G726_40 G728_12_8 G728_16 Voice Codecs FIGURE 3.8. Average voice jitter using Silence Suppression 33
  43. 43. WWW.ThesisScientist.com 3.6.1.2. Average Packet End to End Delay As stated in Chapter 1, packet networks operate on packet switching principles; hence voice in an IP or WiMAX network would be transmitted to the destination as a collection of packets where each one might follow different routes, thus arrive at the destination with different delays. Factors affecting packet end to end delay include Look Ahead Delay, Packetization Delay, Serialization Delay, Network Delay, etc. (Details covered in Chapter 1). Figure 3.9 and 3.10 shows, the Packet End-to-End delay for the voice codec G 723.1 is the highest irrespective of silence suppression. This is because; G 723.1 uses coding rate of 5.3 Kbps or 6.3 Kbps which results in the formation of packets of smaller size and larger count. Now as the number of packets increases in the network, the congestion in the network increases. Congestion directly affects the network packet delay and thus results in increased packet end to end delay. Mean Packet-End-to-End Delay without Silence Suppression 0.14 0.12 (sec) 0.1 0.08 Time 0.06 0.04 0.02 0 G729 G711 G723_1_5_3 G723_1_6_3 G726_16 G726_24 G726_32 G726_40 G728_12_8 G728_16 Voice Codecs FIGURE 3.9. Average packet end to end delay without Silence Suppression Mean Packet End-to-End Delay with Silence Suppression 0.14 0.12 (sec) 0.1 0.08 Time 0.06 0.04 0.02 0 G729 G711 G723_1_5_3 G723_1_6_3 G726_16 G726_24 G726_32 G726_40 G728_12_8 G728_16 Voice Codecs FIGURE 3.10. Average packet end to end delay using Silence Suppression Also the Look Ahead delay of G 723.1 voice codec is 7.5 msec [35] while the same for G 729A is 5 msec and other considered voice codecs is 0 msec. On the other hand the serialization delay of G 726 32Kbps is quite high. As mentioned in Chapter 1 the serialization delay is directly proportional to the payload size and is inversely proportional to the link speed. Since the test network has same link 34
  44. 44. WWW.ThesisScientist.com configuration, the link speed is same for all codecs. Hence, depending on the payload size, the variation in the serialization delay affects the total packet end to end delay significantly. 3.6.1.3. Average MOS Mean MOS without Silence Suppression 4 3.5 3 MOS 2.5 2 1.5 1 0.5 0 G729 G711 G723_1_5_3 G723_1_6_3 G726_16 G726_24 G726_32 G726_40 G728_12_8 G728_16 Voice Codecs FIGURE 3.11. Average MOS without Silence Suppression Average MOS with Silence Suppression 4 3.5 3 MOS 2.5 2 1.5 1 0.5 0 G729 G711 G723_1_5_3 G723_1_6_3 G726_16 G726_24 G726_32 G726_40 G728_12_8 G728_16 Voice Codecs FIGURE 3.12. Average MOS with Silence Suppression The Mean Optimal Score (MOS) as shown in Figures 3.11 and 3.12 is independent of Silence Suppression. As mentioned in Chapter 1, MOS depends on number of packets dropped. G 723.1 is a low bit rate codec which generates packets of size 5.3 Kbps or 6.3 Kbps. This results in network congestion and thereby packet drop. Hence, the MOS value for the voice codec G 723.1 is quite low. Voice having MOS of 3 can be considered to be of considerable quality. Hence, all other codecs considerable with respect to their MOS value. 3.6.2. Scenario 2 – Mobile Nodes Figure 3.13 shows the simulation setup used for WiMAX network. Using the Wireless Deployment Wizard of OPNET a 7 celled WiMAX network, with multiple subscriber stations in the range of a base station is deployed. The 35
  45. 45. WWW.ThesisScientist.com base station is connected to the core network by a server backbone via an IP backbone and an ASN Gateway which controls the mobility of the mobile nodes. The server backbone is further connected to the voice server which is configured as the SIP server. The base station, IP Backbone, Server Backbone, ASN Gateway and the Voice Server together represent the service provider company network. Generic Routing Encapsulation (GRE) tunnel is setup between the ASN gateway and the base stations to control the mobility of the mobile nodes (for further details refer to Appendix A). The cell radius is set to be 30 km. The mobile nodes are configured to move at a rate of 50 km/hr, 100 km/hr and 150 km/hr. The Base Station transmission power is set to 10W and the same for subscriber station is set to 0.5W based on [35]. The number of subscribers in cells 2 is 10 and the same in cells 3 and 7 are 5 each. Voice over IP calls are setup between them using SIP and are shown by the blue lines in the Figure 3.13. With the simulation setup as mentioned before, the voice codecs being used for the Voice over IP calls are varied and the corresponding variation in voice jitter and Packet End-to-end delay are noted. Due to the limitation of OPNET, Mean Optimal Score (MOS) could not be estimated. Also only some of the voice codecs could be modeled. G 723.1 with 5.3 kbps, G 726 with 32 Kbps and G 728 with 16 Kbps could only be modeled. FIGURE 3.13. Network Model for WiMAX 36
  46. 46. WWW.ThesisScientist.com 3.6.2.1. Average Jitter Mean Jitter(sec) 0.02 0.018 0.016 0.014 J i t t e r ( s e c ) 0.012 150km/hr0. 008 50km/hr 0.01 100km/hr 0.006 0.004 0.002 0 G711 G723.1 G726 G728 G729 Voice Codec FIGURE 3.14. Mean Jitter without Silence Suppression Average Packet end to end delay 0.09 0.08 D e l a y ( s e c ) 0.07 0.06 0.05 50km/hr 100km/hr 0.04 P a c k e t 150km/hr 0.03 0.02 0.01 0 G711 G723.1 G726 G728 G729 Voice Codecs FIGURE 3.15 Mean Packet end to end delay without Silence Suppression G 711 codec has packet rate of 64 Kbps which is quite large compared to that of G 723.1 which is about 5.3 Kbps. Thus, G711 has fewer number of packets compared to G 723.1 for fixed amount of voice. For larger packets the overhead due to header is small [38]. As a result, for scenario 1, the delay for G 711 is lesser compared to G 723.1. Moreover, in scenario 1, the nodes are fixed hence the packets follow almost the same route and arrive in order more or less but wireless networks, have a inherent property of discarding any packets containing one or more erroneous bits. So the probability of discarding G723.1 packets are less and as a result the jitter is also minimum for this codec. However in the mobile scenario, scenario 2, the probability of packet loss is more and any packet loss in the physical layer is interpreted as delay by the upper layers due to the hiding property of the MAC or link layer protocols [38]. Since, 37
  47. 47. WWW.ThesisScientist.com the packet size of G 711 is larger; it has higher probability of suffering from packet loss than G 723.1 thereby having higher delay than G 723.1. This is shown in Figure 3.15. As the subscriber stations become mobile, their supporting base stations changes frequently due to handover and the mobile nodes get distributed between different base stations depending upon their trajectory. Hence, the packets may follow different path and arrive out-of-order. Since, the number of packets for G 723.1 is much more than G 711, this problem is more severe than G 711. As a result it suffers from maximum jitter. This is shown in Figure 3.14. 3.7. Conclusion The target of this work is to observe the variation of jitter and packet end to end delay in a WiMAX network under various mobility scenarios. This work shows the variation of the mentioned parameters with respect to a stationary network having all stationary nodes and with respect to a mobile network with varying speed of 50km/hr, 100km/hr and 150km/hr. Average packet end to end delay is more in case of stationary nodes than mobile nodes. Figures 3.7, 3.8 and 3.14 show the variation of jitter in the above mentioned networks. As Figure 3.7 and 3.8 shows, the jitter in a stationary network increases considerably when silence suppression is activated in the network and it increases further when mobility is introduced in the network. This is because as a node becomes mobile it undergoes handover. This creates inconsistency in the order of the packets delivered thereby jitter. Figures 3.9, 3.10 and 3.15 show the variation of packet end to end delay in the above mentioned WiMAX network. From Figures 3.9 and 3.10, it is observed that the packet end to end delay decreases in the overall network when silence suppression is activated. This is due to the reduction in the queuing delay. As the number of packets decreases with silence suppression, the number of packets queued in the intermediate nodes decreases. This reduces the queuing delay. The packet end to end delay decreases considerably when mobility is introduced in the network. With mobility, the mobile nodes get distributed under multiple base stations and this removes the bottle neck of one base station in case of stationary nodes and thereby decreases the delay. With a step towards the 4G mobile networks, this study is used in the following chapters to compare and evaluate the performance of the loosely coupled WiMAX WLAN integrated network. 38
  48. 48. WWW.ThesisScientist.com Chapter 4 VoIP over WLAN Network Chapter Outline:-  Introduction   Background of IEEE 802.11 and WLAN   Protocol Architecture   Overview of IEEE 802.11 MAC Layer  IEEE 802.11 MAC: DCF and PCF   Overview of IEEE 802.11 PHY layer  Frequency Hop Spread Spectrum (FHSS)  Dynamic Sequence Spread Spectrum (DSSS) Infrared   WLAN System Architecture   Experimental Setup  Scenario 1: Simulation Setup and Results Average Jitter  Average Packet End to End Delay Average MOS  Scenario 2: Simulation Setup and Results Average Jitter  Average Packet End to End Delay   Conclusion 39
  49. 49. WWW.ThesisScientist.com VoIP over WLAN Network 4.1 Introduction WLANs [1] are mostly designed for private wired LANs and have been enormously successful for data traffic like (email, media downloads etc). WLANs are being studied as an alternative to the high installation and maintenance costs incurred by traditional addition, removal, and changes experienced in wired LAN infrastructures. It is an economic way to provide users with ubiquitous connectivity - anywhere, anytime - to the Internet and to private and corporate networks. It offers typical gross data rate of 10-50 Mbps as opposed to 10-100 kbps offered by cellular technology and compared to the peak 70 Mbps offered by WiMAX. Voice traffic differs from data traffic mainly in its sensitivity to delay and loss [2]. Although existing WLAN applications are mainly data centric, there is a growing demand for real-time voice services over WLAN. Driven by these two popular technologies, VoIP over WLAN has been emerging as an infrastructure to provide low-cost wireless voice services. 4.2 Background of IEEE 802.11 and WLAN The first generation of WLANs operated in the 900 MHz ISM band, with symbol rates of around 500 Kbps, but they were exclusively proprietary, non- standard systems, developed to provide wireless connectivity for specific niche markets. The second-generation systems came around 1997. They operated in the 2.4 GHz range and provided symbol rates of around 2Mbps. The standards of WLAN are as follows: o IEEE 802.11b (Wi-Fi 2.4 GHz): The goal of the Task Group b was to increase the maximum bit rate in the 2.4 GHz frequency range while maintaining interoperability with the original standard. The MAC layer was kept and the PHY was redefine to only work with DSSS (Direct Sequence Spread Spectrum), thus increasing the spectral efficiency with bit rates of up to 11 Mbps. 40
  50. 50. WWW.ThesisScientist.com o IEEE 802.11a (Wi-Fi 5.2 GHz): The goal of this group was to provide higher data rates and to port IEEE 802.11 to the newly available U-NII at 5.2 GHz. The original MAC layer was kept and the PHY was reworked to provide rates up to 54 Mbps. Since the available band at U-NII is about 300 MHz, eight non-overlapping bands were defined. The spread spectrum technology used in this case was OFDM (Orthogonal Frequency Division Multiplexing), as DSSS was not efficient at working with these high bit rates. o IEEE 802.11g: Task group g is working on an extension to IEEE 802.11b at 2.4 GHz, enabling transmission at symbol rates of 54 Mbps. There are also other IEEE 802.11 Task Groups that focus on different aspects such as: o 802.11i - wireless security at the Mac layer. o 802.11f - roaming between access points o 802.11e - the Quality of Service The IEEE 802.11 WLAN standard that operate in the 2.4 GHz or 5 GHz unlicensed radio bands; the four are summarized in Table 4.1. TABLE 4.1. Characteristic feature of Wireless LAN Standard Maximum Fallback Channels Frequency Radio Bit Rate Rates Provided Band Technique 802.11b 11 Mbps 5.5 Mbps, 2 Mbps 3 2.4 GHz DSSS 1 Mbps 802.11a 54 Mbps 48 Mbps, 36 Mbps, 12 5 GHz OFDM 24 Mbps, 18 Mbps, 12 Mbps, 9 Mbps, 6 Mbps 802.11g 54 Mbps Same as 802.11a 3 2.4 GHz OFDM 4.3 Protocol Architecture Figure 4.1 shows the protocol Stack of IEEE 802.11. Part of the IEEE 802.11 protocol stack is shown in the figure 4.2. In this figure, only the relevant parts of the protocol stack are depicted, as the upper layers are the same than the 41
  51. 51. WWW.ThesisScientist.com other LANs and only the two layers represented (Physical Layer and Data Link Control Layer) differ. FIGURE 4.1. Protocol Stack FIGURE 4.2. Part of the Protocol Stack In the Physical Layer (PHY), several sub layers appear: o Physical Medium Dependent (PMD). This sub layer‟s functions are the modulation and codification of the signal. o Physical Layer Convergence Protocol (PLCP): This level is in charge of adapting the PHY service to the PMD sub layer. o Physical Management. This level selects the channel to transmit with. Just above the Physical Layer, the Data Link Control Layer (DLC) is formed by the following sub layers: o Medium Access Control (MAC). This sub layer is responsible for the access, fragmentation and ciphering of the signal. o Logical Link Control (LLC). This level makes the assignment of the logical channels to the physical channels. o MAC Management. This level is responsible of the synchronization, roaming, consume management, etc. o Finally, the Station Management is responsible of the coordination of the management functions in the two layers. 42
  52. 52. WWW.ThesisScientist.com 4.4 Overview of the IEEE 802.11 MAC 802.11 support two modes of operation. The first, called Distributed Coordination Function (DCF), does not use any kind of central control and the second, called Point Coordination Function (PCF), and uses the base station to control all activity in its cell. So, DCF can be used by all implementations of WLANs and PCF is optional. These two modes will be explained in detail. 4.4.1 DCF and PCF The original IEEE 802.11 standard [3] specifies two channel access mechanisms: a mandatory contention-based distributed coordination function (DCF) and an optional polling based point coordination function (PCF). The DCF uses a carrier sense multiple access with collision avoidance (CSMA/CA) scheme for medium access and the optional four way handshaking request-to-send/clear- to-send [4] mechanism (RTS/CTS). DCF provides a best effort service and is not capable of providing differentiation and prioritization based upon traffic type. While DCF may provide satisfactory performance in delivering best-effort traffic, it lacks the support for QoS requirements posed by real time traffic, and especially VoIP which has stringent delay requirements. These requirements make the DCF scheme a less feasible option to support QoS for VoIP traffic. PCF mode, with a centralized controller, represented another promising alternative to providing QoS in WLAN [5]. The centralized controller called the point coordinator (PC) has its functionality embedded in the access point (AP) of a BSS. The PC enables polling, enabling the polled stations to transmit data without contending for the channel. Nevertheless, studies on carrying VoIP over WLAN in PCF mode in [5] found that when the number of stations in a basic service set (BSS) is large, the polling overhead is high. This results in excessive end-to-end delay and VoIP gets poor performance under heavy load conditions. Although the PCF mode is designed for real-time traffic, it is not widely deployed due to its inefficient polling schemes, limited quality of service (QoS) provisioning, and implementation complexity. Thus, both DCF and PCF have limited support for real-time applications, the IEEE 802.11e has been proposed to enhance the current 802.11 MAC to support applications with stringent QoS requirements [7], but the wide deployment of the standard is yet to be done. 43
  53. 53. WWW.ThesisScientist.com On the other hand, supporting voice traffic over WLANs using the DCF mode poses significant challenges, because the performance characteristics of their physical and MAC layers are much worse than their wired counterparts and hence considered in our system. 4.5 Overview of IEEE 802.11 PHY layer IEEE 802.11 supports three types of physical layer – Infrared, Frequency Hop Spread Spectrum (FHSS), and Direct Sequence Spread Spectrum (DSSS). The later two are based on radio transmission. All of them include the provision of the clear channel assessment signal (CCA). The CCA is used for medium access and indicates if the medium is idle. The transmission technology determines the technique to obtain this signal. The PHY layer offers a service access point (SAP) with 1 or 2 Mbps data rate to the MAC layer. 4.5.1 Frequency Hop Spread Spectrum (FHSS) FHSS is a spread spectrum technique which allows for the coexistence of multiple networks in the same area by separating different network using different hopping sequence. The standard specifies Gaussian shaped FSK (frequency shift keying), GFSK, as modulation for FHSS PHY. For 1Mbps a two level GFSK is used (i.e. 1 bit mapped to 1 frequency) and for 2 Mbps a four level GFSK is used (i.e. 2 bits are mapped to 1 frequency). Sending and receiving in 1Mbps is mandatory while sending and receiving in 2 Mbps is optional. This facilitated the production of low cost devices for the lower rate only and more powerful devices for both transmission rates in the early days of 802.11. A FHSS frame consists of two parts, namely the PLCP part (preamble and header) and the payload part. PLCP part is always transmitted at 1Mbps while the payload can use 1 or 2 Mbps. Also the MAC data is scrambled using the polynomial s(z) = z 7 + z 4 + 1 for DC blocking and whitening of the spectrum. 4.5.2 Dynamic Sequence Spread Spectrum (DSSS) Direct sequence spread spectrum is the alternative spread spectrum method separating by code and not by frequency. In case of IEEE 802.11 DSSS, spreading is achieved by using the 11-chip Barker sequence (+1,-1,+1,+1,-1,+1,+1,+1,-1,-1,-1). The key characteristics of this method are its robustness against interference 44
  54. 54. WWW.ThesisScientist.com and its insensitivity to multipath propagation (time delay spread). However, the implementation is more complex compared to FHSS. IEEE 802.11 DSSS PHY also uses the 2.4 GHz ISM band and offers both 1 and 2 Mbps data rates. The system uses differential binary phase shift keying (DBPSK) for 1 Mbps and differential quadrature phase shift keying (DQPSK) for 2 Mbps as modulation schemes. The symbol rate is 1MHz resulting in a chipping rate of 11 MHz all bits transmitted by the DSSS PHY are scrambled with the polynomial s(z) = z 7 + z 4 + 1 for DC blocking and whitening of the spectrum. 4.5.3. Infrared The PHY layer which is based on infra red (IR) transmission, uses near visible light at 850-950nm. Infra red light is not regulated apart from safety restrictions. The standard does not require a line of sight (LOS) between sender and receiver, but should also work with diffuse light. This allows for point to multipoint (PMP) communication. The maximum range is about 10m if no sunlight or heat sources interfere with the transmission. Typically, such a network will only work in buildings, e.g. classrooms, meeting rooms, etc. frequency reuse is very simple. A simple wall is more than sufficient to shield one IR based IEEE 802.11 network from another. Today, no products are available that offer infra red communication based on 802.11. Proprietary products offer, e.g. up to 4 Mbps using diffused infra red light. Alternatively, directed infra red communication based on IrDA can be used. 4.6 WLAN System Architecture The architecture of a WLAN system is described according to the IEEE 802.11 standard. The main building block of such architecture is the Base Service Set (BSS). There are two modes of configuration that can be used within the standard: o The “Infrastructure” and o The “Ad Hoc” mode. 4.6.1 Infrastructure mode: In the “Infrastructure” mode, the different BSS are interconnected with each other via a component called the Distribution System (DS). Each BSS has one Access Point (AP), through which the Mobile Nodes (MNs) access to the DS. 45
  55. 55. WWW.ThesisScientist.com These interconnected components form the Extended Service Set (ESS). The ESS is a large coverage area where MNs can get handed over from one BSS to another without changes or notification to higher layers in the protocol stack. Finally, a “Portal” is required to integrate the WLAN architecture into the wired network (e.g., Ethernet) and may be integrated with an AP in a single device attached directly to the DS. Fig 4.3 shows the WLAN topology in infrastructure mode. FIGURE 4.3. Infrastructure based architecture 4.6.2 Ad-Hoc mode: In the “Ad-Hoc” mode, each MN can directly reach any MN within the BSS without going through an intermediate node (i.e., AP). There is no backbone network or distributed system associated with the mobile nodes and the BSS are not connected to the wired network. The Adhoc mode of connection usually covers small area and the nodes are allowed to move only within their BSS. There is no concept of handover to support mobility across a large area. Fig 4.4 shows the topology of WLAN in Adhoc Mode. FIGURE 4.4. Adhoc Architecture 46
  56. 56. WWW.ThesisScientist.com 4.7 Experimental Setup To analyze VoIP in a network, it is necessary to study real life scenarios. Hence, OPNET 14.5.A is chosen as the simulation tool as it will reflect the actual deployment of the WiMAX network. 4.7.1 Scenario 1 – Static Nodes: Figure 4.5 shows the simulation setup used for WLAN (IEEE 802.11b) network. Using the Wireless Deployment Wizard of OPNET a 7 celled WLAN network, with multiple subscriber stations in the range of an Access Point (AP) is deployed. The AP is connected to the core network by a server backbone via an IP backbone. The server backbone is further connected to the voice server which is configured as the SIP server. The base station, IP Backbone, Server Backbone and the Voice Server together represent the service provider company network. The cell radius is set to be 100 m. The numbers of subscribers in cells are 10. Voice over IP calls are setup between the subscriber stations of cells 2 and 3 in mesh using SIP. The parameters set in the Access Points and the Subscriber Stations are shown in figures 4.6 and 4.7 respectively. FIGURE 4.5. Network Model for WLAN in OPNET FIGURE 4.6. Access Point Parameters 47
  57. 57. WWW.ThesisScientist.com FIGURE 4.7. Subscriber Station Parameters With the simulation setup as mentioned, the common voice codecs being used for the Voice over IP calls are varied and the corresponding variation in voice jitter, MOS (Mean Opinion Score) and Packet End-to-end delay are noted. 4.7.1.1 Average Jitter Figure 4.8 shows that all voice codecs except G 723.1 has non zero voice jitter with positive variation. Voice quality is considered to be best if the jitter value is zero. Thus, it can be said from this graph that G 723.1 provides best performance with zero value and G 729 on the other hand produce worst performance with highest jitter of 0.001 seconds. But as stated in [5], since G 723.1 is a low bit rate codec, it generates small packets of 5.3 Kbps and 6.3 Kbps and hence can only be used for narrow band communications. Other than G 723.1, G 711 shows lower jitter compared to others at the cost of bandwidth consumption. G 711 is a high bit rate codec and uses 64Kbps bit rate for data transfer. It provides PSTN quality of voice. As already mentioned in chapter 1 voice communication has 35% voice and 65% silence in it. To accommodate more users, silence present in the voice communication needs to be suppressed. This is expected to save bandwidth. Results pertaining to silence suppression are shown in figure 4.9. Figure 4.9 shows that the order of voice jitter decreases to an order of 10 -5 from 10 -3 when silence is suppressed. This is quite less. After silence suppression G 711 shows highest jitter among all the codecs considered. G 726 (32Kbps) shows average position in both the graphs indicating optimal performance. According to [7], G 726 has its roots in the PSTN network and provides considerably good quality of voice. Since, it shows good performance in both the cases (figures 4.7 and 4.8) it can be used in trunks and thereby can accommodate more users by saving bandwidth. 48
  58. 58. WWW.ThesisScientist.com Average Jitter without using silence suppression 0.0012 Second(sec) 0.001 0.0008 0.0006 0.0004 0.0002 0 G729 G711 G723.1 G723.1 G726 G726 G726 G726 G728 G728 5.3K 6.3K 16K 24K 32K 40K 12.8K 16K Voice Codecs FIGURE 4.8. Average voice jitters without Silence Suppression Average Jitter using silence suppression 0.000016 0.000014 0.000012 Seconds(sec) 0.00001 0.000008 0.000006 0.000004 0.000002 0 -0.000002 G729 G711 G723.1 G723.1 G726 G726 G726 G726 G728 G728 5.3K 6.3K 16K 24K 32K 40K 12.8K 16K Voice Codecs FIGURE 4.9. Average voice jitters using Silence Suppression 4.7.1.2 Average Packet End to End Delay The average packet end to end delay in WLAN network for voice codecs without silence suppression is shown in figure 4.10 and is in the order of seconds. This is because WLAN 802.11b provides a bandwidth of 11 Mbps and the Access Points 2 and 3 [refer to figure 4.5] are overloaded and congested with the voice traffic. This results in a bottleneck situation in the Access Points which results in packet drop due to queue overflow in the access points. Any packet loss in the physical and MAC layers is interpreted as delay by the higher layers. Hence, the delay. Except G 723.1, all other voice codecs shows an average delay of more than 2 seconds if silence period is not suppressed. G 723.1 gives a packet end to end delay of less than 0.25 seconds which is acceptable but can only be used for narrow band communication. With silence suppression the packet end to end delay is considerably less and in the order of 10 -2 seconds. The result is shown in figure 4.11. G 723.1 though giving a good performance without silence suppression shows terrible result when silence is suppressed. The packet end to end delay for G 723.1 increases up to 0.1 seconds while the same for other is about 0.06 49
  59. 59. WWW.ThesisScientist.com approximately which is quite less comparatively. G 723.1 creates small packets of larger count. Though silence suppression decreases its number of packets but still its packet count remains quite high compared to the other codecs. This results in network congestion and thereby increased delay. Also, the look ahead delay of G 723.1 is 7.5msec [8] while the same for G 729 is 5sec and others are zero [8]. This adds up to the delay of G 723.1. Average Packet End to End Delay without using silence suppression 5 4 Delay(sec) 3 2 1 0 G729 G711 G723.1 G723.1 G726 16K G726 24K G726 32K G726 40K G728 G728 16K 5.3K 6.3K 12.8K Voice Codecs FIGURE 4.10. Average packet end to end delay without Silence Suppression Average Packet End to End Delay using Silence Suppression 0.12 0.1 (sec) 0.08 0.06 Delay 0.04 0.02 0 G729 G711 G723.1 G723.1 G726 G726 G726 G726 G728 G728 5.3K 6.3K 16K 24K 32K 40K 12.8K 16K Voice codecs FIGURE 4.11. Average packet end to end delay using Silence Suppression 4.7.1.3 Average MOS Figure 4.12 shows the Mean Opinion Score (MOS) obtained by the voice codecs in a WLAN network without using silence suppression. It shows that the voice quality produced by all voice codecs except G 723.1 is unacceptable as they have scored a MOS value of 1 which implies unacceptable quality of voice [5]. Hence, silence suppression has to be introduced and the improved results due to silence suppression are shown in figure 4.13. With introduction of silence suppression, the MOS value shows a considerable rise for all voice codecs except G 723.1. For most of the voice codecs the MOS value has exceeded 3.5 indicating very good quality of voice. Hence to finally select a voice codec in a static WLAN 50
  60. 60. WWW.ThesisScientist.com network accommodating large number of users, the voice codec G 726 is expected to perform the best. Average MOS without using silence suppression 3 MOS 2.5 2 Average 1.5 1 0.5 0 G729 G711 G723.1 G723.1 G726 G726 G726 G726 G728 G728 5.3K 6.3K 16K 24K 32K 40K 12.8K 16K Voice Codecs FIGURE 4.11. Average MOS without Silence Suppression Average MOS using silence suppression 4 MOS 3.5 3 2.5 Average 2 1.5 1 0.5 0 G729 G711 G723.1 G723.1 G726 G726 G726 G726 G728 G728 5.3K 6.3K 16K 24K 32K 40K 12.8K 16K Voice codecs FIGURE 4.12. Average MOS using Silence Suppression 4.7.2 Scenario 2 – Mobile Nodes: Figure 4.13 shows the simulation setup used for roaming enabled WLAN network. Using the Wireless Deployment Wizard of OPNET a 7 celled WLAN network, with multiple subscriber stations in the range of a base station is deployed. The access points are connected to the core network by a server backbone via an IP backbone. The server backbone is further connected to the voice server which is configured as the SIP server. The base station, IP Backbone, Server Backbone and the Voice Server together represent the service provider company network. The cell radius is set to be 100 m. The mobile nodes are configured to move at a rate of 50 km/hr, 100 km/hr and 150 km/hr along the path shown in white lines. The number of subscribers in all cells is 10. Voice over IP calls are setup between nodes of cells 2, 3 and 7 using SIP and are shown by the blue lines in the figure 4.13. 51
  61. 61. WWW.ThesisScientist.com FIGURE 4.13. Network Model for WLAN 4.7.2.1 Average Jitter Figure 4.14 shows the variation of jitter across the mobility supported WLAN network with respect to the various voice codecs considered. From the figure it is observed that the average jitter irrespective of speed decreases with the decrease in data rate. G 711 has the highest data rate of 64Kbps and has highest jitter. G 726 has a data rate of 32Kbps as considered and shows jitter less than G 711. G 728 and G 729 has data rates of 16Kbps and 8Kbps and thereby shows the decrease in the jitter and finally G 723.1 which has a minimum data rate of 5.3Kbps and has minimum jitter. As the data rate decreases the load of the network decreases. As the load decreases, the router processing happens in time and the packets arrive in time. This thereby decreases the jitter. Average Jitter 0.35 0.3 Second s 0.25 50km/hr 0.2 150km/hr 100km/hr 0.15 0.1 0.05 0 G 711 G 723 G 726 G 728 G729 Voice Codecs FIGURE 4.14. Average Jitter 4.7.2.2 Average Packet End to End Delay As mobility is introduced in the network, the network performance degrades to an unacceptable level. As the mobile nodes start moving the packet end-to-end delay increases in the order of 10 seconds. This shows that due to low bandwidth a WLAN network is unable to support voice communication at vehicular speed. Overall we can say that such huge delay makes WLAN incapable of supporting voice communication. 52
  62. 62. WWW.ThesisScientist.com Average Packet End to End Delay 180 160 140 Sec ond s 120 50km/hr60 150km / hr 100 100km/hr 80 40 20 0 G 711 G 723 G 726 G 728 G729 Voice Codecs Fig 4.15 Average Packet end to end delay 4.8 Conclusion The target of this work is to observe the variation of jitter and packet end-to- end delay in a WLAN network under various mobility scenarios. This work shows the variation of the mentioned parameters with respect to a stationary network having all stationary nodes and with respect to a mobile network with varying speed of 50km/hr, 100km/hr and 150km/hr. Unlike the WiMAX network average packet end to end delay is less in case of stationary nodes than mobile nodes. Figures 4.8, 4.9 and 4.14 show the variation of jitter in the above mentioned networks. As figure 4.8 and 4.9 shows, the jitter in a stationary network decreases considerably when silence suppression is activated in the network and it increases quite a lot when mobility is introduced in the network. This is because as a node becomes mobile it undergoes handover. This creates inconsistency in the order of the packets delivered thereby jitter. Figures 4.10, 4.11 and 4.15 show the variation of packet end to end delay in the above mentioned WLAN network. From figures 4.10 and 4.11, it is observed that the packet end to end delay decreases in the overall network when silence suppression is activated. This is due to the reduction in the queuing delay. As the number of packets decreases with silence suppression, the number of packets queued in the intermediate nodes decreases. This reduces the queuing delay. The packet end to end delay increases largely when mobility is introduced in the network. With mobility, the mobile nodes move at a very high speed over the network and while getting handed over from one AP to another it moves to a third AP. This results in the fact that the speed of handover is less compared to the speed of the mobile node. Hence, the network fails and the effect is visible to us in terms of huge delay. With a step towards the 4G mobile networks, this study is used in the following chapters to compare and evaluate the performance of the loosely coupled WiMAX WLAN integrated network. 53
  63. 63. WWW.ThesisScientist.com Chapter 5 VoIP over WLAN-WiMAX Integrated Network Chapter Outline:-  Introduction   WLAN vs. WiMAX   Types of Integration Loose Coupling Tight Coupling   Working of the Integrated Network Scenario 1: Stationary Nodes Scenario 2: Mobile Nodes  Experimental Setup  Scenario 1: Simulation Setup and Results Average Jitter  Average Packet End-to-End Delay Average MOS  Scenario 2: Simulation Setup and results Average jitter  Average Packet End-to-End Delay   Conclusion 54
  64. 64. WWW.ThesisScientist.com VoIP over WLAN-WiMAX Integrated Network 5.1 Introduction With growing demand for anytime anywhere connectivity, wireless networks are gaining popularity and have become an integral part of human life. Researchers of academia and industry are searching for new innovative methods to cater to the requirement of the variety of roaming users. Nowadays the requirement is not only restricted to connectivity but quality of service is also added to the requirement. With voice over IP coming into existence this demand has gained more importance. By distributing high-speed internet access from cable, digital subscriber line (DSL), and other fixed broadband connections within wireless hotspots, WLAN has dramatically increased productivity and convenience. The integration of WLAN into laptops has accelerated the adoption of WLAN and an increased number of handhelds and Consumer Electronics (CE) devices are thereby adding WLAN capabilities. Moreover as shown in Chapter 4, the quality of service for voice application is of acceptable quality in WLAN network. WiMAX on the other hand has taken wireless Internet access to the next level, and over time, could achieve similar attach rates to devices as WLAN. WiMAX can deliver Internet access miles from the nearest WLAN hotspot and blanket large areas -Wide Area Networks (WANs), be they metropolitan, suburban, or rural - with multi-megabit per second mobile broadband Internet access. In the last few years, WiMAX has established its relevance as an alternative to wired DSL and cable, providing a competitive broadband service offering that can be rapidly and cost effectively deployed. Now, Mobile WiMAX, as defined in the Institute of Electrical and Electronic Engineers (IEEE) 802.16e-2005 standard, adds broadband connectivity on the move. Mobile WiMAX, based on scalable Orthogonal Frequency Division Multiple Access (OFDMA) technology, is capable of simultaneously supporting fixed, portable, and mobile usage models. Together, WiMAX and WLAN are ideal partners for service providers to deliver convenient, affordable mobile broadband Internet services in more places. 55
  65. 65. WWW.ThesisScientist.com Both are open IEEE wireless standards built from the ground up for Internet Protocol (IP)-based applications and services. By using WiMAX backbone and WLAN hotspot, service providers can deliver high-speed Internet connectivity that subscriber‟s desire in more places. And WiMAX and WLAN technology synergies enable seamless integration into laptops, CE devices, and generate a new category of devices called “mobile Internet devices”. These devices are expected to have dual stack for both WLAN and WiMAX. 5.2 WLAN vs. WiMAX Although both WiMAX and WLAN provide wireless broadband connectivity, they have been optimized for different usage models: WLAN for very high-speed WLAN connectivity and WiMAX for high-speed Wireless WAN (WWAN) connectivity. By combining WiMAX and WLAN technologies, service providers can offer their subscribers a more complete suite of broadband services in more places. Table 5.1 illustrates how WiMAX and WLAN complement each other from an implementation and deployment perspective and also shows the impact of the WLAN WiMAX integration synergy. TABLE 5.1: WLAN and WiMAX Comparison [42] [43] WiMAX 802.16e WLAN Synergy Impact 802.11b Primary Deployed in wide Deployed in “Best-connected” Application coverage areas, local coverage model: users connect including areas, such as to WiMAX or WLAN metropolitan areas public depending on their for mobile hotspots, location, coverage, broadband homes and and Quality of Service wireless as well as businesses. (QoS) requirements. rural or remote areas for last-mile connectivity and portable service Frequency Licensed/ 2.4 GHz ISM Service providers can Band Unlicensed 2-11 Band leverage both types of 56

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