ENSC 894: COMMUNICATION NETWORKS. Performance Analysis of VoIP Codecs over WiMAX and Wi-Fi Networks. Final Project- Spring 2012.

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1 ENSC 894: COMMUNICATION NETWORKS Performance Analysis of VoIP Codecs over WiMAX and Wi-Fi Networks Final Project- Spring 2012 Khaled Alutaibi Student ID, Address, Web Page: kalutaib/project.html May 17, 2012

2 Abstract Voice over IP (VoIP) applications such as Skype, Google Talk, and FaceTime are promising technologies for providing low cost voice calls to customers over existing data networks. Wireless networks such as WiMAX and Wi-Fi focus on providing Quality of Service (QoS) for VoIP applications. However, there are numerous aspects that affect quality of voice connections over wireless networks. In this paper, we evaluate performance of three VoIP codecs over WiMAX and Wi-Fi networks. OPNET WiMAX and Wi-Fi simulation models are designed. Performance metrics such as Mean Opinion Score (MOS), average end-to-end delay, and jitter are evaluated and discussed.

3 Contents 1 Introduction Overview VoIP over Wi-Fi VoIP over WiMAX Project Motivation Related Work QoS Issue of VoIP Application VoIP Codecs G G G OPNET Implementation Simulations Scenarios Assumption and Configuration Results and Analysis VoIP over Wi-Fi VoIP over WiMAX Conclusion and Future Work Bibliography 11 i

4 List of Figures 2.1 Networks Topology VoIP Traffic Distribution OPNET Configuration Average MOS (Wi-Fi) Average end-to-end delay (Wi-Fi) Average jetter (Wi-Fi) Average MOS (WiMAX) Average end-to-end delay (WiMAX) Average jetter (WiMAX) ii

5 List of Tables 1.1 Mean Opinion Score (MOS) [1] Guideline for the Voice Quality Measurement [2] Common VoIP Codec Simulations Scenarios iii

6 1 Introduction 1.1 Overview The recent Voice over IP (VoIP) applications such as Skype, Google Talk, and FaceTime have changed the way people communicate to each other. Due to the low cost, people find VoIP as an alternative to the expensive traditional Public Switched Telephone Network (PSTN). VoIP has set of parameters that defined its Quality of Service (QoS) such as end to end delay, jitter, packet loss, Mean Opinion Score (MOS), and throughput[3]. The existing wireless networks such as WiMAX and Wi-Fi offer flexibility to support such applications [3]. At the time the IEEE (Wi-Fi) technology showed great success as cheap wireless internet access, the IEEE (WiMAX) provides large coverage (around 50 kilometers) and high data rates (up to 100 Mbps) using radio link [4]. The aim of this project is to examine the QoS needs for VoIP applications in both technologies. OPNET 16.0 simulator is used to analyze the QoS of VoIP application under different codecs. This report is organized as follows: section 1 gives an overview about VoIP over Wi- Fi and WiMAX networks and introduces some related studies in this area of research. Section 2 explains the simulation scenarios. Section 3 presents the simulation results and graphs along with an explanation of the results. Finally, the conclusion and some suggested future work are in section VoIP over Wi-Fi Wi-Fi is commonly used in residential, business, and public areas. It is notable that the perceived throughput in Wi-Fi does not match the real throughput [4]. Also, all users share the access to the channel which is very critical for all real time traffic in general and especially VoIP. The low capacity in Wi-Fi connections has a high impact on the QoS in VoIP [4]. Beside the high traffic generated by users, both protocols, VoIP and Wi-Fi, create large headers which result in high drawback on VoIP performance [4] VoIP over WiMAX WiMAX as a broadband wireless technology is considered as alternative solution to wired networks. It provides up to 100 Mbps data rate and has a coverage area up to 50 Km [4]. Also, it supports QoS requirements by different applications especially real time applications such as VoIP applications. WiMAX has four different traffic classes to support different application needs: 1

7 Best Effort (BE): is designed for applications that do not require QoS such as web browsing [4]. Non Real-Time Polling service (nrtps): supports non real-time applications that require variable size data such as File Transport Protocol (FTP) [4]. Unsolicited grant service (UGS): supports Constant Bit Rate (CBR) application where BS assigns a fixed bandwidth to users such as VoIP without silence suppression [4], [5]. Real-time polling service (rtps): supports real-time applications with variable size data where BS allocates bandwidth beased on SS request such as MPEG [5]. Although WiMAX has been designed to provide Broadband Internet service, VoIP applications are considered as one of its killer applications [6]. 1.2 Project Motivation VoIP applications are real time applications which use UDP for end to end transmission. In VoIP, there are several factors that affect the quality of the voice calls. The choice of codec is a significant factor in all VoIP applications. In this project, three codecs, G.711, G.723, and G. 729, are used to analyze the QoS of VoIP over two types of wireless networks, Wi-Fi and WiMAX. 1.3 Related Work There are several studies addressed the performance issues related to VoIP applications over different technologies. In Qureshi et al. [4], the authors studied and compared VoIP over Wi-Fi and WiMax. Jitter, packet losses, throughput and delay are analyzed using ns2 tool. In [3], integrated WiMAX/Wi-Fi network was simulated via OPNET simulator to test VoIP under different codecs by Islam et al. [3]. In this study, the performance of WiMAX/Wi-Fi network is investigated to choose the appropriate VoIP codec. In Jadhav et al. [7], a simulation study for both WiMAX and UMTS are done in OPNET. They analyzed several performance metrics to find the better technology to support VoIP applications. This study showed that WiMAX is better than UMTS to support VoIP. Another study by Tariq et al. [8], evaluated the performance of different VoIP codecs over the WiMAX network. QualNet simulator is used to verify that varying the jitter buffer size and packetization time affects the quality of VoIP over WiMAX network. Adhicandra examined in [6] the performance of WiMAX network under two different QoS configurations. It tested the capability of a WiMAX network to improve voice and data applications performance. 2

8 1.4 QoS Issue of VoIP Application Nowadays, people get advantage of the existing data networks by enjoying various ways of communication e.g. text messages, voice calls, and video calls. The traditional phone networks can not compete with these type of services due to low equipments and operating cost, and the ability of integrating voice and data in applications [3]. The QoS for VoIP can be measured by evaluating three performance metrics: Mean Opinion Score (MOS), Jitter, and end-to-end delay. Mean Opinion Score (MOS): MOS is a scale from 1 to 5 which measures the quality of the voice. The value of worst quality is 1 and the best quality is 5 [7], Table 1.1. Table 1.1: Mean Opinion Score (MOS) [1]. Quality Scale Score Listening effort Scale Excellent 5 No effort required Good 4 No appreciable effort required Fair 3 Moderate effort required Poor 2 Considerable effort required Bad 1 No meaning understood with reasonable effort Jitter: The variation in arrival time of consecutive packets is called jitter [2]. Before decoding, packets arrive to limited size buffer however some packets may lost or arrive out of order. Jitter can be calculated by computing the difference delay of packets over a period of time [7]. Packet end-to-end delay: The end-to-end delay can be measured by calculating the delay from the speaker to the receiver. This includes network delay, encoding and decoding delay, and compression and decompression delay [2]. Table 1.2 shows the guidelines for voice quality measurement for both jitter and end-toend delay as it is provided by ITU Telecommunication Standardization Sector (ITU-T) [2]. Table 1.2: Guideline for the Voice Quality Measurement [2]. Network Parameter Good Acceptable Poor Delay(ms) >300 Jitter(ms) >50 3

9 1.5 VoIP Codecs VoIP has several codecs which is used to compress and decompress audio samples. Each codec applies different algorithms. Table 1.3 shows some of the most popular codecs [1]. This project evaluates three VoIP codecs which are:g. 711, G. 723, and G Table 1.3: Common VoIP Codec. Codec Data Rate (kbps) MOS Score G G G G G GSM G. 711 G. 711 is a free codec which is used widely in VoIP applications. It was introduced in 1972 by International Telecommunications Union (ITU) [9]. It uses a logarithmic compression which compress each 16-bit sample to 8-bits. As a result of this, its bit-rate is 64 kbps which is considered as the highest bit-rate among the other codecs. G. 711 has very good audio quality and has MOS value of 4.2 [9] G. 723 G. 723 is a licensed codec which is designed for calls over modems links with data-rates 28.8 and 33 kbps. Therefore, it has two versions with different bit-rates: 5.3 and 6.4 kbps [9]. In this project, we are testing the 5.3 kbps which is based on Algebraic code-excited linear prediction (ACELP) algorithm. The MOS value this code 3.7 [10] G. 729 G. 729 is also a licensed codec which is designed to provide good call quality without consuming high bandwidth [9]. It is built on Conjugate-Structure Algebraic-Code-Excited Linear Prediction (CS-ACELP) algorithm with bit-rate is 8 kbps and MOS value is 4.0 [9],[10]. 4

10 2 OPNET Implementation 2.1 Simulations Scenarios In this project, two OPNET models, Wi-Fi and WiMAX, are designed to evaluate the QoS in VoIP. Each model simulates a network that consists of two mobile subnets: Vancouver subnet and Calgary subnet. Both subnets are connected via IP cloud. This IP cloud is connected to VoIP application server. All the links have 10-20% background. Figure 2.1 shows both wireless networks. (a) Wi-Fi Netwok Model (b) WiMAX Network Model Figure 2.1: Networks Topology Each subnet has five subscribers who communicate via VoIP to other subscribers in the network randomly. Each model is tested under three different scenarios. Each scenario is configured to use one of the voice codecs: G. 711, G. 723, and G Table 2.4 shows the six scenarios that are used in this project. Table 2.4: Simulations Scenarios Scenario Scenario Name Codec Number of Clients 1 VoIP over WiMAX G VoIP over WiMAX G VoIP over WiMAX G VoIP over Wi-Fi G VoIP over Wi-Fi G VoIP over Wi-Fi G Assumption and Configuration In this project, OPNET Modeler version 16.0 is used [7]. Two models for both WiMAX and Wi-Fi are designed as shown in

11 The average duration of each call is three minutes [11]. After that, another call is generated to another clients, Figure 2.2. All calls inter arrival-time follow exponential distribution [2]. Each model has three scenarios to examine three different codecs: G711, G723, and G729. The period of each scenario simulation is 60 minutes. Figure 2.3a and 2.3b show the OPNET configuration for both Wi-Fi and WiMAX networks, respectively. Figure 2.2: VoIP Traffic Distribution (a) Wi-Fi Configuration (b) WiMAX Configuration Figure 2.3: OPNET Configuration 6

12 3 Results and Analysis 3.1 VoIP over Wi-Fi This section shows the results of the three scenarios of VoIP over Wi-Fi. Figure 3.4 shows that G. 711 has the highest average MOS value at However, G. 723 and G. 729 have acceptable MOS values between 3.95 to 4.0. Figure 3.4: Average MOS (Wi-Fi) Interestingly, although G. 711 has the highest data-rate in this experiment, it shows the lowest average end-to-end delay as shown in Figure 3.5. On the other hand, G. 723 and G. 729 achieve more than 400 ms as average end-to-end delay which is considered as poor voice connection. Figure 3.6 shows that the jitter slightly is experienced under G. 729 codec. In general, we can say G. 711 codec shows the best codec for VoIP applications over Wi-Fi networks. 7

13 Figure 3.5: Average end-to-end delay (Wi-Fi) Figure 3.6: Average jetter (Wi-Fi) 3.2 VoIP over WiMAX In this section, the performance of VoIP over WiMAX is tested using G.711, G. 723, and G. 729 codecs. Figure 3.7 shows the average MOS value for the three codecs. G. 711 has 4.35 as the best value among the other codecs. However, G. 723 and G. 729 achieve 3.9 and 4.0, respectively. These values are accepted for both schemes as shown in Table

14 Figure 3.7: Average MOS (WiMAX) Figure 3.8: Average end-to-end delay (WiMAX) Figure 3.8 shows all codecs with average end-to-end delay less than 140 ms which is in the range of good voice connection. Also, all three codecs experience no jitter on average as shown in Figure 3.9. We can conclude from these graphs that G. 711, G. 723, and G.729 are all suitable to be used in VoIP application over WiMAX. 9

15 Figure 3.9: Average jetter (WiMAX) 4 Conclusion and Future Work In this project, we evaluated the performance of three different VoIP codecs over Wi-Fi and WiMAX networks. The VoIP performance is measured in six scenarios using OPNET MOS, jitter, and average end-to-end delay are analyzed as performance parameters which define QoS in VoIP. G. 711 codec showed the better codec for VoIP over Wi-Fi. However, all three codecs, G. 711, G. 723, and G. 729, showed acceptable performance quality in VoIP over WiMAX. This project considered voice calls from fixed nodes. The impact of mobility on VoIP performance is suggested as future research. Also, other VoIP codecs such as G. 722, G. 726, and G.728 can be evaluated on both Wi-Fi and WiMAX networks. 10

16 References [1] R. S. Ramakrishnan and P. V. kumar, Performance analysis of different codecs in voip using sip, in Mobile and Pervasive Computing (CoMPC2008), August [2] J. Yu and I. Al-Ajarmeh, Call admission control and traffic engineering of voip, in Digital Telecommunications, ICDT 07. Second International Conference on, p. 11, july [3] S. Islam, M. Rashid, and M. Tarique, Article: Performance analysis of WiMax/WiFi system under different codecs, International Journal of Computer Applications, vol. 18, pp , March Published by Foundation of Computer Science. [4] M. A. Qureshi, A. Younus, M. Saeed, F. A. Sidiqui, N. Touheed, and M. S. Qureshi, Comparative study of voip over wimax and wifi, International Journal of Computer Science, vol. 8, pp , May [5] E. Haghani and N. Ansari, Voip traffic scheduling in wimax networks, in Global Telecommunications Conference, IEEE GLOBECOM IEEE, pp. 1 5, dec [6] I. Adhicandra, Measuring data and voip traffic in wimax networks, CoRR, vol. abs/ , [7] S. Jadhav, H. Zhang, and Z. Huang, Performance evaluation of quality of voip in wimax and umts., in PDCAT, pp , IEEE Computer Society, [8] M. I. Tariq, M. A. Azad, R. Beuran, and Y. Shinoda, Performance analysis of voip codecs over be wimax network, 3rd International Conference on Computer and Electrical Engineering (ICCEE 2010), pp , November [9] V. Toncar, Voip basics: Overview of audio codecs, April Available: Basics Overview of Audio Codecs.html. [10] Voip codecs, April Available: [11] A. M. Noll, Cybernetwork technology: issues and uncertainties, Commun. ACM, vol. 39, pp , dec

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