VOL. 10, NO 19, OCTOBER, 2015 ISSN 1819-6608 ARPN Journal of Engineering and Applied Sciences © 2006-2015 Asian Research Publishing Network (ARPN). All rights reserved. www.arpnjournals.com 8912 PERFORMANCE EVALUATION OF VOICE OVER IP USING MULTIPLE AUDIO CODEC SCHEMES L. Audah 1 , A. A. M. Kamal 1 , J. Abdullah 1 , S. A. Hamzah 1 and M. A. A. Razak 2 1 Optical Communication and Network Research Group, Faculty of Electrical and Electronic Engineering, Universiti Tun Hussein Onn Malaysia, Parit Raja, Batu Pahat, Johor, Malaysia 2 Faculty of Electrical Engineering, Universiti Teknologi Malaysia, Johor Bahru, Malaysia E-Mail: shipun@uthm.edu.my ABSTRACT The evolution of Voice over IP (VoIP) has made it one of the most popular applications over the wired/wireless Internet system due to its flexibility in technology integration and low cost of services. Telco and service operators have used the communication resources to optimize the VoIP architecture in order to provide better quality of service (QoS) to end consumers. The VoIP is a delay-sensitive traffic which requires minimum delay for general applications and minimum loss ratio for specific applications as the key QoS performance parameters. This paper compares the end-to-end (e2e) QoS performance parameters of VoIP codec schemes against multiple traffic connections transmitted over the Internet system. Background traffics are included in the simulations to closely match the real-world Internet scenario. Simulations analysis of bidirectional VoIP communications are done from the network layer perspective to compare the QoS performances of G.711, G.729A, G.723.1 and GSM.AMR codec schemes against the incremental of active connections in the network system. The results show that the G.729A produces at least 2.81% better in term of average accumulative e2e delay. The G.711 produces at least 21.89% better in term of average accumulative e2e jitter but produces the worst e2e packet loss ratio. In addition, GSM.AMR shows the best e2e effective transmission rate ratio ranges between 42.67% and 89.82%. This study has investigated the QoS performance variations of VoIP codecs so that the results could be used as guidelines to estimate the optimal network resources for various traffic requirements as early as in the design stage. As for future works, this study suggests the adaptive priority queue and packet scheduling at Internet getaway to regulate the traffic based on per flow QoS requirements. Keywords: VoIP, audio codecs, QoS, internet, simulation, NS-2. INTRODUCTION Voice over IP (VoIP) applications has been very popular among the millions of Internet users for audio communications. The packet based framework of VoIP and low cost of packet data usage have made it the most preferable choice of future Internet communications to replace the conventional circuit switch telephone network. The flexibility of VoIP application to be implemented as software/hardware based and the ability to connect multiple devices simultaneously through wired/wireless channels have become one of the hot topics in the realm of Internet of Things (IoT). Big players from the telco operators and service providers have participated in the billion dollars market opportunities of VoIP applications for future smart-home and smart-cities. In order to optimize the VoIP applications in future communications, various research works have been done from the physical to application layers. Previous research paper by (El-brak et al., 2011) has compared the VoIP performance over small mobile as-hoc network (MANET) using a pair of source and destination nodes. Besides that, (Ashouri et al., 2014) has analyzed the performance of VoIP using different encryption methods over a local wireless network. Moreover, (Kim et al., 2014) has studied the performance of VoIP QoS over long term evolution (LTE) system from the user perspective by varying the speed, distance and number of mobile nodes. In addition, research by (Cocker et al., 2014) has analyzed from the transport layer perspective for the buffer requirements over time and path taken by VoIP traffic over the Internet system. None of the related research has compared the QoS performance of VoIP codec schemes from the network layer perspective and its robustness against various competing traffics. This paper analyzes and compares the e2e QoS performance parameters of bidirectional VoIP (i.e. delay, jitter, loss ratio and throughput) using different types of audio codecs mainly used for VoIP applications (i.e. G.711, G.729A, G.723.1 and GSM.AMR). Simulations analyses have been done from the network layer perspective using multiple connections of VoIP and background traffics to closely match the real-world Internet system scenario. The simulation results provide insight on the performance comparison of each VoIP codec used for audio communication over the Internet system. The remainder of this paper is organized as follows: Section II explains the NS-2 simulation configuration. The results and analysis are discussed on Section III. Finally, Section IV concludes the findings and suggests future research works.