SIP and RSW: A Comparative Evaluation Study

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(IJCSIS) International Journal of Computer Science and Information Security, Vol. 8, No. 8, 2010

SIP and RSW: A Comparative Evaluation Study
Mahmoud Baklizi1, Nibras Abdullah1, Omar Abouabdalla1, Sima Ahmadpour1. 1: National Advanced IPv6 Centre of Excellence 1: Universiti Sains Malaysia 1: Penang, Malaysia 1: {mbaklizi, abdullahfaqera, omar, sima}@nav6.org

Abstract— Voice over internet protocol (VoIP) is a technology that uses Internet to transmit voice digital information. The Session Initiation Protocol (SIP) and Real time Switching (RSW) are signaling protocols that emerged as a new VoIP which gained popularity among VoIP products. In literature, many comparative studies have been conducted to evaluate signaling protocols, but none of them addressed the targeted protocols. In this paper, we make a comparative evaluation and analysis for SIP and RSW using Mean Opinion Score rating (MOS). We found that RSW performs better than SIP under different networks in terms of (packet delays). Keywords- VoIP; MOS; InterAsterisk eXchange Protocol; Real Time Switching Control Criteria and Session Initiation protocol.

oriented programming languages such as Java and Perl. These allow easy debugging, and most importantly make SIP flexible and extensible. (ii) Less signaling. (iii) transport-layerprotocol neutral (iv) parallel search [5][7]. Real time Switching appeared in late 1993 as a control mechanism for multimedia conferencing. It was designed by Network Research Group (NRG) in school of computer science-University Science Malaysia (USM) .The goal of RSW Control Criteria is how to conduct a conference around a meeting table [8][9][10]. Moreover, RSW is used to handle two issues in multimedia conferencing. The first one is to handle the confusion generated while everyone tries to speak at the same time. The second issue is the tremendous amount of network traffic generated by all participating sites [6]. More recently, InterAsterisk exchange protocol has been emerged as new protocol that improved the voice quality. It has also many features such as simplicity, NAT-friendliness, efficiency and robustness [2][3]. There are several goals for this protocol. The main goals of this protocol are (i) Minimizing bandwidth usage for signaling and media transfer, with a particular emphasis on voice (ii) Ensuring NAT transparency (iii) Ability to exchange dial plans (iv) Efficient implementation of intercom and paging features. On the other hand, IAX can be used with different types of streaming media such as video and voice calls [4]. In this paper, we have selected two case studies that depend on IAX protocol. In the first case study, the researchers made comparison between IAX and RSW [3]. While in the second case study the researchers compared between IAX and SIP [2]. According to the last two comparisons, we have made a comparative evaluation of SIP and RSW by using Mean Opinion Score rating (MOS). II. RELATED WORK

I.

INTRODUCTION

Nowadays, the use of distributed computer network systems has been increased in many areas of government, academia and industry. Video conferencing system is one of computerbased communication applications. The idea of video conferencing appeared for the first time in the 1920s [1]. The task of video conferencing concentrates on individuals to be together in space and time, and makes groups more effective at their work by applying different services such as telephony service over IP networks that are known as IP telephony or Voice over IP. VoIP communication usually consists of two protocols: (i) Signaling protocols that are used to setup a voice conversation and manage voice sessions (ii) Media transfer protocols that are used for exchanging voice data traffic during one session lifetime. [2]. One of the most important functions in the VoIP infrastructure is signaling session. Signaling session should be by any VoIP protocol before transmitting any media type. Therefore, it allows various network components to communicate with each other to setup and to tear down calls [3]. Recently, there is strong focus on the development of scalable VoIP protocols, such as SIP, RSW and IAX. Large efforts have been done to study SIP. It was standardized in the IETF (Internet Engineering Task Force) RFC 2543 and further extended in RFC 3261.It is used for creating, modifying, and terminating sessions with one or more participants [5][6]. Session Initiation Protocol has many features: (i) the service of text-based which allows easy implementation in object

Before transmitting any media, the signaling session, which is the most important function in VoIP, allows different network components to communicate between each other to set up and terminate calls. Mean Opinion Score (MOS) is used as a method to assess call quality. In addition, it is used to measure subjective voice quality. To measure network performance, we use the MOS rating which is the most widely used assessment technique. The listening subjects were given a scale from 1 to

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(IJCSIS) International Journal of Computer Science and Information Security, Vol. 8, No. 8, 2010

5, where 1 = bad, 2 = poor, 3 = fair, 4 = good, and 5 = excellent [11]. Recently, there are many researchers concentrated on this function and compared between different VoIP protocols. The study conducted by [2] compared the performance between IAX and SIP protocols. They have indicated that both protocols perform comparably in the presence of fixed delay. Therefore IAX appeared to perform slightly better as shown in Figure 1 [2].

that the packet delay in IAX is better than that in SIP and RSW. The gained results are extracted using the SPSS statistical tool. The Interclass Correlation Coefficient statistical test is used to analyze IAX with comparison to the previous two studies. As a result, the correlation between the two variables is high. The single measure Interclass correlation coefficient is 0.995, the test value for absolute agreement is 0.99999, and p is approximately equal to 1. The IAX protocol regression formula and curve were identical in previous two studies. An indirect comparison between RSW and SIP was made directly in terms of fixed packet delay. Figure 3 indicates that the RSW protocol performs slightly better in the presence of fixed packet delay than SIP protocol.

Figure 1: Packet Delay - SIP versus IAX [2]

The researchers in [3] compared the performance between IAX and RSW protocols. They indicated that both protocols perform comparably in the presence of fixed delay. Therefore IAX appeared to perform slightly better than RSW as shown in Figure 2 [3].

Figure 3: Packet Delay - SIP versus RSW

CONCLUSION This paper suited the performance of the RSW and SIP in term of packet delay. From the evaluation and analysis, we conclude that RSW achieves more improvement than SIP in VoIP in relation to packet delay via the MOS score. Further research on RSW control criteria and SIP protocols and their performance under various conditions is recommended. Acknowledgment We would like to thank Hani Mimi and Abdullah Dahbali for their support and guidance throughout the study and analysis.

REFERENCES
Figure 2: Packet Delay - RSW versus IAX [3] [1] [2] E. M. Schooler, “Conferencing and collaborative computing,” Multimedia Systems. Vol. 4, pp. 210-225, 1996 T. Abbasi, S. Prasad, N. Seddigh, and I. Lambadaris, "A comparative study of the SIP and IAX VoIP protocols," Electrical and Computer Engineering, Canadian Conference, pp.179-183, 1-4 May 2005 S. Manjur, M. Mosleh, O. Abouabdalla, W. Tat , and A. Manasrah, “Comparative Evaluation and Analysis of IAX and RSW,”(IJCSIS).Vol. 6,2009

III.

ANALYSIS AND EVALUATION
[3]

The predicated evaluation is based on previous two comparisons that were conducted by [2] and [3]. We can see

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(IJCSIS) International Journal of Computer Science and Information Security, Vol. 8, No. 8, 2010
M. Spencer, F. Miller, “IAX Protocol Description,” February 2005, http://www.cornfed.com/iax.pdf.  [5] I. Dalgic and H. Fang, “Comparison of H.323 and SIP for IP telephony signaling,” in Proc. of Photonics East, (Boston, Massachusetts), SPIE, Sept. 1999. [6] O. Abouabdalla, R. Sureswaran, "Enable Communications between The RSW Control Criteria and SIP Using R2SP," Distributed Frameworks for Multimedia Applications, 2006. The 2nd International Conference on, vol., no., pp.1-7, May 2006. [7] Y. Zhang, “SIP-based VoIP network and its interworking with the PSTN,” Electronics & Communication Engineering Journal, December 2002, Volume 16, Issue 6, pg 273-282 [8] R. Sureswaran, “A Reflector Based System to Support the RSW Multimedia Conferencing Control Criteria,” IASTED International Conference on Networks, Orlando, January 1996. [9] R. Sureswaran, Subramanian, R.K, H. Guyennet, and M. Trehel, “Using the RSW Control Criteria To Create A Distributed Environment for Multimedia Conferencing,” In Proceedings of REDECs '97. Penang, Malaysia. 27-29 November 1997. [10] R. Sureswaran, “A Distributed Architecture to support Multimedia Applications Over the Internet and Corporate Intranets,” In Proceedings of SEACOMM '98. Penang, Malaysia. 12-14 August 1998. [11] http://www.itu.int/rec/T-REC-P.800/en Mahmoud Khalid Baklizi is a researcher pursuing his PhD in Computer Science at the National Advanced IPv6 Center of Excellence in University Sains Malaysia. He received his first degree in Computer Science from Yarmouk University, Jordan, 2002 and his Master degree in Computer Information System from the Arab Academy for Banking and Financial Sciences, Jordan in 2008. His research area of interest includes Multimedia Networking. Nibras Abdullah Faqera received his Bachelor of Engineering from College of Engineering and Petroleum, Hadhramout University of science and technology, Yemen, 2003. He obtained his Master of Computer Science from School of Computer Science, Universiti Sains Malaysia in 2010. He is academic staff member in Hodeidah University, Yemen. He is researcher pursuing his PhD in Computer Science at the National Advanced IPv6 Center of Excellence in University Sains Malaysia. His research area of interest includes Multimedia Conferencing System (MCS). Dr. Omar Amer Abouabdalla is a senior lecturer and head of the technical department in the National Advanced IPv6 Centre (NAv6) - University Science Malaysia (USM). Dr. Omar is the Chairman of multimedia working group (a sub working groupin APAN), Asia Pacfic Advanced Network (APAN) is a high bandwidth network that will interconnect the Asia Pacifc Countries. He is also a member of Internet Engineering Task Force (IETF) and a member of Editorial Board for Journal of IT in Asia. Dr. Omar is heavily involved in researches carried by NAv6 center, such as Multimedia Conferencing System (MCS) and IPv6 over Fiber project. He has more than five years experience in the field of IPv6 and more than ten years in the field of Multimedia Network. [4]

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