Analysis of Secure Real Time Transport Protocol on Voip over
Mohd Nazri Ismail / (IJCSE) International Journal on Computer Science and Engineering Vol. 02, No. 03, 2010, 898-902
Analysis of Secure Real Time Transport Protocol on VoIP over Wireless LAN in Campus Environment
Mohd Nazri Ismail
Department of MIIT, University of Kuala Lumpur (UniKL), MALAYSIA mnazrii@miit.unikl.edu.my
Abstract- In this research, we propose to implement Secure Real Time Transport Protocol (SRTP) on VoIP services in campus environment. Today, the deployment of VoIP in campus environment over wireless local area network (WLAN) is not considered on security during communication between two parties. Therefore, this study is to analyzed SRTP performance on different VoIP codec selection over wired. We have implemented a real VoIP network in University of Kuala Lumpur (UniKL), Malaysia. We use softphone as our medium communication between two parties in campus environment. The results show that implementation of SRTP is able to improve the VoIP quality between one-to-one conversation and multi conference call (many-to-many). In our experiment, it shows that iLBC, SPEEX and GSM codec are able to improve significantly the multi conference (many-to-many) VoIP quality during conversation. In additional, implementation of SRTP on G.711 and G.726 codec will decrease the multi conference (many-to-many) VoIP quality. Keywords- Codecs, Softphone, SRTP, WLAN
I. INTRODUCTION AND RELATED WORKS University of Kuala Lumpur (UniKL) has implemented a real VoIP over wireless LAN in campus environment. This implementation is not covered any security features. Therefore, the objective of this study is to enable the security function using Secure Real Time Transport Protocol (SRTP). We will study the performance of SRTP on different codec such as G.711, G.726, GSM, iLBC and SPEEX. iLBC is a speech codec developed for robust voice communication over IP, it uses 13.33 Kbps. It provides low delay and high packet loss robustness for low-bit rate codec’s. SPEEX codec is
Analysis of Secure Real Time Transport Protocol on VoIP over Wireless LAN in Campus Environment
Mohd Nazri Ismail
Department of MIIT, University of Kuala Lumpur (UniKL), MALAYSIA mnazrii@miit.unikl.edu.my
Abstract- In this research, we propose to implement Secure Real Time Transport Protocol (SRTP) on VoIP services in campus environment. Today, the deployment of VoIP in campus environment over wireless local area network (WLAN) is not considered on security during communication between two parties. Therefore, this study is to analyzed SRTP performance on different VoIP codec selection over wired. We have implemented a real VoIP network in University of Kuala Lumpur (UniKL), Malaysia. We use softphone as our medium communication between two parties in campus environment. The results show that implementation of SRTP is able to improve the VoIP quality between one-to-one conversation and multi conference call (many-to-many). In our experiment, it shows that iLBC, SPEEX and GSM codec are able to improve significantly the multi conference (many-to-many) VoIP quality during conversation. In additional, implementation of SRTP on G.711 and G.726 codec will decrease the multi conference (many-to-many) VoIP quality. Keywords- Codecs, Softphone, SRTP, WLAN
I. INTRODUCTION AND RELATED WORKS University of Kuala Lumpur (UniKL) has implemented a real VoIP over wireless LAN in campus environment. This implementation is not covered any security features. Therefore, the objective of this study is to enable the security function using Secure Real Time Transport Protocol (SRTP). We will study the performance of SRTP on different codec such as G.711, G.726, GSM, iLBC and SPEEX. iLBC is a speech codec developed for robust voice communication over IP, it uses 13.33 Kbps. It provides low delay and high packet loss robustness for low-bit rate codec’s. SPEEX codec is
References: [1]. Moura N.T.; Vianna B.A.; Albuquergue C.V.N; Rebello V.E.F & Boeres C. “MOS-Based Rate Adaption for VoIP Sources”. IEEE International Conference on Communication, pp. 628-633, 2007. [2]. Masuda M. & Ori K. “Delay Variation Metrics for Speech Quality Estimation of VoIP”. Institute of Electronics, Information and Communication Engineers (IEIC) Technical Report, Vol. 101(11), pp. 101-106, 2001. [3]. R.G. Cole & J.H. Rosenbluth. “Voice over IP Performance Monitoring”. SIGCOMM Computer Communication Rev. Vol. 31(2), pp. 9-24, 2001. [4]. L. Ding & R. Goubran. “Speech Quality Prediction in VoIP Using the Extended e-Model. Global Telecommunication Conference, GLOBECOM ’03. IEEE, Vol. 7, pp. 3974-3978, 2003. [5]. Alexandre P.; Edjair M.; & Edjard M. “Analysis of the Secure RTP Protocol on Voice over Wireless Networks using Extended MedQoS”. Proceedings of the 2009 ACM symposium on Applied Computing, pp. 86 – 87, 2009. [6]. M. Baugher, D. McGrew, M. Naslund, E. Carrara, & K. Norrman. “The Secure Real- Time Transport Protocol (SRTP)”. RFC 3711 (Proposed Standard), March 2004. [7] Douglas C. Sicker & Tom L. “VoIP Security: Not an Afterthought”, FEATURE: Q focus: Voice Over IP, Vol. 2(6), pp. 56-64, 2004. [8] Vesselin I., Theodor T., & Amdt T. “Experiences in VoIP telephone network security policy at the University of Applied Sciences (FHTW) Berlin”, Proceedings of the 2007 international conference on Computer systems and technologies, Bulgaria, Vol. 285(3), 2007. [9] Wafaa B. D., Samir T., & Carole B. “Critical vpn security analysis and new approach for securing voip communications over vpn networks”, Proceedings of the 3rd ACM workshop on Wireless multimedia networking and performance modelling,Chania, Crete Island, Greece, pp. 92-96, 2007. [10] Nekita A. C., & Chhabria S. A. “Multiple design patterns for voice over IP security”, Proceedings of the International Conference on Advances in Computing, Communication and Control, Mumbai, India, pp. 530 – 534, 2009. Figure 3.8: VoIP Conversation over Multi Conference Call over WLAN Figure 3.9: VoIP Conversation over One-to-One Call over WLAN IV. CONCLUSION AND FUTURE WORK Based on the results, implementation of SRTP using GSM, iLBC and SPEEX codecs are able to generate high quality of VoIP conversation WLAN for one-to-one conversation and multi conference call (many-to-many). After implemented SRTP for multi conference call (many-to-many), the MOS result indicates that G.711 and G.726 codec will decrease the performance of VoIP conversation over WLAN. Overall of our finding, it confirms that enable SRTP will improve and increase the quality of one-to-one VoIP conversation and VoIP over multi conference call (only for iLBC, GSM and SPEEX codecs). Since the manual/human MOS tests are quite subjective and less than productive in many ways, there are nowadays a number of software tools that carry out automated MOS testing in a VoIP deployment. Although they lack the human touch, the good thing with these tests is that they take into account all the network ISSN : 0975-3397 902 Copyright of International Journal on Computer Science & Engineering is the property of Engg Journals Publications and its content may not be copied or emailed to multiple sites or posted to a listserv without the copyright holder 's express written permission. However, users may print, download, or email articles for individual use.