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 firstname.lastname@example.org 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 open source patent-free audio compression format designed for speech. Codec is an algorithm used to encode and decode the voice conversation. Secure Real Time Transport Protocol (SRTP) defines a profile of Real Time Transport Protocol (RTP), intended to provide encryption, message authentication and integrity and replay protection to the RTP data in both unicast and multicast applications. Previous
work is to evaluate the trade-off existing between quality of service and security when SRTP  is employed to protect RTP (Real Time Protocol) sessions on VoIP calls . There is no such study has been conducted on comparison of VoIP one-to-one call and multi conference call (many-to-many) performance using SRTP functionality. With its promise of inclusion, innovation, and growth, VoIP also brings challenges. VoIP is not easy to secure. It suffers all of the problems associated with any Internet application, and VoIP security is complicated by its interconnection to the PSTN. A host of trust, implementation, and operational complexities make securing VoIP particularly complex. In fact, the same aspects that make the VoIP software model so powerful—its flexible, open, distributed design—are what make it potentially problematic . Various security requirements have to be met to secure VoIP transmission: Authentication, Privacy and Confidentiality, Integrity, Non repudiation, Non replay and Resource availability . The threats faced by a VoIP are similar to other applications including: unwanted communication (spam), privacy violations (unlawful intercept), impersonation (masquerading), theft-of service, and denial-of-service . II. METHODOLOGY
We have setup a real wireless network environment to analyze and measure implementation of VoIP service using security function (SRTP) at University of Kuala Lumpur (UniKL) in Malaysia. This study posits several research questions: i) what is the STRP performance level of the VoIP over WLAN based on one-to-one call...