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 accredited Time Transport confabulations protocol on VoIP over radio receiver LAN in Campus Environment Mohd Nazri Ismail Department of MIIT, University of Kuala Lumpur (UniKL), MALAYSIA email sheltered unikl. edu. my Abstract- In this research, we propose to implement Secure Real Time Transport communications 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 amid two parties.Therefore, this study is to analyzed SRTP cognitive process on disparate VoIP codec selection over wired. We induct implemented a real VoIP network in University of Kuala Lumpur (UniKL), Malaysia. We part softph 1 as our medium communication between two parties in campus environment. The results show that implementation of S RTP is able to improve the VoIP quality between one-to-one conversation and multi throng holler ( galore(postnominal)-to-many). In our experiment, it shows that iLBC, SPEEX and GSM codec are able to improve substantively the multi group discussion (many-to-many) VoIP quality during conversation.In additional, implementation of SRTP on G. 711 and G. 726 codec volition decrease the multi multitude (many-to-many) VoIP quality. Keywords- Codecs, Softphone, SRTP, WLAN I. asylum 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 object glass of this study is to enable the security function using Secure Real Time Transport Protocol (SRTP).We pass on 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 pr ovides low delay and high packet freeing robustness for low-bit rate codecs. 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 ncryption, message authentication and integrity and rematch 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 6 is employed to protect RTP (Real Time Protocol) sessions on VoIP calls 5. 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 squall 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 a pplication, and VoIP security is complicated by its interconnection to the PSTN. A host of trust, implementation, and operational complexities make securing VoIP specially complex. In fact, the same aspects that make the VoIP software model so powerfulits flexible, open, distributed designare what make it potentially problematic 78. Various security requirements have to be met to secure VoIP transmission Authentication, Privacy and Confidentiality, Integrity, Non repudiation, Non replay and Resource availability 9.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 10. 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 and multi conference call? nd ii) which codecs are able to provide better improvement of VoIP conversation? throw 2. 1 and inning 2. 2 show the function of VoIP conversation call between one-to-one and multi conference. We measure our voice quality using human perception. Mean Opinion Score (MOS) technique is the best court to measure and ISSN 0975-3397 898 Mohd Nazri Ismail / (IJCSE) International Journal on Computer Science and Engineering Vol. 02, No. 03, 2010, 898-902 validate voice quality between one-to-one call and multi conference call. fancy 2. 3 shows the measurement of VoIP performance over WLAN using SRTP implementation.We also test on different codecs selection such G. 711, G. 726, GSM, iLBC and SPEEX. III. ANALYSIS AND RESULTS Figure 2. 1 VoIP over one-to-one talk This section measures and compares VoIP performance over WLAN using SRTP function. In voice and video communication, quality us ually dictates whether the picture is a dependable or bad one. Besides the qualitative description we hear, like quite good or very bad, there is a numerical manner of expressing voice and video quality. It is called Mean Opinion Score (MOS). MOS can be tested using i) human perception ii) simulation model and iii) automated frame 1 2.MOS gives a numerical indication of the perceived quality of the media received after being transmitted and eventually compressed using codecs. MOS is expressed in one number, from 1 to 5, 1 being the worst and 5 the best. MOS is quite subjective as it is based figures that result from what is perceived by people during tests (refer to card 3. 1). We will select five different users to evaluate and rate the VoIP performance using SRTP and without SRTP functionality. When users cannot get a dial tone or there are inordinate delays in ringing the other partys phone, VoIP performance is unacceptable.Call quality is a function of packet loss rate, de lay, and jitter is typically represented as a MOS 3, 4. Table 3. 1 Mean Opinion Score (MOS) Ratings Mean Opinion Score (MOS) Ratings Excellent 5 (Perfect. Like face-to-face conversation Figure 2. 2 VoIP over Many-to-Many (Multi Conference) conference Good Fair Poor Bad or radio reception) 4 (Fair. Imperfections can be perceived, but sound still clear. This is (supposedly) the throw away for cell phones) 3 (Annoying) 2 (Very annoying. Nearly impossible to communicate) 1 (Impossible to communicate) Figure 3. shows the configuration of codec protocol such as G. 711, G. 726, GSM, iLBC and SPEEX. This 3CX softphone is able to active Echo Cancellation and SRTP. The VoIP experiments will receive two types of modes i) one-to-one call conversation ii) multi conference call (many-to-many). Figure 3. 2 shows the result of VoIP one-to-one conversation. Figure 3. 3 shows the result of VoIP multi conference (many-to-many) call. Figure 2. 3 Measurement and Evaluation of VoIP over WLAN using SRTP Approach ISSN 0975-3397 899 Mohd Nazri Ismail / (IJCSE) International Journal on Computer Science and Engineering Vol. 2, No. 03, 2010, 898-902 improvement on VoIP quality performance and at the same time able to provide element of security (refer to Table 3. 3 and Figure 3. 5). The significant improvement is GSM and SPEEX codecs after implemented SRTP. Table 3. 2 Multi Conference without SRTP drug user Codec Figure 3. 1 3CX Softphone Codec and SRTP Configuration G. 711 G. 726 GSM iLBC SPEEX User 1 3 4 1 2 5 User 2 3 3 1 2 4 User 3 2 3 1 3 4 User 4 3 4 1 2 4 User 5 2 4 1 2 5 Figure 3. 2 one-to-one Call Conversation Result Figure 3. 4 Users Rate VoIP for Multi Conference Call Without SRTP Table 3. Multi Conference with SRTP User Codec G. 711 G. 726 GSM iLBC SPEEX Figure 3. 3 Multi Conference Call (many-tomany) Conversation Result Most of the users concord and judge this VoIP without SRTP will provide a good quality for G. 711 and G. 726 codecs. Other users agreed and rates 4 to 5 ratings for SPEEX codec without using SRTP during multi conference conversation (refer to Table 3. 2 and Figure 3. 4). After implemented SRTP on VoIP during multi conference session occurs, it shows few User 1 2 3 4 5 5 User 2 1 3 4 5 5 User 3 1 2 3 4 5 User 4 2 2 3 4 5 User 5 1 2 3 4 5 ISSN 0975-3397 900Mohd Nazri Ismail / (IJCSE) International Journal on Computer Science and Engineering Vol. 02, No. 03, 2010, 898-902 Table 3. 5 One-to-One Call with SRTP User Codec G. 711 G. 726 GSM iLBC SPEEX User 1 2 3 2 4 5 User 2 1 3 2 4 4 User 3 1 2 2 4 5 User 4 2 2 2 4 4 User 5 2 3 2 4 5 Figure 3. 5 Users Rate VoIP for Multi Conference Call With SRTP Most of the users agreed and rates this VoIP oneto-one call without SRTP will also provide low quality for G. 711, G. 726 and GSM codecs. Other users agreed and rates 3 and 5 ratings for iLBC and SPEEX codecs without using SRTP during one-to-one call (refer to Table 3. and Figure 3. 6). After implemented SRTP on VoIP during one-to-one sessio n occurs, it shows significant improvement on VoIP quality performance for G. 711, G. 726, GSM, iLBC and SPEEX over WLAN (refer to Table 3. 5 and Figure 3. 7). Table 3. 4 One-to-One Call Without SRTP User Codec G. 711 G. 726 GSM iLBC SPEEX User 1 2 1 2 3 5 User 2 2 2 2 3 4 User 3 2 1 2 4 4 User 4 1 1 1 3 4 User 5 1 2 2 4 4 Figure 3. 7 Users Rate VoIP for One-to-One Call with SRTP Figure 3. 8 and Figure 3. 9 show the comely MOS tag for VoIP conversation over one-to-one call and multi conference call (many-to-many), respectively.VoIP Conversation over Multi Conference Call Before implemented SRTP, the average MOS score for G. 711 is 2. 5, 3. 5 for G. 726, 1 for GSM, 2. 1 for iLBC and 4. 5 for SPEEX. After implemented SRTP, the average MOS score for G. 711 and G. 726 are decreased the ratings approximately 1 to 2. 5. GSM, iLBC and SPEEX codecs show the average MOS score are 3. 5, 4. 5 and 5. GSM, iLBC and SPEEX codec show the increasing of VoIP performance after implemented SRTP (ref er to Figure 3. 8). VoIP Conversation over One-to-One Call Before implemented SRTP, the average MOS score for G. 711 is 1. , 1. 4 for G. 726, 1. 8 for GSM, 3. 5 for iLBC and 4. 2 for SPEEX. After implemented SRTP, the average MOS score shows the significant improvement for G. 711, G. 726, GSM, iLBC and SPEEX codecs. Therefore, implementation of SRTP can improve the VoIP quality performance for one-to-one call over WLAN (refer to Figure 3. 9). Figure 3. 6 Users Rate VoIP for One-to-One Call without SRTP ISSN 0975-3397 901 Mohd Nazri Ismail / (IJCSE) International Journal on Computer Science and Engineering Vol. 02, No. 03, 2010, 898-902 dependency conditions that could influence voice quality.Future work, we will flourish our experiment on VoIP over VPN implementation in Campus environment. 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, 20 07. 2. Masuda M. & Ori K. Delay Variation Metrics for name and address 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. peal & 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 rethink, 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. 2-96, 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(a) WORK Based on the results, implementation of SRTP using GS M, 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 gain 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 holders express written permission. However, users may print, download, or email articles for individual use.