ABSTRACT
VoIP users increase each day. However, the documentation on the behavior of VoIP applications is still lacking. The needs to understand and generalize the behavior of applications like Skype, GoogleTalk and SIP based applications grow each day. There are many factors that influenced the performance of a VoIP application such as bandwidth, packet loss rate, delay, jitter, codec type and CPU power of the end devices. The user experience of the service is important since, VoIP is a real time application running over the best effort internet. Since VoIP data co-exist with other data on the internet, extracting, transforming, loading and analyzing the selected VoIP application is a challenge. We design an instrument to do data collections, data massaging, data analysis and data interpretation of large amounts of network packet. The result shows that GoogleTalk, Skype and Express Talk are more sensitive to the impairments due to packet loss rate and jitter rather than to the impairment due to delay. Bandwidth and other resources like a de-jitter buffer and a gateway's CPU and memory are important in order to produce a good quality VoIP service. The lack of these resources would result in several packets lost before they reach a destination or the packets arrive too late to join the other packets in the de-jitter buffer at the destination gateway. The gateway would drop these packets if the de-jitter buffer is full or not enough memory or CPU powers to process the packets. A gateway closer to the receiver end decapsulates IPSec or TLS packets. The gateway also decodes the voice packets before the packets entering the receiving machine. High throughputs do not imply high Perceptual Evaluation of Speech Quality for Wideband (PESQ-WB) scores. The throughput size is determined by the codec type and the security features that are implemented on the infrastructure.
- S.-H. Choi and M. Handley, "Designing TCP-friendly window-based congestion control for real-time multimedia applications," Inproceedings of PFLDNeT, 2009.Google Scholar
- G. Eason, E. Brosh, S. Baste, V. Misra, D. Rubenstein, and H. Schulzrinne, "The Delay-Friendliness of TCP for Real-Time Traffic," IEEE/ACM Transactions on Networking, vol. 18, no. 5, pp. 1478--1491, 2010. Google ScholarDigital Library
- P. Hunter, "Journey to the centre of big data," Engineering Technology, vol. 8, pp. 56--59, April 2013.Google ScholarCross Ref
- C. Hattingh and T. Szigeti, "End-to-End QoS Network Design: Quality of Service in LANs," WANs, and VPNs, CiscoPress, USA, p. 44, 2004. Google ScholarDigital Library
- D. Kuhn, T. Walsh, and S. Fries, "Security considerations for voice over IP systems," NIST special publication, pp. 800--58, 2005.Google ScholarDigital Library
- F. Michaut and F. Lepage, "Application-oriented network metrology: Metrics and active measurement tools," IEEE Communications Surveys & Tutorials, vol. 7, no. 2, 2005. Google ScholarDigital Library
- M. Aida, N. Miyoshi, and K. Ishibashi, "A scalable and lightweight QoS monitoring technique combining passive and active approaches," in INFOCOM 2003. Twenty-Second Annual Joint Conference of the IEEE Computer and Communications, vol. 1, pp. 125--133, IEEE, 2003.Google Scholar
- A. Saad, I. Phillips, and A. Salagean, "A Framework for Monitoring the Performance of Secure VoIP," in Proceedings of the Third Internatioal conference on Internet Technologies and Applications (ITA 09), pp. pp.495--505, 2009.Google Scholar
- J. Broß and C. Meinel, "Can VoIP Live up to the QoS Standards of Traditional Wireline Telephony?," in AICT '08. Fourth Advanced International Conference on, pp. 126--132, June 2008. Google ScholarDigital Library
- FIRST. Forum of Incident Response and Security Teams, "Common Vulnerability Scoring System (CVSS-SIG)." https://www.first.org/cvss, Last Visited Jan 2016.Google Scholar
- M. Ranganathan and L. Kilmartin, "Performance analysis of secure session initiation protocol based VoIP networks," Computer Communications, vol. 26, no. 6, pp. 552--565, 2003. Google ScholarDigital Library
- A. Mohiuddin and M. Abdul Malik, "CPU dimensioning on performance of Asterisk VoIP PBX," in Proceedings of the 11th communications and networking simulation symposium, pp. 139--146, ACM, 2008. Google ScholarDigital Library
- L. Gaspary, M. Barcellos, A. Detsch, and R. Antunes, "Flexible security in peer-to-peer applications: Enabling new opportunities beyond file sharing," Computer Networks, vol. 51, no. 17, pp. 4797--4815, 2007. Google ScholarDigital Library
- K. Salah and M. Hamawi, "Impact of CPU-bound Processes on IP Forwarding of Linux and Windows XP," J. UCS, vol. 16, no. 21, pp. 3299--3313, 2010.Google Scholar
- J. Balen, G. Martinovic, and Z. Hocenski, "Network performance evaluation of latest windows operating systems," in Software, Telecommunications and Computer Networks (SoftCOM), 2012 20th International Conference on, pp. 1--6, IEEE, 2012.Google Scholar
- I. BARONÁK and M. Halas, "Mathematical representation of VoIP connection delay," Radioengineering, vol. 16, no. 3, p. 77, 2007.Google Scholar
- B. Kyrbashov, I. Baronak, M. Kovacik, and V. Janata, "Evaluation and investigation of the delay in VoIP networks," vol. 20, no. 2, pp. 540--547, 2011.Google Scholar
- D. T. Tran and E. Choi, "A reliable UDP for ubiquitous communication environments," in Proceedings of WSEAS International Conference on Computer Engineering and Applications, Citeseer, 2007. Google ScholarDigital Library
- X. Shi, G. Tian, and Y.-C. Tian, "A reliable real-time transport protocol for control systems over wireless networks," in Telecommunication Networks and Applications Conference, Australasian, pp. 1--6, IEEE, 2012.Google Scholar
- T. Le, G. Kuthethoor, C. Hansupichon, P. Sesha, J. Strohm, G. Hadynski, D. Kiwior, and D. Parker, "Reliable user datagram protocol for airborne network," in IEEE Military Communications Conference, pp. 1--6, IEEE, 2009. Google ScholarDigital Library
- J. Yan, W. Mühlbauer, and B. Plattner, "Analytical framework for streaming over TCP," TIK Report, no. 333, 2010.Google Scholar
Index Terms
- Big data analysis on secure VoIP services
Recommendations
An empirical evaluation of VoIP playout buffer dimensioning in Skype, Google talk, and MSN Messenger
NOSSDAV '09: Proceedings of the 18th international workshop on Network and operating systems support for digital audio and videoVoIP playout buffer dimensioning has long been a challenging optimization problem, as the buffer size must maintain a balance between conversational interactivity and speech quality. The conversational quality may be affected by a number of factors, ...
Effective packet loss estimation on VoIP jitter buffer
IFIP'12: Proceedings of the 2012 international conference on NetworkingThe paper deals with an influence of network jitter on effective packet loss in dejitter buffer. We analyze behavior of jitter buffers with and without packet reordering capability and quantify the additional packet loss caused by packets dropped in ...
The Effect of Packet Delay on Voip Speech Quality: Failure of Hurst Method
CSIE '09: Proceedings of the 2009 WRI World Congress on Computer Science and Information Engineering - Volume 07Voice over Internet Protocol (VoIP) is a technology that transports voice data packets across packet switched networks using the Internet Protocol (IP). However, the current Internet, which was originally designed for data communications, provides best-...
Comments