Performance analysis of voice codec for VoIP
Recently, VoIP (Voice over Internet Protocol) is a great interesting voice communication over the Internet, with high level quality of service (QoS) along with circuit switch and cellular. The objective in this project is to assess to what extend today’s internet in meeting this expectation via st...
محفوظ في:
| المؤلف الرئيسي: | |
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| التنسيق: | أطروحة |
| اللغة: | English |
| منشور في: |
2008
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| الموضوعات: | |
| الوصول للمادة أونلاين: | http://eprints.utm.my/id/eprint/9553/1/AliAbdKhutherMFC2008.pdf |
| الوسوم: |
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my-utm-ep.9553 |
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| record_format |
uketd_dc |
| institution |
Universiti Teknologi Malaysia |
| collection |
UTM Institutional Repository |
| language |
English |
| topic |
HE Transportation and Communications HE Transportation and Communications |
| spellingShingle |
HE Transportation and Communications HE Transportation and Communications Abd. Khuther, Ali Performance analysis of voice codec for VoIP |
| description |
Recently, VoIP (Voice over Internet Protocol) is a great interesting voice communication over the Internet, with high level quality of service (QoS) along with circuit switch and cellular. The objective in this project is to assess to what extend today’s internet in meeting this expectation via studying VoIP performance and its QoS. However, the methodology in this project is, first the CODECs are selected by some criteria then apply them on SIP server to finally come out with the result from the simulation in order to make comparison and analysis the QoS. This work implements VoIP protocols for two connected user using SIP server with its three CODEC algorithms. After define the main problems in this area set of parameters are taken into account due to their affection to the performance of the voice, such as jitter, packet loss, packet delay and throughput. This project is simulated three existing CODECs (converting the voice from analog to digital and compressing the packets) using the most common CODECs with VoIP, they are G.711, G.723 and G.729. However, the simulation will use NS2 platform with vary values of packet size and number of calls. Finally, the main objective from this project is to obtain a high quality of voice by make a proper decision for choosing the codec voice. As conclusion, G.711 is a preferred technique when the quality is required because of the high throughput from its packets, while G.723 perform well with the high bandwidth means it can handle many user. Finally, G.729 the high level compression is the proper technique for many user and heavy data only when the quality is not taken into account. |
| format |
Thesis |
| qualification_level |
Master's degree |
| author |
Abd. Khuther, Ali |
| author_facet |
Abd. Khuther, Ali |
| author_sort |
Abd. Khuther, Ali |
| title |
Performance analysis of voice codec for VoIP |
| title_short |
Performance analysis of voice codec for VoIP |
| title_full |
Performance analysis of voice codec for VoIP |
| title_fullStr |
Performance analysis of voice codec for VoIP |
| title_full_unstemmed |
Performance analysis of voice codec for VoIP |
| title_sort |
performance analysis of voice codec for voip |
| granting_institution |
Universiti Teknologi Malaysia, Faculty of Computer Science and Information System |
| granting_department |
Faculty of Computer Science and Information System |
| publishDate |
2008 |
| url |
http://eprints.utm.my/id/eprint/9553/1/AliAbdKhutherMFC2008.pdf |
| _version_ |
1747814753359626240 |
| spelling |
my-utm-ep.95532018-10-14T07:19:57Z Performance analysis of voice codec for VoIP 2008-10 Abd. Khuther, Ali HE Transportation and Communications QA75 Electronic computers. Computer science Recently, VoIP (Voice over Internet Protocol) is a great interesting voice communication over the Internet, with high level quality of service (QoS) along with circuit switch and cellular. The objective in this project is to assess to what extend today’s internet in meeting this expectation via studying VoIP performance and its QoS. However, the methodology in this project is, first the CODECs are selected by some criteria then apply them on SIP server to finally come out with the result from the simulation in order to make comparison and analysis the QoS. This work implements VoIP protocols for two connected user using SIP server with its three CODEC algorithms. After define the main problems in this area set of parameters are taken into account due to their affection to the performance of the voice, such as jitter, packet loss, packet delay and throughput. This project is simulated three existing CODECs (converting the voice from analog to digital and compressing the packets) using the most common CODECs with VoIP, they are G.711, G.723 and G.729. However, the simulation will use NS2 platform with vary values of packet size and number of calls. Finally, the main objective from this project is to obtain a high quality of voice by make a proper decision for choosing the codec voice. As conclusion, G.711 is a preferred technique when the quality is required because of the high throughput from its packets, while G.723 perform well with the high bandwidth means it can handle many user. Finally, G.729 the high level compression is the proper technique for many user and heavy data only when the quality is not taken into account. 2008-10 Thesis http://eprints.utm.my/id/eprint/9553/ http://eprints.utm.my/id/eprint/9553/1/AliAbdKhutherMFC2008.pdf application/pdf en public http://dms.library.utm.my:8080/vital/access/manager/Repository/vital:1112 masters Universiti Teknologi Malaysia, Faculty of Computer Science and Information System Faculty of Computer Science and Information System Adaptive Digital Technologies. (2007). U.S. : Adaptive Digital from : http://www.adaptivedigital.com. Advance Waterfall Research Methodology. (1999). Malaysia : Research Methodology from : www.dspace.fsktm.um.edu.my/bitstream/1812/160/5/Chapter3.pdf Alias, M. and Ong, L. L. (2006). Performance of Voice over IP (VoIP) over a wireless LAN (WLAN) for different audio/voice codecs. Malaysia: Universiti Teknologi Malaysia. Amrir, M., Tariq, M.and Noor, M.(2005). Evaluation of VoIP Quality over the Pakistan Internet Exchange (PIE) Backbone. Pakistan. Christin, N., (2004). Building ns-2 on Cygwin[version 2.27, 2.26, and 2.1b9a(*), UC Berkeley – School of Information Management and Systems. Cisco Document Server, “Traffic Analysis for Voice over IP, Posted September 2002. Daniel, M. and Emma, M. (2002). Delivering Voice Over IP Networks. (2nd Ed). U.S. : Wiley. Davis, Y. P. (1993). Digital Audio Compression. A survey: IEEE, Part D. TJA03P8. INTERNATIONAL TELECOMMUNICATION UNION. (1988). Switzerland: Geneva. ITU-T Recommendation G.107. 2003, The E-model, a computational model for use in transmission planning. ITU-T Recommendation G.114, 2003, One way transmission time. ITU-T Recommendation P.59. Artificial conversational speech. Ixia. (2005). Assessing VoIP Call Quality Using the E-model. CA 91302. Calabasas : West Agoura Road. Jim, D. and Neil, A. (2006). Internet Phone Services Simplified. U.S.: Indianapolis Cisco Press. Jonthan, D. , James, P. , Manoj, B. Satish, K. and Sudipto, M. (2007). Voice Over IP Fundamentals. (2nd Ed). U.S.: Indianapolis Cisco Press. Kee,N.T.,Yin,and Moh. (2005). Voice Performance Study on Single Radio Multihop IEEE 802.1 lb Systems with Chain Topology. Malaysia: Asian Research Center. Kerry, C. (2006). A Guide to Open Source Audio Streaming. (1st Ed). U.S. : IT Security Consultant. Lin, C., Xuemin, S., Jon, W., Lin C.(2005). Voice Capacity Analysis of WLAN with Unbalanced Traffic.China. Michele, L.(1996). SIP Module for Network Simulator 2.Italia. Monroe Street Santa Clara (2000). CA 95051-1450. U.S. : Extreme Networks. Network Simulator 2. http://www.isi.edu/nsnam. Oliver, C. I. 2002, Converged Network Architectures, Delivering Voice and Data over IP, ATM, and Frame Relay.1st edition. United States of America: John Wiley & Sons, Inc. Oliver, H. , David, G. and Jean, P. (2000). IP Telephony Packet-Based Multimedia Communications Systems. U.S. : Pantek Arts, maidstone. Richard, S. (2002). Comparison of Voice over IP with circuit switching Techniques. A survey: CiteSeer.ist. Toral, H. and Torres, D. (2005). Traffic Analysis for IP Telephony. IEEE , 05EX1097. Mexico: Guadalajara. VOCAL Technologies. (2007). ITU Recommendation G.711. from: http://www.vocal.com/data_sheets/g711.html . What is a VoIP codec ?.(2005). Retrieved on March 1, 2008, from http://www.tech-faq.com/voip-codec.shtml. |