Performance analysis of voice codec for VoIP

• 第一著者: Abd. Khuther, Ali
• フォーマット: 学位論文
• 言語: 英語
• 出版事項: 2008
• 主題: HE Transportation and Communications; QA75 Electronic computers. Computer science
• オンライン･アクセス: http://eprints.utm.my/9553/1/AliAbdKhutherMFC2008.pdf

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.