A Scheme For High Quality VoIP Service achieved by Fast TCP Mechanism

• Main Authors: Hunewei Shu, 許宏偉
• Other Authors: Mong-Fong Horng
• Format: Others
• Language: zh-TW
• Published: 2006
• Online Access: http://ndltd.ncl.edu.tw/handle/74636487624661603443

碩士 === 樹德科技大學 === 資訊工程學系 === 94 === Voice over Internet Protocol (VoIP) has recently been widely used because of the growth of internet bandwidth . VoIP is a technique for transmitting voice data over IP networks. The following steps are performed during transmission: digitization of the analog signal; encoding/compress the digital signal; packet encapsulation and transmission of the packets on the network. Most VoIP applications usually perform over an RTP stack that is implemented on the top of UDP/IP protocol. However the UDP has weaknesses in some situations. For example, UDP offers no congestion control and acknowledgement mechanisms. Traditional TCP protocol although can solve the above problems under UDP , but a disadvantage may occur, the throughput of VoIP at startup may be too low to meet the requirement of VoIP Codec because of the TCP “sliding widow”. The quality of voice will degrade seriously during startup period under TCP. In this paper we will discuss above problems and we propose a scheme to provide high throughput service for VoIP services achieved by adjusting TCP congestion window parameters before transmitting VoIP data. The simulation studies presented in this paper were conducted using NS2 . We will pre -sent the results of the experimental and show the improvement of throughput under our scheme. The goal of our investigation is to understand whether the throughput of VoIP service could be improved by using our scheme under TCP protocol. And our results will show that the scheme we proposed can improve the throughput of VoIP via a series of simulations.