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Please use this identifier to cite or link to this item: http://ir.lib.stu.edu.tw:80/ir/handle/310903100/1294

Title: 利用快速TCP機制實現高品質VoIP服務之可行性研究
A Scheme For High Quality VoIP Service achieved by Fast TCP Mechanism
Authors: 許宏偉
Hunewei Shu
Contributors: 洪盟峰
Mong-Fong Horng
資訊工程學系
Keywords: 網路電話;TCP 連線;吞吐量
VoIP;TCP Connection;throughput
Date: 2006
Issue Date: 2011-05-24 15:12:08 (UTC+8)
Publisher: 高雄市:[樹德科技大學資訊工程學系]
Abstract: 近年來由於寬頻網路的普及,VoIP(Voice over Internet Protocol)的應用隨之興起,網路上開始出現大量的VoIP服務。所謂VoIP服務是指透過IP網路來傳輸聲音語音訊號,所以VoIP就是一種可以在分封交換的網路上互傳類比音訊的技術。簡單地說,VoIP的運作機制是藉由連續的採樣轉碼、編碼、壓縮、封裝與傳送等程序來完成。
現有VoIP 協定中,大部份VoIP協定之連線均為採用UDP方式來傳送語音封包,少部份VoIP在某些環境(時機)才會採用TCP來傳語音封包。既有的UDP 方式傳送語音資料雖有很多優點,但仍然有些是UDP無法達到的,例如流量控制與封包遺失等問題;TCP是可以解上述問題,但傳統TCP也並非是完全適合VoIP服務傳送的協定,最大的問題之一就是起步的吞吐量明顯的受到緩啟動的因素所抑制而不能滿足VoIP Codec所需,因而導致VoIP連線剛建立時的語音品質不佳。
在本文中主要是在討論原有TCP 方式傳送VoIP時起步吞吐量不足的問題;並且我們提出 “快速TCP機制”的方式取代原來的TCP方式傳送VoIP資料。快速TCP機制藉由在傳送語音資料前修改TCP初始視窗大小,讓起步傳送時吞吐量拉高到Codec傳輸所需,因而解決起步吞吐量不足的問題。
在本文中我們以NS2網路模擬工具進行一連串的實驗,藉由實驗中調整不同的環境參數觀察吞吐量與RTT時間、封包大小與Codec傳輸率大小之間的關係;同時在傳統TCP與快速TCP兩者吞吐量對照圖表中,比較出快速TCP機制的改善效果。
最後我們在文中討論使用快速TCP機制來傳送高品質的VoIP服務的可行性及其應用範圍;並且提出目前該機制在吞吐量需求產出的問題,意即需要改進的部份。然後討論未來使用TCP傳送VoIP在服務品質方面可發展的方向。
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.
Appears in Collections:[資訊工程系(所) ] 博碩士論文

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