臺灣博碩士論文加值系統

IP網路電話系統最近幾年在網際網路上是一種滿受歡迎的應用。本論文首先討論一些會影響整體IP網路電話系統品質的關鍵性因素，再針對這些問題一一提出解決改善的方法。這些會影響通訊品質的問題包括語音壓縮之處理效能、保持語音連續性的暫存區控制以及如何維護管理已上線用戶之服務品質。本論文首先討論語音壓縮的問題，我們選擇G.723.1語音壓縮編解碼器作為IP網路電話系統中之語音壓縮處理方式，這是一個非常重要的音訊壓縮標準，然而此語音壓縮編解碼器的原理及程序十分複雜，所以提昇語音編解碼處理時間是很重要的。再來良好的暫存區控制機制能夠有效的解決語音不連續的問題，進而提高IP網路電話系統的通話品質。我們提出一個動態聲音播放方法(DSRM)去克服播放顫動之問題。服務品質代理(Qos Agent)被發展用來動態管理網路之頻寬使用，我們發展自動偵測網路流量演算法來建立允入控制機制(Admission Control Mechanism)，決定是否要讓用戶上線與對方通話。一旦網路電話用戶上線之後，在IP網路電話系統中使用資源保留協定(RSVP)來保留所使用到的頻寬資源，使之在網際網路上保持良好的通訊品質。資源保留協定將會確保封包在網路上的傳輸品質。基於以上所使用之方法，我們可得到良好的IP網路電話系統之通訊品質。

Recently, Voice-over-IP (VoIP) is a popular application on the Internet. Some critical issues affect seriously the total quality of a VoIP system. The critical issues include the computation performance of audio codec, and the buffer control scheme to keep speech smooth, and the quality of VoIP service. We propose some approaches to solve the problems in VoIP system. To raise the codec performance, G.723.1 codec is chosen and improved as the VoIP system audio codec. In this thesis, some techniques are developed to speed up the performance of G.723.1 codec. Then a novel buffer control mechanism is proposed to solve the problem of discontinuous speech. This mechanism is based on a proposed Dynamic Sound Replay Method (DSRM), which can solve the replay jitter problem. In order to guarantee the service quality on network, the Resource Reservation Protocol (RSVP) is used to preserve the bandwidth for each VoIP connection. A Qos agent, which can dynamically manage the network bandwidth and perform admission control, is developed. The admission control algorithm can automatically detect the traffic situation in network to make proper QoS decision. Based on the proposed approaches, the quality of VoIP systems can be raised excellently.

Chapter 1 Introduction
1.1 Background
1.2 Motivation
1.3 Organization of thesis
Chapter 2 Critical Issues on VoIP System Design
2.1 System Architecture
2.2 Delay Analysis of VoIP System
2.3 Audio Codec
2.3.1 Description of Encoder Block
2.3.2 Description of Decoder Block
2.4 Buffer Control in VoIP System
2.5 RSVP Protocol Overview
2.5.1 Operation of RSVP Protocol
2.5.2 RSVP Features
Chapter 3 Improvement of G.723.1 Audio Codec
3.1 Analysis of G.723.1 Audio Codec
3.2 Refinement Methods for G. 723.1 Audio Codec
3.2.1 Refinement with no Distortion
3.2.2 Refinement with Distortion
3.3 Refinement Result of G.723.1
Chapter 4 Methodology of Buffer Design
4.1 Significance of Buffer Design in VoIP System
4.2 Methodology of Buffer Design
4.3 Design Procedure of VoIP Buffer
Chapter 5 Bandwidth-Efficient QoS Mechanism with Performance Evaluation
5.1 M/M/1 Queue Model
5.2 Admission Control based on Traffic Measurement for RSVP
5.3 Experimental Results and Performance Evaluation
Chapter 6 Discussion and Conclusion
6.1 Conclusions
6.2 Future Works

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