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研究生:陳維浩
研究生(外文):Wei-Hao Chen
論文名稱:網際網路上語音傳輸機制之研究
論文名稱(外文):Delivering Voice over the Internet
指導教授:郭經華郭經華引用關係
指導教授(外文):Chin-Hwa Kuo
學位類別:碩士
校院名稱:淡江大學
系所名稱:資訊工程學系
學門:工程學門
學類:電資工程學類
論文種類:學術論文
論文出版年:1999
畢業學年度:87
語文別:中文
論文頁數:75
中文關鍵詞:語音封包播放延遲錯誤復原重複封包發送率控制網際網路
外文關鍵詞:Packetized AudioPlayout DelayError RecoveryRedundant PacketRate ControlInternet
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  • 收藏至我的研究室書目清單書目收藏:1
  現今網際網路採用TCP/IP通訊協定群,傳輸資料無服務品質 (Quality of Service) 的保證,再加上各網路頻寬和軟硬體能力的不同,因而會造成封包延遲差異 (packet delay jitter) 和封包遺失 (packet loss) 等不利的情形。因此,在網路環境沒有明顯改善的情況之下,必須提供適切的機制才能在網際網路上進行良好的即時性語音傳輸。
  本論文提出三個機制。一、可調適語音播放機制:藉由線上統計最近一段時間內封包的網路延遲時間和封包丟失情況以求得適當的播放延遲時間,使語音儘可能以一定的品質連續地播放。二、錯誤復原機制:藉著傳送重複資訊 (redundant information) 使得遺失封包的主要資訊 (main information) 可從隨後到達封包的重複資訊加以復原,如此可以降低語音內容遺失的程度。三、發送率控制機制:根據網路擁塞狀況和封包網路丟失率來決定封包主要資訊和重複資訊的組合策略以克服網路頻寬需求的問題。
  本方案有效地解決語音封包經由網際網路傳輸而遺失以及多種編碼方式下封包的主要資訊和重複資訊該如何組合的問題。本論文所提出的機制可提供網路系統開發者運用在網路電話、網路廣播、視訊會議等方面,藉以改善在目前網路環境下即時語音傳輸品質不佳的情況。

   Nowadays, the Internet adopts TCP/IP, which grants no assurance of Quality of Service. Furthermore, the individual network bandwidth differs and software as well as hardware devices diverge, both facts cause packet delay jitter and packet loss. If the network environment is the same still, we must provide proper mechanisms in order to transfer real-time audio in the Internet with acceptable quality.
   This thesis provides three mechanisms. Firstly, the adaptive audio playout mechanism gets proper playout delay time by on-line statistics of the packet delay time and packet loss rate of the recent time intervals in order to make the audio stream playback continuous with certain quality. Secondly, the error recovery mechanism makes the main information of the loss packets to be recovered from the following packets with redundant information so as to reduce the degree of content loss. Thirdly, the rate control mechanism selects the combination strategy of the main information and redundant information of the packets based on the network congestion state and the network loss rate of the packets in order to conquer the issue of the need of the network bandwidth.
   This project effectively solves the problems of packet loss out of delivering over the Internet and of the combination strategy of the packets under several codecs. Network system developers can apply the mechanisms provided here to Internet Telephony, Internet Broadcast, and Video Conferencing.

第一章 緒論
 1.1 研究動機
 1.2 研究內容
 1.3 論文架構
第二章 網際網路上語音傳輸技術
 2.1 語音數位化
  2.1.1 語音編碼
  2.2.2 語音壓縮與國際標準
 2.2 靜音偵測與迴音消除
 2.3 即時傳輸協定
第三章 語音傳輸服務品質控制
 3.1 互動性考量
 3.2 可調適語音播放機制
  3.2.1 延遲差異與控制
  3.2.2 以線性遞迴方式估計播放時間
  3.2.3 以分佈統計方式估計播放時間
 3.3 錯誤復原機制
  3.3.1 封包丟失與丟失復原
  3.3.2 資料重複式之正向錯誤修正
  3.3.3 重建延遲時間
 3.4 發送率控制機制
  3.4.1 TCP-friendly
  3.4.2 結合錯誤復原機制於發送率控制
第四章 可調適即時語音傳輸方案
 4.1 延遲與丟失測量機制
  4.1.1 封包i非新的談話時期開始封包
  4.1.2 封包i為新的談話時期開始封包
 4.2 服務品質調整機制
第五章 結論與未來發展
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