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研究生:李建用
研究生(外文):Jian-Yong Lee
論文名稱:一個支援雙模式之語音傳輸:以變動框架長度為基礎之遺失率估算法
論文名稱(外文):Supporting two-mode voice transportation: A loss rate estimation based on variable frame size
指導教授:陳俊麟陳俊麟引用關係
指導教授(外文):Chin-Ling Chen
學位類別:碩士
校院名稱:國立屏東商業技術學院
系所名稱:資訊管理系
學門:電算機學門
學類:電算機一般學類
論文種類:學術論文
論文出版年:2005
畢業學年度:93
語文別:中文
論文頁數:109
中文關鍵詞:遺失率交通量測RTCPRTP
外文關鍵詞:RTPRTCPloss ratetraffic estimation
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本文假設在傳送端有一個雙模式的編碼器,可支援兩個不同音質之檔案(PCM 和 GSM)。傳送端根據以變動框架長度為基礎之遺失率估算法,來估算目前網路上封包遺失的狀況,並以此作為切換高低音質檔案的依據。此演算法可過濾短期突發交通,並呈現長期交通之趨勢。除具有穩定與敏捷的特性外,此演算法亦能避免音質切換過度頻繁。
In this paper, we assume that the sender supports a two-mode codec with two audio sample files (PCM and GSM). The receiver tracks the packet loss status according to the algorithm�o loss rate estimation based on variable frame size and sends the information back to the sender via RTCP. The sender switches two audio sample files in accordance with RTCP report. The proposed algorithm smoothes short term variations in loss rates, while responds quickly to real changes in the loss rate of the traffic. The ability of the algorithm is qualified by the agility, stability and avoidance of frequent switching for two sample files.
摘  要 I
英文摘要 II
誌 謝 III
目 錄 IV
圖 目 錄 VIII
表 目 錄 XII
1. 緒論 1
1.1 研究背景與動機 1
1.2 研究目的 3
1.3 研究步驟 4
1.4 論文架構 5
2. 文獻探討 7
2.1 語音品質評量 7
2.2 即時傳輸協定(REAL-TIME TRANSPORT PROTOCOL, RTP ) 10
2.2.1 RTP 封包表頭格式 12
2.2.2 RTCP 封包表頭格式 19
2.2.2.1 SR 封包協定格式 20
2.2.2.2 RR 封包協定格式 27
2.2.2.3 SDES 封包協定格式 28
2.2.2.4 BYE 封包格式 30
2.2.2.5 APP 封包格式 31
2.3 網路交通量測與估計方法 33
2.3.1 量測指標 33
2.3.2 量測方法 34
2.3.3 估計方式 35
2.3.3.1 Exponentially Weighted Moving Average(EWMA) 36
2.3.3.2 Loss Rate estimation with Fixed frame size(LRF) 36
3. LOSS RATE ESTIMATION WITH VARIABLE FRAME SIZE-以變動框架長度為基礎之遺失率估算法 38
3.1 LRV估計演算法概念 38
3.2 LRV相關名詞解釋 38
3.3 LRV估計之演算法 42
4. 系統設計 47
4.1 開發環境與工具 47
4.1.1 Java簡介 47
4.1.2 JMF簡介 47
4.1.3 WinPcap 與 JPcap 簡介 48
4.2 系統架構 50
4.2.1 RTP傳送端架構 50
4.2.2 RTP接收端架構 55
4.3 系統特色 59
5. 實驗及分析結果 60
5.1 實驗架構環境 60
5.2 實驗與結果 61
5.2.1 實驗一:區域網路QoS實驗 61
5.2.1.1 實驗方式 61
5.2.1.2 實驗結果 63
5.2.2 實驗二:EWMA及LRF之比較 69
5.2.2.1 實驗方式 69
5.2.2.2 實驗結果 70
5.2.3 實驗三:LRV遺失率估計演算法參數實驗 73
5.2.3.1 實驗方式 73
5.2.3.2 實驗結果 74
5.2.4 實驗四:遺失率估計演算法之比較 99
5.2.4.1 實驗方式 100
5.2.4.2 實驗結果 101
5.2.5 實驗五:LRV與EWMA的切換次數比較 102
5.2.5.1 實驗方式 102
5.2.5.2 實驗結果 103
6. 結論與未來展望 105
參考文獻 107
參考文獻

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[20]Java 2 Platform, Standard Edition, http://java.sun.com/j2se/corejava/ index.jsp
[21]Java Media Framework, http://java.sun.com/products/java-media/jmf/
[22]J. Light and A. Bhuvaneshwari, “Performance analysis of audio codecs over Real-Time Transmission Protocol (RTP) for voice services over internet protocol”, Second Annual Conference on Communication Networks and Services Research (CNSR'04), May 2004, pp. 351-356
[23]The Free Packet Capture Library for Windows, http://winpcap.polito.it/
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