臺灣博碩士論文加值系統

壅塞控制是網路管理的重大問題。目前的網路應用程式多半使用TCP或UDP這兩個傳輸協議來傳遞資料。TCP具有壅塞控制機制可以隨著網路狀況調整資料傳送速率，但其重送機制所導致的時間延遲不利於時效性的網路服務。而UDP的傳輸速率多半是在傳輸之前先行設定，在資料的傳輸過程中不再改變，對於網路壅塞並無任何調節作用，不利於網路之和諧共用。因此DCCP 這種具有壅塞控制機制的不可靠傳輸協議被提出，期望取代UDP 成為不可靠傳輸的主流協議。網路電話大部分使用UDP作為傳輸層協定，UDP不具壅塞控制機制，有礙於網路之和諧共用，而且網路的品質也會因為網路壅塞而遺失封包導致品質受損，要達成網路電話在網路壅塞時有壅塞控制能力，且能有好的通話品質，必須要根據網路的狀況調整封包傳送速度或是封包大小。
本研究利用多層編碼配合DCCP形成與TCP友善的網路電話。壅塞控制必須針對網路中不同程度的壅塞作出適當程度的反應，因此我們改進語音編解碼器，設計出多層語音編碼並設計出可以與之配合的DCCP，讓DCCP依照網路的狀況送出不同層級的語音封包。本研究透過實際網路的實驗環境中評估以CBR over UDP、Flexible Bit-Rate以及Scalable Codec三種方式傳輸網路電話封包的效能。並也評估Scalable Codec VoIP與TCP同時存在於頻寬不足的網路中，對於頻寬競爭能力的表現。實驗結果顯示在網路壅塞最嚴重的情況下可以和一般的CBR配合UDP的方法比較封包遺失率達到40%左右的改善，語音品質評估指標MOS達到1.5分的改善，與Flexible Bit-Rate方法比較封包遺失率也達到8%以上的改善，MOS達到1分的改善，讓整體網路狀況更為穩定，並因為封包遺失的減少讓語音品質提升。而在和TCP頻寬競爭實驗中可看出，在頻寬不足的情況下，此研究提出的方法可以和TCP友善且公平的競爭頻寬。

Congestion control is one of the major problems of network management. Most current network applications use either TCP or UDP to transport data. TCP is equipped with a congestion control mechanism but is not suitable for real-time multimedia applications due to its instability of delay time. On the other hand, UDP fixes its data rate and doesn't change it during the period of transmission even when the network is congested. Under this circumstance, DCCP, which is an unreliable transport protocol but has a congestion control mechanism, is proposed to replace UDP to support real-time network applications such as VoIP. Our previous study showed that a flexible bit rate CODEC to support VoIP over DCCP can effectively control network congestions while maintaining a good voice quality. However, it has an implementation issue yet to be addressed: it requires a bidirectional interaction between DCCP and CODEC.
This thesis proposes to use a scalable CODEC approach to support flexible bit rate VoIP over DCCP. The CODEC sends the entire spectrum of input voice stream to DCCP. DCCP then selects the appropriate voice activation level to compose output stream according to the measured network status, which is feedbacked from the receiver side. The interaction between DCCP and CODEC is avoid. The proposed scheme was evaluated in a real local area network against two other protocols under various VoIP environments, CBR over UDP and Flexible Bit-Rate. Experimental results show that the proposed scheme can outperform CBR over VoIP in the most serious network congestion (under our lab configuration) by 40% in packet loss rate and 1.5 in MOS. It can outperform Flexible Bit-rate over VoIP by 8% in packet loss rate and 1.0 in MOS. Finally,the fairness test shows that our scheme can coexist with TCP with a fairness index higher than 95% even when network is congested.

第一章 緒論 1
1.1 DCCP 2
1.2 VoIP over DCCP 3
1.3 VoIP with Flexible Bit-rate 5
1.4 Scalabe Codec over DCCP 5
1.5 研究目的與方法 6
1.6 論文結構 7
第二章 背景與相關研究 8
2.1 網路壅塞 8
2.2 壅塞控制 9
2.3 TCP壅塞控制機制 10
2.4 DCCP壅塞控制機制 12
2.4.1 CCID 2: TCP-Like 13
2.4.2 CCID 3: TFRC (TCP-Friendly Rate Control) 13
2.5 網路電話 14
2.5.1聲音取樣以及編碼 15
2.5.2語音封包 16
2.5.3影響語音品質的因素 17
2.6 網路語音品質評量指標 17
2.6.1 Mean Opinion Score 17
2.6.2 E-Model 18
2.7 具壅塞控制功能之VoIP的相關研究 20
第三章 Scalable Codec over DCCP 24
3.1需求分析與設計目標 24
3.2 Scalable Codec over DCCP 基本觀念 26
3.3 Speex 26
3.4 Scalable Codec設計 29
3.5 演算法設計 32
3.5.1 壅塞偵測 32
3.5.2 High Delay Response 演算法 36
3.5.3 High Loss Response演算法 39
3.5.4 Low Delay Response演算法 41
第四章 實驗與效能評估 43
4.1評估指標 43
4.2實驗環境 43
4.3實驗一 44
4.3.1 CBR over UDP VoIP實驗結果與分析 44
4.3.2 Flexible Bit-rate VoIP實驗結果與分析 50
4.3.3 Scalable Codec VoIP實驗結果與分析 55
4.3.4 三種傳輸方式實驗結果與分析 60
4.4實驗二 62
4.4.1 CBR over UDP VoIP vs TCＰ實驗結果與分析 62
4.4.2 Scalable Codec VoIP vs TCP實驗結果與分析 66
4.4.3 Scalable Codec VoIP之間的頻寬競爭實驗結果與分析 70
4.4.4 頻寬競爭能力實驗結果與分析 73
第五章 結論與未來研究 79
參考文獻 80

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