臺灣博碩士論文加值系統

隨著網際網路成長快速，多樣化的網路應用與服務也進入我們的生活中，而VoIP大舉入侵傳統電信市場，消耗掉大量的網路頻寬。在有限的頻寬下，面對大量的資料傳輸，如何讓頻寬做最有效的應用，在近年來成為相當重要的課題，但對於使用私有或加密的通訊協定的VoIP軟體來說，管理者往往無法控管，導致頻寬使用量無法預估或限制。
為了解決上述問題，本研究以Skype為例，主要針對使用私有或加密通訊協定的語音傳輸所作的頻寬管理。此機制是透過Conntrack獲取現有已建立連線的會談，並利用Pcap擷取封包內容並且分析夾帶的資訊，如：傳輸速率、每秒傳輸封包個數、通訊埠…等。利用K-平均法及最近鄰居分類法來進行群聚分析和樣式識別。最後交由流量管理工具配置所需頻寬。
使用此機制可以確保使用VoIP時，具有服務品質保證，且對於使用者的VoIP軟體或硬體並不需要做任何修改或設定，即可達到良好的通話品質。

With the rapid growth of Internet, various applications and information has developed dramatically in the last couple of years. VoIP, being a significant portion of the network traffic today, constitutes a highly desirable class for identification. Accurate classification of proprietary VoIP traffic is a challenging problem, and becomes even more challenging when we are constrained to use only transport-layer header information and encrypted packets.
In this paper, we present a new approach for proprietary VoIP traffic identification that uses fundamental characteristics of proprietary VoIP protocols, such as constant bandwidth consumption and frequent sending rate. We do not use any application specific information and still could identify proprietary VoIP protocols in a simple and efficient way. A bandwidth management system is also built to handle the traffic and guarantee the QoS in bandwidth limited network environment.
Finally, this mechanism is implemented and evaluated on a network processor board. Based on the network traffic of department electrical engineering of Chung Cheng University, the evolution result shows our system can recognize 90% Skype sessions. It could be transplanted to identify other encrypted Internet voice traffic easily.

第一章 緒論 1
1.1 簡介 1
1.2 研究動機 2
1.3 章節大綱 3
第二章 相關研究及背景介紹 4
2.1 VoIP簡介 4
2.1.1 VoIP應用模式 6
2.1.2 VoIP與傳統電信比較 8
2.1.3 VoIP面臨之挑戰 8
2.2 Skype簡介 9
2.2.1 Skype架構及運作方式 11
2.2.2 Skype核心技術 14
2.2.3 Skype優勢與挑戰 17
2.3 服務品質保證 19
2.3.1 頻寬管理機制 20
2.3.2 Linux排程機制 21
2.4 封包過濾與連線分析 26
2.4.1 Netfilter 26
2.4.2 Connection Tracking 28
2.5 網路處理器 31
2.6 群聚分析與樣式辨識技術 33
2.6.1 群聚分析 35
2.6.2 樣式辨識 39
2.7 語音品質評價指標 40
2.7.1 平均意見分數 40
2.7.2 E-Model 42
第三章 系統設計與實作 46
3.1 系統架構 46
3.2 實作環境 48
3.3 實作方法 48
3.4 Skype行為分析 49
3.4.1 Skype會談特性 49
3.4.2 Skype會談辨識 51
3.5 系統模組實作 54
3.5.1 會談分析模組實作 54
3.5.2 頻寬配置模組實作 58
3.5.3 流量控制指令模組實作 59
3.5.4 Admin UI模組實作 60
3.5.5 整體運行架構 62
第四章 系統量測與驗證 64
4.1 測試平台與設備 64
4.2 辨識成必v分析 65
4.3 網路品質分析 67
4.4 語音品質評量分析 68
4.5 系統延展性 71
第五章 結論與未來展望 72
5.1 結論 72
5.2 未來展望 73
參考文獻 74

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