臺灣博碩士論文加值系統

VoIP（Voice over IP）係利用分封網路來傳遞傳統中透過電路交換網路傳送的語音資料，以其低建置成本、高應用彈性，成為近來網路應用中的熱門議題，隨著無線區域網路（WLAN）的普及，移動式裝置（Mobile Node，MN）與VoIP語音服務的結合，也架構出語音會談應用的新面貌。
Session Initiation Protocol（SIP）通訊協定，主要是用來建立、終止、以及控制網路上進行的語音會談，為了讓使用者在移動狀態中進行SIP語音會談，有學者提出了SIP Mobility機制，來賦予SIP語音會談更進一步的應用層移動性。但透過SIP機制來解決移動性問題，會因封包的延遲或漏失造成語音通話的中斷，所以如何在語音通話進行中，降低換手過程中的延遲，達成順暢換手的需求，對於通話品質的維持，便顯得十分重要。
本篇論文對於MN在無線訊號接取點（Access Point，AP）間移動並變更IP過程中所產生的延遲提出了改進的方法。同時，在傳統的SIP Mobility機制中，點對點的RTP封包傳輸，需等待SIP 信令繞送完成後才會建立，為了避免因SIP信令繞送的延遲所帶來的語音會談中斷問題，本論文提出了一個使用RTP Header Extension欄位，來即時傳遞IP更新資訊的機制，藉以避免應用層的SIP信令傳輸延遲，達成SIP在WLAN環境中順暢換手的需求。

關鍵字：會談初始協定、即時傳輸協定、無線網路、換手、順暢換手

The Session Initiation Protocol (SIP) is constantly gaining in popularity and acceptance as the signaling protocol for voice communications. However, as the popularity of WLANs and mobile devices, the mobility management of SIP becomes an important issue.
In order to solve the mobility problem, the SIP mobility was proposed. Unfortunately, the SIP mobility has the disadvantage of the long SIP signaling delay that will disturb the voice communication.
In this paper, we propose and simulate a handover method to reduce the SIP signaling delay. The idea is to use the RTP Header Extension field in RTP packets to carry the IP update information. The simulation results demonstrate the feasibility of this proposal.

Keywords：Session Initiation Protocol(SIP), RTP, Wireless Local Area Network(WLAN), handover, seamless handover

封面內頁
簽名頁
授權書 iii
中文摘要 iv
英文摘要 v
誌謝 vi
目錄 vii
圖目錄 ix
表目錄 xi

第一章　序論 1
1.1　研究動機 1
1.2　問題探討 3
1.3　研究目的 5
1.4　論文章節 5
第二章　背景探討 7
2.1　VoIP 7
2.1.1　VoIP系統元件 9
2.1.2　VoIP通訊協定 10
2.2　信令控制協定（Signaling Control Protocol） 11
2.3　RTP（Real-time Transport Protocol，即時傳輸協定） 13
2.4　SIP（Session Initiation Protocol，會談初始協定） 16
2.5　Mobility Support 19
2.6　SIP Mobility 21
第三章　改良式SIP換手機制 29
3.1　系統架構 29
3.2　MAC Layer預先授權機制 30
3.3　預先IP取得機制 30
3.4　Application Layer換手機制 33
3.4.1　RTP Header Extension欄位 34
3.4.2　改良式SIP換手機制 35
第四章　模擬與效能分析 44
4.1　模擬環境 44
4.2　模擬參數 44
4.3　效能分析 46
第五章　結論 52

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