臺灣博碩士論文加值系統

無線通訊的首要工作，在如何解決移動式端點（Mobile Node，MN）與不同無線存取點（Acess Point，AP)間，進行換手（handover）動作，切換訊號的同時仍可維持語音傳送的穩定與品質。
Session Initation Protocol（SIP），是用來啟始、控制與結束網路中的語音會談協定，同時SIP Mobility機制，是用來達成SIP語音通話的移動性需求。但位於應用層的SIP機制，在不穩定的 WLAN環境中換手時，會因本身重傳機制產生使用者難以接受的延遲，進而造成語音通話的中斷。
本文提出的『改良式換手機制』，主要是透過語音通道建立後，點對點的即時傳輸協定（Real Time Transport Protocol，RTP）中的標頭延伸欄位（Header Extension field），來傳送交換訊息，並運用網際網路控制協定（Internet Control Message Protocol，ICMP）進行不同路徑間的往返時間（Round-Trip Time，RTT）偵測。利用該訊息準確估算最佳換手的時間點，藉此達成平順換手的需求。

The Session Initiation Protocol (SIP) is an application-layer protocol for creating, modifying, and terminating media sessions with one or more participants. However, in the WLAN environment, the mobility management of SIP voice sessions becomes an important issue.
The SIP mobility was proposed for the SIP voice session mobility problem. However, the SIP mobility has the disadvantage of the long SIP signaling delay which significantly degrades the quality of voice communications.
In this paper, we propose and simulate a handover method to reduce the SIP signaling delay. The proposed method requires the correspondent node (CN) to estimate the round-trip times between the correspondent node and different APs. Then, according to the estimated round-trip times, the mobile node switches to the target AP at an appropriate time. The simulation results demonstrate the feasibility of our proposal.

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

第一章　前言 1
第二章　相關通訊協定之研究 3
2.1　會議初始化協定（Session Initation Protocol，SIP）簡介 3
2.2　即時傳輸協定（Real-Time Transport Protocol，RTP）簡介 6
2.3　網際網路控制訊號協定（Internet Control Message Protocol，ICMP）協定介紹 10
2.4　會議初始化協定行動性（SIP Mobility）介紹 12
第三章　以RTP附加資訊達到平順換手的方法 19
3.1　現有SIP架構中的問題 19
3.2　傳統無線網路環境中之換手情形 21
3.3　改良式換手機制 22
3.3.1　WLAN中Layer1/Layer2與Layer2層延遲的改良方法 23
3.3.2　減少網路層DHCP延遲的方法 24
3.3.3　應用層中之改良式換手機制 26
第四章　效能模擬與分析 42
4.1　模擬環境與參數 42
4.2　模擬數據之分析 44
第五章　結論與未來的方向 52
參考文獻 54

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