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研究生： 蔡世富
Shih-Fu Tsai
論文名稱： 建置以會談啟始通訊協議(SIP)為基礎的VoIP系統之研究
Research of setting up Session Initiation Protocol based VoIP system
指導教授： 黃文吉
Hwang, Wen-Jyi
學位類別： 碩士
Master
系所名稱： 資訊工程學系
Department of Computer Science and Information Engineering
論文出版年： 2006
畢業學年度： 94
語文別： 中文
論文頁數： 78
中文關鍵詞： 會談啟始協議 、網路電話 、視訊會議
英文關鍵詞： Session Initiation Protocol, SIP, VoIP, Video Conference
論文種類： 學術論文
相關次數： 點閱：199  下載：10 
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    • 本論文的目的是建立以會談啟始協議(SIP , Session Initiation Protocol)為基礎的網路電話系統(VoIP , Voice over Internet Protocol)環境之相關研究。一個完整的會談啟始協議網路電話必需包含可供一般使用者操作以會談啟始協議為基礎的使用者端程式，以及可以支援會談啟始協議的伺服器端程式。

    本論文的研究方法，是利用可獲得之合法開放性軟體的原始碼(Open Source)，並加以修改、編譯以建立可支援會談啟始協議的伺服器端系統，進而提供各使用者可以透過會談啟始協議達成建立會談的環境，以及進行多媒體資訊的串流傳輸。

    除此之外為了驗證SIP伺服器端系統是否正常運作，並符合SIP 2.0的通訊協議標準。我們還必須使用由其他組織所發展釋出，同樣以SIP為連線基礎的使用者端網路電話程式，進而與我們所架設的SIP伺服器系統進行註冊、邀請、連線…等動作。

    本論文所架設的SIP伺服器端系統是符合標準的SIP 2.0規範 (定義於RFC-3261 中)，所以無論提出要求的伺服器端程式是如何撰寫而成，只要符合SIP 2.0規範均能正確地登入到伺服器端的使用者資料庫中，並針對其他資料庫中的使用者提出啟始會談的要求。

    而自行架設SIP伺服器端系統的優點，在於可以建立以SIP為基礎的獨立通訊環境，再搭配任何符合SIP 2.0規範的使用者端程式即可建構出一個完整的VoIP系統。除此之外，擁有獨立的SIP伺服器也能輔助實驗室在SIP使用者端程式及其相關應用上的研究。而在設計使用者端程式的流程中，軟體的測試也可以不需考慮其他外在因素的干擾，進而加快系統的開發效率。

    This paper presents a research of setting up a SIP-based (Session Initiation Protocol) VoIP(Voice over IP) system environment. A complete SIP-based VoIP system can be separated into two parts. The first is a SIP-based UA(User Agent) for client to use, the second is a SIP-based server system that can provide SIP message service. The research of this paper is based on using licensed share open source code, with modifying and compiling to set up a server system that can support SIP message service. After setting up a complete server system, any user in this system can create a session with SIP to transmit multimedia streaming data.
    In order to verify the functionality of our SIP system and make sure that it can support standard SIPv2 message flow. We used other SIP-based client program open source that support SIPv2 form different develop group to connect with our server system. We must make sure that the UA can register, invite, and connect… with our SIP server.
    Because we used a standard SIPv2 server system, no matter what programming language the UA used, as long as it can tally with SIPv2 message, we can use this UA to register in our server database and invite any other user in the same system to initiate a session.
    The benefit of setting up a SIP server is to create an independent communication environment. It can support the development of SIP programming in our lab, and the research about the multimedia applications.

    致謝...............................................................................................................................Ⅰ 中文摘要.......................................................................................................................Ⅱ 英文摘要.......................................................................................................................Ⅳ 目錄...............................................................................................................................Ⅴ 第一章 緒論.................................................................................................................1 1.1 研究背景……….............................................................................................1 1.2 研究動機.........................................................................................................3 1.3 研究目的.........................................................................................................4 1.4 全文架構…………………………………………………………………….7 第二章 SIP概論……...................................................................................................8 2.1 SIP會談啟始協議...........................................................................................8 2.2 SIP系統架構...................................................................................................9 2.3 SIP的主要功能.............................................................................................10 2.4 SIP訊息架構及流程.....................................................................................13 第三章 SIP 伺服器系統之實現...............................................................................20 3.1 VOCAL系統.................................................................................................20 3.2 VOCAL架構.................................................................................................21 3.3 VOCAL內部server之運作..........................................................................22 3.4 VOCAL系統之架設說明.............................................................................26 3.4.1 系統環境需求.....................................................................................27 3.4.2 環境安裝.............................................................................................28 3.4.3 VOCAL原始碼之修改.......................................................................30 3.4.4 VOCAL系統安裝...............................................................................32 3.4.5 VOCAL系統之圖形化操作介面.......................................................38 3.4.6 CDR Server之建立.............................................................................41 第四章 SIP使用者代理程式之介紹與安裝.............................................................44 4.1 SIP使用者端程式.......................................................................................44 4.2 SIPSet..........................................................................................................44 4.2.1 SIPSet軟體安裝..................................................................................45 4.2.2 SIPSet操作簡介..................................................................................45 4.3 SIP Communicator........................................................................................46 4.3.1 SIP Communicator軟體安裝..............................................................48 4.3.2 SIP Communicator操作簡介..............................................................50 4.4 Linephone......................................................................................................52 4.4.1 Linephone軟體安裝............................................................................52 4.4.2 Linephone操作簡介............................................................................53 4.5 Jain SIP Phone...............................................................................................55 4.5.1 Jain SIP Phone軟體安裝.....................................................................55 4.5.2 Jain SIP Phone操作簡介.....................................................................55 第五章 測試結果與討論...........................................................................................57 5.1 SIP伺服器系統討論.....................................................................................57 5.1.1 VOCAL系統之操作...........................................................................57 5.1.2 伺服器端之封包分析.........................................................................62 5.2 使用者代理程式的測試...............................................................................64 5.3 SIP Communicator測試環境........................................................................66 5.3.1 SIP Communicator伺服器端連線測試..............................................67 5.3.2 SIP Communicator使用者端連線測試..............................................68 5.3.3 SIP Communicator效能測試..............................................................68 5.4 Linphone測試環境.......................................................................................70 5.4.1 Linphone伺服器端測試結果.............................................................71 5.4.2 Linphone使用者端測試結果.............................................................71 5.4.3 Linphone與SIP Communicator連線測試..........................................72 5.5 Jain SIP Phone測試環境..............................................................................72 5.5.1 Jain SIP Phone伺服器端測試結果.....................................................73 5.5.2 Jain SIP Phone使用者端測試結果.....................................................74 5.6 結論...............................................................................................................74 第六章 未來展望.......................................................................................................76 參考著作.......................................................................................................................77

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    http://www.vovida.org/

    [2] David G. Kelly, Cullen Jennings, Luan Dang, ” Practical VoIP Using VOCAL”, First Edition July 2002.

    [3] J. Rosenberg et al.,“Session Initiation Protocol”, Network Working Group, Request for Comments: 3261, June 2002.
    http://www.ietf.org/rfc/rfc3261.txt

    [4] McGraw-Hill, “SIP Demystified”, Chapter 4 ,”The Session Initiation Protocol:SIP”, McGraw-Hill Companies , Inc. ,2002.

    [5] M. Handley et al. ,” Session Description Protocol”, Network Working Group, Request for Comments: 2327 , April 1998.
    http://www.ietf.org/rfc/rfc2327.txt

    [6] The Apache Software Foundation, Copyright © 1999-2006.
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    [8] H. Schulzrinne et al., ” RTP Profile for Audio and Video Conferences with Minimal Control”, Audio-Video Transport Working Group, Request for Comments: 2327 , January 1996.
    http://www.ietf.org/rfc/rfc1890.txt

    [9] San Antonio Road et al., “Standard JavaTM Interface to the Session Initiation Protocol (SIP)”, 2000 Sun Microsystems, Inc. ,v1.1 - 23rd June 2003.

    [10] San Antonio Road et al., ”JavaTM Media Framework API Guide”,
    Sun Microsystems, Inc., November 19 ,1999.

    [11] Doxygen, “Libosip Reference Manual 0.8.5”, May 17, 2002.

    [12] Speex, a free codec for free speech
    http://www.speex.org/

    [13] Ulf Lamping et al., “Ethereal User's Guide 18189 for Ethereal 0.10.14”, 2004.
    http://www.ethereal.com/docs/eug_html_chunked/

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