臺灣博碩士論文加值系統

SIP(Session Initiation Protocol)是近幾年網際網路因應多媒體傳輸需求所快速發展出來的通訊協定，由於其結構簡單，彈性以及擴充性極佳，近年來逐漸受到重視與廣泛應用，越來越多的多媒體應用軟體和硬體都以SIP為主要的通訊協定。然而SIP協定在經過防火牆及網路位址轉換時會發生不能通過的狀況，這篇論文會針對這個問題去做分析和解決。
首先，介紹防火牆及網路位址轉換和穿越防火牆及網路位址轉換的問題，和各種穿越方法的介紹。然後對SIP通訊協定做簡介，和介紹我們的VoIP實驗架構和號碼規劃。再來針對SIP網路電話和MSN Messenger和Skype這三種語音通話，做出比較，探討穿越防火牆及NAT的問題。

SIP (Session Initiation Protocol) is the fast-developing protocol in Internet field in recent years. It comes into being as a result of the multi-media transmission request. Because of its simple structure, excellent expansion and flexibility, it is paid more and more emphasis and it is applied extensively. More and more multi-media application software and hardware all make use of SIP as their main protocol. However, when SIP protocol tries to pass through firewall and NAT (Network Address Translation), it might fail. This thesis will put emphasis on this problem and do some analysis to provide remedy.
First, I will introduce the firewall, NAT and the problems about passing through the firewall and NAT. In addtion, I'll introduce a variety of ways on how to pass through firewall. Second, I'll do some concise introduction about SIP protocol, our VoIP experimental structure and numbering planning. Last, I'll compare among SIP Internet Telephone, MSN Messenger and Skype these three voice communication , and probe into how to pass through the firewall and NAT.

第一章 簡介 1
1.1前言 1
1.2論文章節安排 1
第二章 防火牆和網路位址轉換(NAT) 2
2.1防火牆 2
2.2網路位址轉換 4
2.3穿透防火牆及網路位址轉換的問題 5
2.4穿透防火牆及網路位址轉換的方法介紹 5
2.5各種穿越方法比較 7
第三章 SIP及VoIP實驗架構 8
3.1議程初始協定(SIP)通訊協定簡介 8
3.2 SIP訊息介紹 12
3.3 Session建立和結束 15
3.4 VoIP號碼規劃與實驗架構 21
第四章 SIP / MSN /Skype比較分析 24
4.1在SIP探討穿越防火牆及NAT 24
4.2在MSN探討穿越防火牆及NAT 29
4.3在Skype探討穿越防火牆及NAT 37
4.4 SIP/MSN/Skype比較 46
第五章 結論及未來工作 72
Reference 78
Appendix 80

[1]J. Rosenberg, H. Schulzrinne, G. Camarillo, A. Johnston, J. Peterson, R. Sparks, M. Handley, E. Schooler, "SIP: Session Initiation Protocol", RFC 3261, IETF, June 2002.
[2]M. Handley, H. Schulzrinne, E. Schooler, J. Rosenberg, „“ SIP:
Session Initiation Protocol“, Internet Engineering Task Force RFC 2543, June
1999
[3]Handley M, Jacobson V. SDP: Session description protocol. Internet RFC 2327, 1998.
[4]J. Rosenberg J. Weinberger C. Huitema R. Mahy, "STUN - Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs)", RFC 3489, IETF, March 2003.
[5]Rosenberg, J., "Traversal Using Relay NAT (TURN)", draft-rosenberg-midcom-turn-02, March 2003.
[6]UPnP forum, Internet Gateway Device(IGD), V1.0 For Universal Plug and Play Version1.0, http:// www.upnp.org, November 19, 2001
[7]張澍元, “探討SIP通過防火牆與網路位置轉換的方法”, 電腦與通訊雜誌, 第105期, 2003 九月.
[8]陳景章、蘇暉凱、姚志臻、劉柏廷 ，"NAT與Call Center整合服務之實作與測試"，2003年全國電信研討會.
[9]The partysip SIP proxy server http://www.nongnu.org/partysip/partysip.html
[10]SIP Server SIP Express Router http://www.iptel.org/ser/
[11]D. Collins “Carrier Grade Voice Over IP” Second Edition, International Editions 2003 , ISBN 0-07-123115-2
[12]B. Goode, “Voice over Internet Protocol(VoIP)”Proceeding of the IEEE, VOL.90,NO.9 Sep. 2002
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软件学报, Vol.16, No.2
[14]H.Sinnreich , A.B.Johnston ;　"Internet Communications Using SIP" ,
John Wiley & Sons INC , ISBN:0-471-41399-2
[15]Mehta, P. Udani, S. ; ”Voice over IP” Potentials, IEEE , Volume: 20 ,
Issue: 4 , Oct.-Nov. 2001
[16]P. Srisuresh, K. Egevang, “Traditional IP Network Address Translator (
Traditional NAT)”, RFC 3022, January 2001.