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博碩士論文 etd-0904105-093610 詳細資訊
Title page for etd-0904105-093610
論文名稱
Title
建立在NAT環境下使用VoIP服務的負載分享機制
Implement a loading sharing mechanism of using VoIP service behind NAT
系所名稱
Department
畢業學年期
Year, semester
語文別
Language
學位類別
Degree
頁數
Number of pages
54
研究生
Author
指導教授
Advisor
召集委員
Convenor
口試委員
Advisory Committee
口試日期
Date of Exam
2005-07-20
繳交日期
Date of Submission
2005-09-04
關鍵字
Keywords
會議控制、網路位址轉換器、會議初始通訊協定
SIP, NAT, Session Controller
統計
Statistics
本論文已被瀏覽 5718 次,被下載 56
The thesis/dissertation has been browsed 5718 times, has been downloaded 56 times.
中文摘要
本文一開始會先介紹當前VoIP主流的SIP通訊協定,簡介它發展的過程,應用上發展的方向,接著舉例說明SIP通訊協定運作的情形,以例子說明兩個SIP客戶端(SIP Client)是如何透過SIP Proxy建立起通訊連線的,在了解SIP通訊協定的運作之後,說明為什麼在NAT之後的SIP客戶端會遭遇到連線失敗的問題,而又已經有那些方法發展出來以解決這個問題,將各個克服SIP客戶端在NAT之後會遭遇連線失敗的方法簡介說明一下,大略比較一下各個方法,接著說明在採用狀態式SIP Proxy克服SIP客戶端在NAT之後會遭遇連線失敗的方法時,同時也改變了SIP客戶端彼此之間的通訊行為,由端點對端點分散的通訊方式改變成所有端點對伺服器集中的方式之後,可能會遭遇到怎樣影響通話品質的問題,而本文又提出怎樣的方法以改善這個問題,以及如何實作出提出改善問題的方法及環境,最後是測試通話的品質,以及抓取網路上的SIP訊息封包及RTP媒體封包,觀察SIP訊息封包內的內容及RTP媒體封包的流向。
Abstract
At beginning of this paper, it will introduce the SIP protocol major used on the VoIP service at present time. Introduce the development and the application of SIP. And then explain the operation of the SIP protocol by an example. Explain how a SIP client establish the communication with another one by the SIP proxy. After understanding the operation of the SIP, explain why the SIP client behind NAT will meet the fail and introduce the solutions that have developed to conquer the fail. Briefly describe each of these solutions and compare all of them. Next is to describe how the solution of statteful SIP proxy used to avoid the fail of NAT changes the behavior between the SIP clients from end-to-end to master-to-slave. The change of behavior between the SIP clients may affect the quality of the communication in some conditions. To improve this, this paper brings out an idea and implements it. Finally, test the quality of the communication, catch the packet carrying the SIP messages and packet of RTP, and watch the content of SIP messages and the routing of the RTP packet.
目次 Table of Contents
1.緒論 8
1.1. 簡介 8

2. VOIP介紹 9
2.1 VOIP的通訊原理 9
2.2 SIP簡介 9
2.3 SIP運作簡介 11
2.4 SIP在穿越NAT上的問題 15
2.5 SIP解決穿越NAT問題的方法 17
2.5.1 靜態指定(Static assignment) 17
2.5.2 虛擬私有網路(Virtual Private Network) 18
2.5.3 通用隨插即用(Universal Plug and Play) 18
2.5.4 STUN (Simple Traversal of UDP Through NAT) 19
2.5.5 Session Controller 20
2.5.6 各個方法的比較 21

3. 建立在SESSION CONTROLLER環境下的負載分享機制 24
3.1. SESSION CONTROLLER環境下可能遭遇到的問題 24
3.2. MULTI-RTP PROXY 25
3.3. 實作環境 30
3.3.1 SIP Express Router平台介紹 30
3.3.2 Mediaproxy模組中可供設定的參數 31
4.3.3 Mediaproxy模組中可供呼叫的功能 32
3.3.4 SIP Proxy的架設及mediaproxy模組的設定 33
3.3.5 SER設定檔(ser.cfg) 34
3.3.6 RTP Proxy的選擇規則 37
3.3.7 Mediaproxy伺服器的改寫 38
3.3.8 Proxy dispatcher中修改的部分 39
3.4. 觀察MULTI-RTP PROXY運作狀況 41
3.4.1 SIP Proxy及RTP Proxy功能的確認 41
3.4.2 RTP Proxy選擇規則的觀察 42
3.4.3 Mediaproxy伺服器效能的量測 43

4. 問題和改進方式 46
4.1 通話量的快速增加 47
4.2 頻寬的使用率太低 47
4.3 MEDIAPROXY伺服器上網路介面的流量和負載的不平衡 47
4.4 運作MULTI-RTP PROXY的網路環境 47
4.5 改進方式 48

5. 結論與未來工作 49

參考文獻 50
參考文獻 References
[1] B. Goode, “Voice over Internet Protocol (VoIP), ”Proceedings of the IEEE, vol. 90, pp. 1495-1517, Sept. 2002.

[2] P. Srisuresh, M. Holdgree, “IP Network Address Translator (NAT) Terminology and Considerations”, RFC-2663, August 1999.

[3] J. Rosenberg, H. Schulzrinne, G. Camarillo, A. Johnston, J. Peterson, R. Sparks, M. Handley, E. Schooler, “SIP: Session Initiation Protocol”, RFC-3261, June 2002.

[4] Victor Paulsamy, Samir Chatterjee, “Network Convergence and the NAT/Firewall Problems”, IEEE, 2002

[5] H. Schulzrinne, S. Casner, R. Frederick, V. Jacobson, “RTP: A Transport Protocol for Real-Time Applications”, RFC-3550, July 2003.

[6] H. Schulzrinne, A. Rao, R. Lanphier, “Real Time Streaming Protocol (RTSP)”, RFC-2326, April 1998.

[7] M. Arango, A. Dugan, I. Elliott, C. Huitema, S. Pickett, “Media Gateway Control Protocol (MGCP)”, RFC-2715, October 1999.

[8] M. Handley and V. Jacobson, “SDP: Session Description Protocol”, RFC-2327, April 1998.

[9] Y. Rekhter, B. Moskowitz, D. Karrenberg, G. J. de Groot, E. Lear, “Address Allocation for Private Internets”, RFC-1918, February 1996.

[10] B. Gleeson, A. Lin, J. Heinanen, G. Armitage, A. Mails, “A Framework for IP Based Virtual Private Networks”, RFC-2764, February 2000.

[11] UPnP(TM) Forum, http://www.upnp.org

[12] J. Rosenberg, J. Weinberger, C. Huitema, R. Mahy “STUN - Simple Traversal of User Datagram Protocol (UDP)Through Network Address Translators (NATs)”, RFC3489, March 2003.

[13] J. Rosenberg, H. Schulzrinne, “ An Extension to the Session Initiation Protocol (SIP) for Symmetric Response Routing”, RFC 3581, August 2003.

[14] Ya-Lin Huang, Whai-En Chen, Quincy Wu and Yi-Bing Lin “SIP Traversal over NAT Mechanisms on NTP VoIP Platform”, Department of Computer Science and Information Engineering National Chiao Tung University, 2004.

[15] voip-info.org : RTP Symmetric, http://www.voip-info.org/wiki-RTP+Symmetric

[16] cisco.com : VoIP Call Admission Control, http://www.cisco.com/univercd/cc/td/doc/cisintwk/intsolns/voipsol/cac.htm#902188

[17] iptel.org SIP Server : SIP Express Router, http://www.iptel.org/ser/

[18] P. Faltstrom , “E.164 number and DNS”, RFC-2916, September 2000.

[19] M. Day, S. Aggarwal, G. Mohr, J. Vincent, “Instant Messaging / Presence Protocol Requirements”, RFC-2779, February 2000.

[20] P. Saint-Andre, “Extensible Messaging and Presence Protocol (XMPP): Instant Messaging and Presence”, RFC-3921, October 2004.

[21] ag-projects.com : SER MediaProxy, http://www.ag-projects.com/MediaProxy.html

[22] xten.com : X-Lite, http://www.xten.com/index.php?menu=products&smenu=xlite

[23] Python, http://www.python.org/

[24] J. Rosenberg, “Traversal Using Relay NAT (TURN)”, Internet Draft, Internet Engineering Task Force, November 2001, Work in progress.

[25] SIPv6 Analyzer, http://pcs.csie.nctu.edu.tw/~yhsung/sipv6_analyzer/

[26] ag-projects.com : SER MediaProxy, http://mediaproxy.ag-projects.com/INSTALL
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