臺灣博碩士論文加值系統

由於網路的普及以及頻寬的增加，越來越多人利用P2P的方式使用多媒體通訊，而語音通訊更是其中之一，但是在大多數系統以及使用者的使用習慣中，一對一的語音通訊仍然為主要的通訊方式；然而，隨著遊戲產業的發達，越來越多玩家在網路上彼此在遊戲中聊天，因此，我們希望能夠發展出一套讓多人通訊的系統，不會因為NAT的阻隔而無法使用網路語音通訊，並且發展出高容錯性的網路語音通訊系統，讓不論是語音品質或是使用的便利性都能更上一層樓，不只充分利用網際網路，也可以利用本系統拉近現代人的生活。

基於多人語音通訊系統的特性，由於節點進出頻率會較一對一語音通訊頻繁，因此在維持語音品質上的困難度更為提升，但是多人語音通訊系統也同樣具有多條傳輸路徑的性質，本論文利用這個性質結合HyperCube來達到語音容錯的目標，且讓在NAT後的使用者可以透過其他的節點加入多人群組並發送以及接收語音。最後將本系統和線上遊戲系統整合，讓正在玩遊戲的玩家可以利用本系統語音聊天。

Since Internet is growing, more and more people communicate with voice via P2P network. Since the game industry is growing, more and more online game player chat in the game, we hope to develop a group voice communication system without NAT restricted and with high fault tolerance mechanism. The better voice quality and convenience of group voice communication will make people closer.

Due to the nature of group voice communication system, its node transaction frequency is higher than those in one-to-one voice communication. Accordingly, the difficulty of assure its quality is greatly raised. Nevertheless, group voice communication system also has the characteristic of using multi-path transmission. This paper combines this characteristic with Hypercube to increase fault tolerance, and enable users behind NAT to communicate and form groups through different nodes. Finally, this system is integrated with online games in order to let players talk through this system.

摘要 i
誌謝 iii
目錄 iv
表目錄 vi
圖目錄 vii
第一章、 緒論 1
1.1 研究動機與目的 1
論文章節說明 2
第二章、 背景說明 3
2.1相關系統 3
2.1.1 TeamSpeak 3
2.1.2 MSN Messenger 3
2.1.3 Yahoo! Messenger 4
2.1.4 Google Talk 5
2.1.5 Skype 5
2.2 P2P相關技術 6
2.2.1 NAT 6
2.2.1.1 NAT介紹 6
2.2.1.2 NAT的問題 13
2.2.1.3 UDP Hole-Punching 13
2.2.1.4 無法穿越的NAT 16
2.2.1.5 STUN 17
第三章、 系統介紹 18
3.1 網路傳輸 18
3.1.1 語音傳輸方式 18
3.1.2 容錯 18
3.2 網路架構方式 18
3.2.1 以Hypercube為基礎之網路架構 19
3.2.2以Hypercube為基礎之網路架構的問題 19
3.2.3 提出的解決方法 21
3.3 運作方式 24
3.3.1 節點加入 25
3.3.2 節點離開 25
3.3.3 廣播 25
第四章、 系統實作 26
4.1系統架構 26
4.1.1 客戶端系統架構 26
4.1.2 伺服端系統架構 29
4.2 實作問題 30
4.2.1 頻寬問題 30
4.2.2 頻寬問題解決方法 31
4.3實驗結果 33
4.3.1 連線測試 34
4.3.1.1 實驗設計 34
4.3.1.2 實驗結果分析 34
4.3.2 延遲模擬 35
4.3.2.1 實驗設計 35
4.3.2.2 實驗結果分析 36
4.4 實作結果 36
4.4.1 實作環境 36
4.4.1 單人模式下的介面 37
4.4.2 遊戲模式下的介面 37
4.4.3 遊戲模式下之連線以及傳輸狀況 39
第五章、 結論及未來展望 41
5.1 結論 41
5.2 未來展望 41
參考資料 43

[1] TeamSpeak Systems, TeamSpeak, http://www.goteamspeak.com/index.php?page=about, 2004-2007.
[2] Microsoft Corp., MSN Messenger,
http://get.live.com/messenger/overview, 2007.
[3] Yahoo! Taiwan Inc., Yahoo! Messenger,
http://tw.messenger.yahoo.com/features.php, 2007.
[4] Google Inc., Google Talk,
http://www.google.com/talk/intl/zh-TW/about.html, 2005-2006.
[5] Skype Limited, Skype,
http://www.skype.com/, 2003-2006.
[6] Mike Mintz, MSN Messenger Protocol,
http://www.hypothetic.org/docs/msn/, 2002-2004.
[7] Paal E. Engelstad, Geir Egeland, Peer-to-Peer Telephony, January 2006.
[8] MSN Teleo, http://teleo.msn.com/, March 2006.
[9] Yahoo! Messenger, http://en.wikipedia.org/wiki/Yahoo_messenger, July 2007.
[10] K. Egevang, P Francis, RFC1631 The IP Network Address Translator (NAT), May 1994.
[11] P. Srisuresh, K. Egevang, RFC3022 Traditional IP Network Address Translator (Traditional NAT), Jan 2001.
[12] J. Rosenberg, J. Weinberger, C. Huitema, R. Mahy, RFC3489 STUN - Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs), March 2003.
[13] Brayan Ford, Pyda Srisuresh, Dan Kegel, Peer-to-Peer Communication Across Network Address Translators, April 2003.
[14] Saikat Guha and Paul Francis, Simple traversal of UDP through NATs and TCP too (STUNT), http://nutss.gforge.cis.cornell.edu/.
[15] H. Schulzrinne, S. Casner, R. Frederick and V. Jacobson, RFC 1889 - RTP: A Transport Protocol for Real-Time Applications, January 1996.
[16] H. Schulzrinne, S. Casner, R. Frederick and V. Jacobson, RFC 3550 - RTP: A Transport Protocol for Real-Time Applications, July 2003.
[17] Chinmay Padhye, Kenneth J. Christensen and Wilfrido Moreno, A New Adaptive FEC Loss Control Algorithm for Voice Over IP Applications Performance, Computing, and Communications Conference, 2000.
[18] Rajiv Malik，Alka Munjal and Dr. Shampa Chakraverty, Adaptive Forward Error Correction (AFEC) based Streaming using RTSP and RTP Telecommunications, Feburary 2006.
[19] Martin John Lipka, An Analysis of Error Handling Techniques in Voice over IP, 2005.
[20] Scott, D.S., Efficient All-to-All Communication Patterns in Hypercube and Mesh Topologies, 1991.
[21] Patrick T. Gaughan, Sudhakar Yalamanchili, Adaptive routing protocols for hypercube interconnection networks, May 1993.
[22] Yi J. Liang, Nikolaus Färber and Bernd Girod, Adaptive Playout Scheduling and Loss Concealment for Voice Communication Over IP Networks, December 2003.
[23] 賈文康, SIP會談起始協議操典 Session Initiation Protocol(SIP) Methodology Handbook, October 2005.
[24] 汪益賢, 群想網路科技客戶頻寬分析 內部文件, March 2007.