VoIP（Voice-over-IP）以Internet傳送語音資料，相較於傳統電信的PSTN（Public Switched Telephone Network）網路，有節省通話和設備費用等好處，已成為網路應用新寵，而SIP（Session Initiation Protocol）挾帶方法簡單、擴充性佳和開放協議的優勢，儼然成為VoIP研究主流。現今SIP使用的主從式架構（Client-Server），雖然架構簡單、維護容易，但是卻有服務品質的優劣需仰賴伺服器性能之缺陷，為改善此缺陷，IETF（The Internet Engineering Task Force）草案討論了將P2P（Peer-to-Peer）架構應用於SIP的議題，希望藉由P2P架構，良好的容錯性和傳輸效能，提供良好SIP服務品質。
本論文以實做P2PSIP架構於嵌入式用戶端裝置（User Agent）為目的。首先，分析各種P2P架構優劣，探討P2P架構帶來的搜尋問題；其次，考量到嵌入式系統十分有限的資源，提出運用Cache的實際可行方案；最後，再對實做上面臨到的問題，作討論和解決。

Instead of using the public switched telephone network, VoIP (Voice-over-IP) services exchange voice information over Internet. Therefore, VoIP can services with their advantage of low rates. SIP (Session Initiation Protocol) is scalable, easy to implement, and requires less setup time than its predecessor protocols. Therefore, SIP becomes one of the major signaling protocols used in VoIP. SIP uses Client-Server architecture which is simple and easy to maintain. But Client-Server architecture may cause problems while the clients increase. For the reason, IETF (The Internet Engineering Task Force) drafts discusses using P2P (Peer-to-Peer) architecture in SIP protocol to avoid the weaknesses of Client-Server architecture, and hopes to provide good voice quality by using P2P architecture.
In this paper, we discuss how to design and implement an embedded P2PSIP(Peer-to-Peer Session Initiation Protocol) user agent based on SIP standard. First, we analyze all kind of the major P2P structures, and discuss the advantage and disadvantage of them. Second, our design needs to consider the embedded systems which have limited resources. Using cache is a good solution. Finally, we confront the problem and try to solve it.

致謝 I
摘要 II
Abstract III
目錄 IV
圖目錄 VI
表目錄 VIII
1. 導論 - 1 -
1.1. 研究動機 - 1 -
1.2. 論文架構 - 2 -
2. 相關研究 - 3 -
2.1. P2P介紹 - 3 -
2.1.1. P2P簡介 - 3 -
2.1.2. P2P分類 - 3 -
2.1.2.1. 中央集權式P2P網路（Centralized P2P Network） - 4 -
2.1.2.2. 分散式P2P網路（Decentralized P2P Network） - 5 -
2.1.2.3. 混合式P2P網路（Hybrid P2P network） - 5 -
2.1.2.4. 非結構化P2P系統（Unstructured P2P Network） - 6 -
2.1.2.5. 結構化P2P網路（Structured P2P Network） - 7 -
2.1.3. P2P搜尋方法 - 8 -
2.1.3.1. 洪氾式廣播（Flooding） - 8 -
2.1.3.2. Modified-BFS方法 - 9 -
2.1.3.3. Iterative Deepening搜索方法 - 9 -
2.1.3.4. 隨機遊走（Random Walk） - 10 -
2.1.3.5. 外分支度（Out Degree）與內分支度（In Degree）的管理 - 10 -
2.1.3.6. Gnutella2的搜索方法 - 10 -
2.1.3.7. 以興趣偏好來建立覆蓋式網路 - 11 -
2.1.3.8. 分散式雜湊表（Distributed Hash Table, DHT） - 11 -
2.1.4. P2P SIP - 14 -
2.2. SIP介紹 - 16 -
2.2.1. 用戶代理（User Agent） - 19 -
2.2.2. 代理伺服器（Proxy Server） - 19 -
2.2.3. 登錄伺服器（Registrar Server） - 20 -
2.2.4. 重定向伺服器（Redirect server） - 20 -
2.2.5. SIP訊息 - 21 -
2.2.5.1. SIP請求訊息 - 21 -
2.2.5.2. SIP回應訊息 - 21 -
2.2.6. SIP Call運作流程 - 22 -
3. 實驗環境 - 25 -
3.1. 嵌入式裝置功能 - 25 -
3.2. 硬體架構 - 25 -
3.2.1. VINETIC DSP架構 - 26 -
3.3. 軟體架構 - 27 -
3.3.1. Sip UA架構及其運作流程 - 28 -
4. 系統設計與實做 - 31 -
4.1. 系統架構 - 31 -
4.2. 伺服器端設計 - 32 -
4.2.1. Server 運作流程 - 32 -
4.2.2. 重新取得設定檔Re-configuration - 34 -
4.3. 使用者端設計 - 35 -
4.3.1. Call Forward - 37 -
4.3.2. Out Going Call - 38 -
4.3.3. In Coming Call - 40 -
4.4. P2P Cache元件設計 - 41 -
4.4.1. 資料結構 - 41 -
4.4.2. P2P Cache之搜尋 - 42 -
4.4.3. P2P Cache之新增 - 43 -
4.4.4. P2P Cache之刪除 - 44 -
4.5. 傳遞效能 - 45 -
4.6. 問題與解決方案 - 52 -
4.6.1. UA Change IP or Offline - 53 -
4.6.2. Virtual IP - 53 -
5. 結論 - 54 -
參考文獻 - 55 -

[1] M. Handley, V. Jacobson, “SDP: Session Description Protocol”, RFC-2327, April 1998.
[2] M. Handley, H. Schulzrinne, E. Schooler and J.Rosenberg, "SIP: Session Initiation Protocol", RFC-2543, March 1999.
[3] San Jose, CA, David A. Bryan, “Peer-to-Peer SIP”, Invited Talk at Cisco Systems, July 2005.
[4] J. Rosenberg, H. Schulzrinne, G. Camarillo, A. Johnston, J. Peterson, R. Sparks, M. Handley, E. Schooler, “SIP: Session Initiation Protocol”, RFC-3261, June 2002.
[5] J. Rosenberg, J. Weinberger, C. Huitema, and R. Mahy. STUN - simple traversal of user datagram protocol (UDP) through network address translators (nats). RFC 3489, Internet Engineering Task Force, March. 2003.
[6] J. Rosenberg, H. Schulzrinne, “An Extension to the Session Initiation Protocol (SIP) for Symmetric Response Routing”, RFC-3581, August 2003.
[7] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, “RTP: A Transport Protocol for Real-Time Applications”, RFC-3550, July 2003.
[8] H. Schulzrinne, S. Petrack, “RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals”, RFC-2833, May 2000.
[9] Henry Sinnreich, Alan Johnston, “SIP, P2P, and Internet Communications”, Internet-Draft draft-johnston-sipping-p2p-ipcom-02, IETF, March 2006.
[10] p2psip.org, http://www.p2psip.org/
[11] OpenDHT, http://www.opendht.org/
[12] Skype, http://www.skype.com/
[13] The eXtended osip library, http://savannah.nongnu.org/projects/exosip
[14] The GNU oSIP library, http://www.gnu.org/software/osip/
[15] OpenSIPS Project, http://www.opensips.org/
[16] A Network Protocol Analyzer: Wireshark (http://www.wireshark.org/)
[17] “ADM-5120 Data Sheet Rev. 1.1”, Infineon Technologies., March 2005.
[18] “VINETIC-2CPE System Description, v1.1”, Infineon Technologies, March 2006.
[19] “VINETIC-2CPE Device Driver and API Description, v2.1”, Infineon Technologies. Jan. 2006.
[20] Daniel P. Bovet, Marco Cesati, “Understanding the Linux Kernel, 3/e”, O’Relly, November 2005.
[21] W. Richard Stevens, Bill Fenner, Andrew M. Rudoff, “Unix Network Programming: The Sockets Networking API, Vol. 1, 3/e”, Addision Weslsy. 2004.