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博碩士論文 etd-0729108-175210 詳細資訊
Title page for etd-0729108-175210
論文名稱
Title
設計與實作一個智慧型SIP終端設備以改進SIP訊息傳遞效率
Design and Implementation of an Intelligent SIP User Agent to Improve Efficiency of SIP Signaling Delivery
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
55
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
2008-07-08
繳交日期
Date of Submission
2008-07-29
關鍵字
Keywords
網路電話、嵌入式系統
gateway, VoIP, SIP
統計
Statistics
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The thesis/dissertation has been browsed 5677 times, has been downloaded 0 times.
中文摘要
因為Skype而聲名大噪的網際網路語音傳輸協定(VoIP)一向被認為是新一代網際網路的殺手級應用,其明顯的成本優勢以及不錯的語音品質有漸漸取代傳統公眾交換電話網路(PSTN)的趨勢,而SIP(Session Initiation Protocol)也正是新世代網路(NGN;Next Generation Network)所建議的語音,多媒體網路控制協定之首選,現今SIP主流的主從式(Client-Server)架構雖然架構簡單、維護容易、擁有比P2P (Peer to Peer)更快的反應時間,但是伴隨著使用者增加,伺服器必定會因為長期負擔龐大的資訊流量而容易發生問題,仰賴伺服器維持的VoIP網路也會因此停止服務。
在這篇論文中,我們設計與實作一個支援SIP通訊協定的嵌入式終端設備,利用嵌入式系統極有限的資源,透過使用者撥打網路電話之特性改善其訊息傳送效率,使撥打網路電話時減少伺服器提供服務的次數,並設計廣為使用者所熟悉的圖形化介面,除了讓使用者更方便使用,也使本系統之方法更有效率。
Abstract
Because of Skype, Voice over IP(VoIP) becomes much more hot and popular. It has always been considered to be a killer application of new generation of the Internet. With its distinct economic advantage and good voice quality, it has the tendency to replace the traditional Public Service Telephone Network(PSTN). Moreover, Session Initiation Protocol(SIP) is proposed by Next Generation Network(NGN) to be the first choice of voice and multimedia network control protocol. Client-server is the key architecture of SIP. Although this architecture is simple and easy to maintain, and even it has faster response time than P2P, the server may cause problems while the client is increasing. Moreover, it is possible that the large load may cause the service stop anytime.
In this paper, we discuss how to design and implement an embedded VoIP user agent based on SIP standard. It can improve efficiency of SIP signaling delivery through the user characteristic of making calls with limited resources and reduce the number of services that servers to provide while calling. We also design a graphical user interface. The interface lets users feel friendly and make the method of the system more efficient.
目次 Table of Contents
摘要........................................................................................................................................... I
ABSTRACT............................................................................................................................II
致謝........................................................................................................................................ III
目次........................................................................................................................................ IV
表目錄..................................................................................................................................VII
圖目錄................................................................................................................................ VIII
CHAPTER 1 導論...................................................................................................................1
1.1、 研究動機......................................................................................................................................1
1.2、 論文架構......................................................................................................................................1
CHAPTER 2 相關研究...........................................................................................................2
2.1、 SIP 介紹.......................................................................................................................................2
2.1.1 SIP 元件介紹.............................................................................................................................5
用戶代理(User Agent) ...................................................................................................................................5
代理伺服器(Proxy Server) .............................................................................................................................5
登錄伺服器(Registrar Server) ........................................................................................................................6
重定向伺服器(Redirect server) ......................................................................................................................6
2.1.2 SIP 訊息....................................................................................................................................7
2.1.2.1 SIP 請求訊息.....................................................................................................................................7
2.1.2.2 SIP 回應訊息.....................................................................................................................................8
2.1.3 SIP Call 運作流程......................................................................................................................8
2.2、 SIP 之P2P..................................................................................................................................12
2.2.1 混合式P2P 系統....................................................................................................................12
2.2.2 純分散式P2P 系統...............................................................................................................14
2.2.3 Distributed Hash Table (DHT) .................................................................................................15
CHAPTER 3 系統架構.........................................................................................................17
3.1、 系統功能....................................................................................................................................17
3.2、 硬體架構....................................................................................................................................17
3.2.1 VINETIC DSP 架構..................................................................................................................19
3.3、 軟體架構....................................................................................................................................20
3.3.1 Sip UA 架構及其運作流程......................................................................................................20
3.3.2 Web 功能及其執行流程...........................................................................................................23
CHAPTER 4 改進SIP 訊息傳遞效率方法之設計.............................................................25
4.1、 系統設計之概念及流程.............................................................................................................25
4.1.1 Out going call ...........................................................................................................................26
4.1.2 In coming call ...........................................................................................................................28
4.2、 P2P CACHE 元件設計..................................................................................................................29
4.2.1 資料結構................................................................................................................................29
4.2.2 P2P Cache 之維護...................................................................................................................29
4.2.2.1 P2P Cache 之搜尋.............................................................................................................................30
4.2.2.2 P2P Cache 之新增.............................................................................................................................31
4.2.2.3 P2P Cache 之刪除.............................................................................................................................32
4.3、 加入FRIEND LIST.........................................................................................................................33
4.3.1 延伸功能................................................................................................................................35
4.4、 無意義IP 之判別........................................................................................................................36
4.4.1 Virtual IP .................................................................................................................................36
4.4.2 通過STUN 穿透NAT 之SIP call...........................................................................................38
4.5、 改進效率之方法可能遇到的問題與解決方案..........................................................................39
4.5.1 無意義之IP ............................................................................................................................39
4.5.2 錯誤之號碼對應.....................................................................................................................40
CHAPTER 5 結論.................................................................................................................42
參考文獻................................................................................................................................43
參考文獻 References
[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] OpenDHT, http://www.opendht.org/
[11] Skype, http://www.skype.com/
[12] SIP Express Router (SER) (http://www.iptel.org/ser/)
[13] The eXtended osip library, http://savannah.nongnu.org/projects/exosip
[14] The GNU oSIP library, http://www.gnu.org/software/osip/
[15] A Network Protocol Analyzer: Wireshark (http://www.wireshark.org/)
[16] “ADM-5120 Data Sheet Rev. 1.1”, Infineon Technologies., March 2005.
[17] “VINETIC-2CPE System Description, v1.1”, Infineon Technologies, March
2006.
[18] “VINETIC-2CPE Device Driver and API Description, v2.1”, Infineon
Technologies. Jan. 2006.
[19] Daniel P. Bovet, Marco Cesati, “Understanding the Linux Kernel, 3/e”, O’Relly,
November 2005.
[20] W. Richard Stevens, Bill Fenner, Andrew M. Rudoff, “Unix Network
Programming: The Sockets Networking API, Vol. 1, 3/e”, Addision Weslsy.
2004.
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