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博碩士論文 etd-0726106-135454 詳細資訊
Title page for etd-0726106-135454
論文名稱
Title
Web與網路電話系統介接之設計與實作
The Design and Implementation of Integration of Web Page with VoIP System
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
73
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
2006-07-14
繳交日期
Date of Submission
2006-07-26
關鍵字
Keywords
會談啟始通訊協定、電話、網頁、網路電話
VoIP, Web, Phone, SIP
統計
Statistics
本論文已被瀏覽 5647 次,被下載 12
The thesis/dissertation has been browsed 5647 times, has been downloaded 12 times.
中文摘要
網際網路的應用為人類帶來相當多的便利,在網路電話的使用上就是一個良好的例子。基於網路使用的普及,人類能利用網路電話遠距溝通,更由於網路電話無通話費用的限制,可結合傳統的電話交換機系統應用,在機關團體使用上成功降低公司在對內、對外時聯繫的成本,並衍伸出另一種提供客戶及使用者免付費的諮詢方式。
利用和公司內部的網路電話系統建立連結,內部人員或網際網路上的參觀者及臨時性的訪客可即時性的發問、或諮詢。但是,在建立網路電話連線時,網路電話系需要許多繁雜的安裝設定步驟,諸如:註冊帳號及密碼、配置網路電話伺服器位址、通訊埠、通用協定、加密、對外的OutBound Proxy等,每家公司機關團體設定相互差異,每次聯繫需重覆建立操作,並下載不同的客戶端軟體、閱讀安裝說明文件等繁瑣且複雜,對於一般性訪客及不熟悉操作的使用者來說,更是增加對方的困擾並降低使用的興趣,網路的便利與即時性則大打折扣。
有鑑於此,本論文便是希望利用Web結合網路電話,利用Web使用的普遍性與便利性,讓訪客在網頁上瀏覽相關聯絡資訊時,便能透過點擊在網頁上的按扭觸發網頁元件,達到建立網路電話通訊,立即在網路上通話。同時,可避免將系統伺服器位址、帳號密碼等相關資訊曝露在公開場合之中,以必免不必要的安全性問題發生。
Abstract
It is very convenient for human to use Internet for communications. The VoIP service is a good example. Because voice transmission through Internet becomes mature, people can remotely talk with each other by IP phones inexpensively. Thus, VoIP system can be integrated with Public Switched Telephone Network (PSTN) in some groups, organizations, or companies to reduce the cost for communication. Additionally, it also can be another kind of free consultation channel for customers and users.
To take advantage of VoIP system, the people who may be inside a company, be the Internet visitors, or be provisional guests can receive services and question answers immediately and freely. However, it needs a lot of procedures to connect to the VoIP system, e.g., to install a softphone which must be booted to register an VoIP server, to even configure IP address, port number, protocol, encryption, and outbound proxy, etc. This configuration sometimes is difficult and it may need to setup every time when customers want use VoIP phone. It is not an easy job for a user, for example, to download the software from Internet, and to read some documents to setup system parameters and then to operate it, especially for those people who are not familiar with computers. Thus, this inconvenience may cause VoIP service not to be easily promoted.
Therefore, the purpose of this paper is to solve this problem and make users have a web interface and without worrying about how to setup the system. I combine Web interface with VoIP phone to become a “webphone”. To take advantage of the characteristics of generality and facility of Web, guests can click the button on the Web pages to trigger the VoIP component inside the Web page to connect to the VoIP system, then make communication with other people. At the same time, it can avoid exposing the information of server address, account, password to public and ward off dangerous attacks from Internet.
目次 Table of Contents
1、導論 7
1.1、研究動機 7
1.2、論文架構 8
2、相關研究 9
2.1、相關背景 9
2.1.1、VoIP 9
2.1.2、SIP協定 9
2.1.3、SIP UA 11
2.2、相關客戶端軟體 15
2.1.1、sipXphone 16
2.1.2、sipXezPhone – (續sipXphone) 18
2.1.3、X-Lite 19
2.1.4、SJPhone 21
2.3、相關技術 23
2.3.1、SIP函式庫: 23
3、系統架構 26
3.1系統設計考量 26
3.2 系統架構 26
3.3系統運作流程 28
3.3.1、ActiveX載入 28
3.3.2、註冊 29
3.3.3、建立通話 29
3.3.4、結束通話 30
3.3.5、接聽電話 31
4、實作平台及方法 33
4.1、發展平台及工具 33
4.2、軟體架構 33
4.2.1、SIP程式 34
4.2.2、實作流程 37
4.2.3、ActiveX與瀏覽器的溝通 45
5、效能分析及評估 49
5.1、程式大小比較: 49
5.2、程式載入所需時間: 50
5.3、程式載入之最高CPU使用率 51
5.4、實作成果展示 53
6、結論及未來發展方向 58
6.1、結論 58
6.2、未來發展 58
REFERENCES 60
APPENDIX A. 實作之補充 61
補充一:IRTCEVENTNOTIFICTION實作 61
補充二:CCONTAINWINDOW類別 66
補充三:實作安全性介面 68
APPENDIX B. 狀態變化對應圖 71
參考文獻 References
[1] U. D. Black, Voice Over IP, 2nd Edition, Prentice Hall, Jan. 2002.
[2] Microsoft, “RTC Client API,” http://msdn.microsoft.com/library/default.asp?url=/library/en-us/rtcclnt/rtc/real_time_communications_rtc_client_start_page.asp.
[3] SIPFoundry, “SIPfoundry Open Source VoIP Solutions,”http://www.sipfoundry.org/use.html.
[4] Xten, “Xten products download,” http://www.xten.com/index.php?=menu=products&smenu=download.
[5] SJ Labs, “Basic softphone for PC,” http://www.sjlabs.com/sjp.html.
[6] OpenH323, “Open Phone Abstraction Library,” http://www.openh323.org/.
[7] Vovida, “Vovida Open Communications Applications Library,” http://www.vovida.org/applications/.
[8] GNU, “GNU oSIP library,” http://www.gnu.org/software/osip/
[9] CSDN, “幾種開源SIP協議棧對比, ” http://blog.csdn.net/wffy/archive/2004/10/07/127003.aspx.
[10] IEIF, “SIP: Session Initiation Protocol,” http://www.ietf.org/rfc/rfc2543.txt, Mar. 1999.
[11] IEIF, “SIP: Session Initiation Protocol,” http://www.ietf.org/rfc/rfc3261.txt, Jun. 2002.
[12] IEIF, “Reliability of Provisional Responses in Session Initiation Protocol (SIP),” http://www.ietf.org/rfc/rfc3262.txt, Jun. 2002.
[13] IEIF, “Session Initiation Protocol (SIP): Locating SIP Servers,” http://www.ietf.org/rfc/rfc3263.txt, Jun. 2002.
[14] IEIF, “An Offer/Answer Model with Session Description Protocol (SDP),” http://www.ietf.org/rfc/rfc3263.txt, Jun. 2002.
[15] IEIF, “Session Initiation Protocol (SIP)-Specific Event Notification,” http://www.ietf.org/rfc/rfc3265.txt, Jun. 2002.
[16] IEIF, “The Session Initiation Protocol (SIP) Refer Method,” http://www.ietf.org/rfc/rfc3515.txt, Apr. 2002.
[17] IEIF, “The Session Initiation Protocol (SIP) UPDATE Method,” http://www.ietf.org/rfc/rfc3311.txt, Oct. 2002.
[18] IEIF, “The SIP INFO Method,” http://www.ietf.org/rfc/rfc2976.txt, Oct. 2000.
[19] IEIF, “The Reason Header Field for the Session Initiation Protocol (SIP),” http://www.ietf.org/rfc/rfc3326.txt, Dec. 2002.
[20] IEIF, “SDP: Session Description Protocol,” http://www.ietf.org/rfc/rfc3326.txt, Dec. 2002.
[21] H. Schulzrinne and J. Rosenberg,“The IETF Internet telephony architecture and protocols,” IEEE Network, pp.18-23, May/June 1999. [1] U. D. Black, Voice Over IP, 2nd Edition, Prentice Hall, Jan. 2002.
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