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標題: 以SIP為基礎之語音導覽系統設計與實作
A Design and Implementation of a SIP-based Voice-Guiding System
作者: 陳維廉
Chen, Wei-Lien
關鍵字: SIP
出版社: 資訊科學與工程學系所
引用: [1] M. Handley, H. Schulzrinne, E. Schooler, J. Rosenberg, “SIP: Session Initiation Protocol”, IETF RFC 2543, March 1999. [2] J. Rosenberg, H. Schulzrinne, G. Camarillo, A. Johnston, J. Peterson, R. Sparks, M. Handley, E. Schooler, "SIP: Session Initiation Protocol", IETF RFC 3261, June 2002. [3] J. Rosenberg, H. Schulzrinne, “Reliability of Provisional Responses in the Session Initiation Protocol (SIP)”, IETF RFC 3262, June 2002. [4] J. Rosenberg, H. Schulzrinne, “Session Initiation Protocol (SIP): Locating SIP Servers”, IETF RFC 3263, June 2002. [5] J. Rosenberg , H. Schulzrinne , “An Offer/Answer Model with the Session Description Protocol (SDP)”, IETF RFC 3264, June 2002. [6] A. B. Roach, “Session Initiation Protocol (SIP)-Specific Event Notification”, IETF RFC 3265, June 2002. [7] M. Handley, V. Jacobson, "SDP: Session Description Protocol", RFC 2327, April 1998. [8] H. Schulzrinne, S. Casner, R. Frederick, V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", IETF RFC 3550, July 2003. [9] R. Rivest, "The MD5 Message-Digest Algorithm", IETF RFC 1321, April 1992. [10] J. Franks, P. Hallam-Baker, J. Hostetler, S. Lawrence, P. Leach, A. Luotonen, L. Stewart, "HTTP Authentication: Basic and Digest Access Authentication", IETF RFC 2617, June 1999. [11] D. Eastlake, S. Crocker, J. Schiller, "Randomness Recommendations for Security", IETF RFC 1750, December 1994. [12] M. Handley, V. Jacobson, "SDP: Session Description Protocol", IETF RFC 2327, April 1998. [13] A. Johnston, S. Donovan, R. Sparks, C. Cunningham, K. Summers, "Session Initiation Protocol (SIP) Basic Call Flow Examples", IETF RFC 3665, December 2003. [14] A. Johnston, R. Sparks, C. Cunningham, S. Donovan, K. Summers, "Session Initiation Protocol Service Examples", IETF RFC 5359, October 2008. [15] E1 Sawda,S. E1 Sqwda,R. Urien,P. Haiieh, "Non Repudiation for SIP Protocol - SIP Sign", Proceedings of International Conference on Information and Communication Technology: From Theory to Applications (ICTTA), pp. 1-5, April 2008. [16] M. Spencer, B. Capouch, E. Guy, F. Miller, K. Shumard, "IAX: Inter-Asterisk eXchange Version 2", IETF RFC 5456, February 2009 [17] C. Krishna Sumanth, J. Ilow, "Integration of Open Source and Enterprise IP PBXs", Proceedings of International Conference on Testbeds and Research Infrastructure for the Development of Networks and Communities, pp. 1-6, May 2007. [18] T.Abbasi, S.Prasad, N.Seddigh, I. Lambadaris, "A comparative study of the SIP and IAX VoIP protocols", Proceedings of Canadian Confernce on Electrical and Computer Engineering, pp. 179-183, May 2005. [19] M.Z. Alam, S. Bose, M. M.Rahman, M. Abdullah Al-Mumin, "Small Office PBX Using Voice Over Internet Protocol (VOIP)", Proceedings of International Conference on Advanced Communication Technology, pp. 1618-1622, Feb. 2007. [20] A. Imran, M.A. Qadeer, "Conferencing, Paging, Voice Mailing via Asterisk EPBX", Proceedings of International Conference on Computer Engineering and Technology (ICCET), pp. 186-190, Jan. 2009. [21] M.A. Qadeer, A. Imran, "Asterisk Voice Exchange: An Alternative to Conventional EPBX", Proceedings of International Conference on Computer and Electrical Engineering (ICCEE), pp. 652-656, Dec. 2008. [22] K.L. Larsen, G. Castro, H.-P.Schwefel, V.S. di Carlo, "Corporate Convergence with the 3GPP IP Multimedia Subsystem", Proceedings of International Conference on Next Generation Mobile Applications, Services and Technologies (NGMAST), pp. 29-35 , Sept. 2007. [23] Leif Madsen , Jared Smith, Jim Van Meggelen, “The Future of Telephony”, O’Reilly, ISBN: 0-596-00962-3., September 2005. [24] K. Singh, G. Nair and H. Schulzrinne, “Centralized Conferencing using SIP”, Proceedings of the 2nd IP-Telephony Workshop (IPTel), April 2001. [25] D. Song, Y. Mo and F. Wang, “Architecture of Multiparty Conferencing using SIP”, Proceedings of International Conference on Wireless Communications, Networking and Mobile Computing(WCNM), pp. 1361-1364, Sept. 2005. [26] Jianyun Ni, Jing Luo, “Design of Multimedia Conference Control System based on SIP”, Eighth ACIS International Conference on Software Engineering, Artificial Intelligence, Networking , and Parallel/Distributed Computing (SNPD), pp. 810-814, Aug. 2007. [27] 林宏熒,"從語音到影音:談博物館如何規劃掌上型數位導覽",博物館學季刊 第20卷第1期,台中市,2006。 [28] 賈文康,"Session Initiation Protocol Methodology Handbook Second Edition",文魁資訊,台北市,2008。 [29] Asterisk, [30] VOIP Wiki, [31] SIPfoundry, [32] SIPXecs Wiki, [33] X-Lite, [34] Asterisk Dialplan Commands, Asterisk+-+documentation+of+app lication+commands [35] Asterisk variables, variables#InheritanceofChannelVariables [36] Asterisk Manager API, API [37] Asterisk Manaer:Events, asterisk+manager+ events
摘要: 隨著網路的發展與普及,網路電話(VoIP)亦日趨受重視。網路電話早在多年前遂已成型,但因早期網路頻寬不足,使得網路電話發展受阻,如今隨著網路頻寬的增長,各電信廠商已大力投入網路電話的開發。 本論文將利用SIP網路電話技術,實作一個展覽館的無線語音導覽系統,以無線網路話機(WiFi-phone/SIP phone)為導覽設備,利用類似撥打電話方式聽取導覽語音;由櫃檯以網頁介面將導覽手機租借給參觀觀眾,觀眾取得導覽手機後,以手機撥打展品所標示之展品編號,再透過IP交換機(IP Private Branch Exchange,IP-PBX)轉接至互動語音系統IVR (Interactive Voice Response),互動語音系統接受客戶輸入展品編號,並查詢資料庫取得指定語言(國語、台語、英語…等)及展品語音檔,進而播放該展品導覽語音。 我們已成功實作並驗證我們的系統,由實驗結果顯示我們的系統可以成功的利用SIP技術達到無線語音導覽功能。
With the incredible development of the network technologies and the rapid increase of network bandwidth, Voice over Internet Protocol (VoIP) is a promising technology that attracts lots people's attention. Many telecommunication companies have also engaged in the development of VoIP products and applications. In this thesis, we utilize the SIP phone technology to implement a mobile voice-guiding in exhibition halls or museums. Firstly, we implement a guiding device that uses the IEEE 802.11 wireless network as the transmission media and utilizes the SIP technology as the call signaling mechanism to setup a call connection. Consequently, requesting the guiding voice data is similar to dial-up a telephone call. The users can dial the number of each exhibition by his/her guiding device. Then, through the IP-PBX (IP Private Branch Exchange) device, a SIP signaling connection is setup between the guiding device and the IVR (Interactive Voice Systems) systems. Once the IVR receives the number, it accesses the database server to request the voice-guiding audio file according to the languages selected by visitors. Finally, the audio-file is transmitted to and played by the guiding device. Notably, to facilitate the management of guiding devices, we also implement a web-based software for renting and returning the guiding devices. We have implement a prototype system and show that our system can work fine as well. The result shows that SIP technology can be successfully applied in the mobile voice guiding systems.
其他識別: U0005-1808200912523400
Appears in Collections:資訊科學與工程學系所



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