研究生: |
陳俞安 Yu-an Chen |
---|---|
論文名稱: |
採用SIP之網路語音廣播系統 Network Audio Broadcasting System Based on SIP |
指導教授: |
邱炳樟
Bin-chang Chieu |
口試委員: |
柳宗禹
Tzong-yeu Leou 許新添 Hsin-teng Hsu |
學位類別: |
碩士 Master |
系所名稱: |
電資學院 - 電子工程系 Department of Electronic and Computer Engineering |
論文出版年: | 2008 |
畢業學年度: | 96 |
語文別: | 中文 |
論文頁數: | 72 |
中文關鍵詞: | 語音廣播系統 、SIP |
外文關鍵詞: | Audio Broadcasting System, SIP |
相關次數: | 點閱:240 下載:3 |
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在許多環境中如校園、工廠或是公共場所等,都可能會需要使用語音廣播系統,利用語音告知此環境下的人某些資訊。鑑於傳統語音廣播系統擴充性低、建構維護費用高等諸多缺點,因此本論文中提出一個以SIP為基礎的網路語音廣播系統。SIP是一個高擴充性的會議控制層協定,只負責傳送建立、更改或是結束媒體會議的控制訊息,另外需要配合SDP和RTP/RTCP等協定進行媒體的溝通及語音的傳輸。
本系統架設在現有的網路環境底下,主要由SIP代理伺服器、廣播端和多個播放端所組成。在伺服器端部份採用模組化架購,也可以自行定義訊息處理的流程;廣播端的程式是使用跨平台且開放原始碼的函式庫開發,在不用更改程式碼的情況下就能移植到不同的機器或平台上,讓系統具備很高的擴充性。
In many environments such as campus, factories or public places may need using audio broadcasting system to inform certain important information. Because of low scalability and high construction costing of traditional audio broadcasting system, we present the network audio broadcasting system based on Session Initiation Protocol (SIP). SIP is a media session conrol layer protocol using to send invite, update or terminate contol message of media session. It also co-operates with SDP, RTP and RTCP.
This system constructs on existing network infrastructure and has three components: SIP proxy server, broadcasting-side and receiving-side. You can design your own message handling processes in the server-side. The broadcasting-side is developed by open-source and corss-platform library, can port to different platform easily.
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[2]F. Andreasen and B. Foster, “Media Gateway Control Protocol (MGCP) Version 1.0”, Internet Engineering Task Force, RFC 3435, January 2003.
[3]Knarig Arabshian and Henning Schulzrinne, “A SIP-based Medical Event Monitoring System”, IEEE Conference on Mobile and Ubiquitous Systems, 2004.
[4]E. W. Burger and O. Frieder, “A novel system for remote control of household devices using digital IP phones”, IEEE Transactions on Consumer Electronics, 2006.
[5]H. Handley, H. Schulzrinne, E. Schooler, and J. Rosenberg, “SIP: Session Initiation Protocol”, Internet Engineering Task Force, RFC 2543, March 1999.
[6]J. Rosenberg, H. Schulzrinne, G. Camarillo, A. Johnston, J. Peterson, R. Sparks, M. Handley, and E. Schooler, “SIP: Session Initiation Protocol”, Internet Engineering Task Force, RFC 3261, June 2002.
[7]H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, “RTP: A Transport Protocol for Real-Time Applications”, Internet Engineering Task Force, RFC 1889, January 1996.
[8]H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, “RTP: A Transport Protocol for Real-Time Applications”, Internet Engineering Task Force, RFC 3550, July 2003.
[9]H. Schulzrinne, A. Rao, and R. Lanphier, “Real Time Streaming Protocol (RTSP)”, Internet Engineering Task Force, RFC 2326, April 1998.
[10]F. Cuervo, N. Greene, A. Rayhan, C. Huitema, B. Rosen, and J. Segers,
Megaco Protocol Version 1.0”, Internet Engineering Task Force, RFC 3015, November 2000.
[11]M. Handley and V. Jacobson, “SDP: Session Description Protocol”, Internet Engineering Task Force, RFC 2327, April 1998.
[12]J. Rosenberg and H. Schulzrinne, “Reliability of Provisional Responses in the Session Initiation Protocol (SIP)”, Internet Engineering Task Force, RFC 3262, June 2002.
[13]S. Donovan, “The SIP INFO Method”, Internet Engineering Task Force, RFC 2976, October 2002.
[14]B. Campbell, J. Rosenberg, H. Schulzrinne, C. Huitema, and D. Gurle, “Session Initiation Protocol (SIP) Extension for Instant Messaging”, Internet Engineering Task Force, RFC 3428, December 2002.
[15]B. Roach, “Session Initiation Protocol (SIP) – Specific Event Notification”, Internet Engineering Task Force, RFC 3265, June 2002.
[16]R. Sparks, “The Session Initiation Protocol (SIP) Refer Method”, Internet Engineering Task Force, RFC 3515, April 2003.
[17]M. Handley, C. Perkins, and E. Whelan, “Session Announcement Protocol”, Internet Engineering Task Force, RFC 2974, October 2000.
[18]J. Rosenberg and H. Schulzrinne, “An Offer/Answer Model with the Session Description Protocol (SDP)”, Internet Engineering Task Force, RFC 3264, June 2002.
[19]PJPROJECT, “Open source SIP stack and media stack for presence, im/instant messaging, and multimedia communication”; http://www.pjsip.org/
[20]wxWidgets, “wxWidgets: Cross-Platform GUI Library”;
http://www.wxwidgets.org
[21]OpenSER, “OpenSER – the Open Source SIP Server”; http://www.openser.org