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研究生:余丁坤
研究生(外文):Ting-Kuan Yu
論文名稱:SIP多組對等多點式網路語音會議之設計與實作
論文名稱(外文):SIP P2P Multipoint Audio ConferencingDesign and Implementation
指導教授:吳曉光吳曉光引用關係
指導教授(外文):Hsiao-kuang Wu
學位類別:碩士
校院名稱:國立中央大學
系所名稱:資訊工程學系碩士在職專班
學門:工程學門
學類:電資工程學類
論文種類:學術論文
論文出版年:2004
畢業學年度:92
語文別:英文
論文頁數:47
中文關鍵詞:SIPMultipoint ConferenceP2PMSIP
外文關鍵詞:MSIPP2PMultipoint ConferenceSIP
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己有很多論文在討論如何利用SIP來實現網路媒體會議,但是大部分的設計都是利用網路會議伺服器的方式達成,也有一些是用使用者橋接方式。目前己看到的產品也是以這兩種為主。 另一種實現網路會議的方法即是使用對等式的網狀拓樸結構。相較於上述兩種方法,網狀拓樸結構並不需要會議伺服器或一會議參與者來扮演訊息橋接的角色。並且支援動態網路會議的建立及支持良好的移動性,在今日手持式可上網裝置普及的時代,實在合適於用此方法來實作網路會議。

MSIP [12] 提出了一個方法來為以SIP為基礎的網路會議建構一個網狀拓樸結構。而這方法也被PIMRC這個機構認可,並將被刊登在PIMRC 2004。這份論文的初始目的也就是為了改進MSIP並為這個方法實作出一網路會議的應用程式。

在這篇論文�堙AMSIP將在設計的那一章節被分析及改進。改進後的版本命名為SPPMC。SPPMC不但改進原來的功能, 還增強之以支持同時啟動且運作多個網路會議。在實作的那一章節,實作的方法及系統架構也重點性的被描述。以SPPMC的方法完成的網路會議程式被命名為SMC-Media。這個程式支援了 multicast PCM 語音串流。 我們將程式的一些快取圖樣置於展示那一章節,用來說明會議的建立及參與的方法。頻寬消耗實測,包含利用UDP或TCP來傳送SIP信號,及會議建立耗時的實測也會被列舉出,以供參考。
Internet conferencing is a program for conference on Internet. Most SIP Internet conferencing use centralized conference server model, and some ones use end user mixing model for signaling. Respect to the above two models, mesh model has difference advantages. Every peer in mesh has the same capability to initial a conference session in dynamic without a conference server or a bridging endpoint. Besides, SIP supports mobility, the combination of mesh conference model and SIP is a suitable method to implement Internet conferencing for mobile handset devices.

MSIP had proposed a method, which creates a mesh topology SIP signaling, for Internet conferencing. Besides, this method will be presented on PIMRC 2004. This thesis is initialed for improving and implementing MSIP.

In this thesis, MSIP is analyzed and improved to support simultaneously creating multiple conference sessions. We call the improved method SPPMC. The improvements of MSIP, detail design of SPPMC, and system architecture for implementation are all listed in design sections and implementation section. A conferencing program, named SMC-Media, with multicast PCM audio stream is implemented. The snapshots of the program demonstrated for conference establishment scenario, and conference joining scenario, and the result of some measurements, including network bandwidth consuming and conference forming Time with TCP and UDP underling transport are described in demonstration section.
Chapter 1 Introduction ………………………………………………………………...1
1.1 Internet Conferencing Consideration …………………………………….……1
1.2 Contribution of this thesis …………………….……………………………….3
Chapter 2 Related Works …………………………………………………….………..4
2.1 Internet Phone Call Processing Protocols…………………………….…….….4
2.2 Internet Multi-Media Conferencing for SIP….…………………….……….…5
2.2.1 Conference Control …………………………………………………..…6
2.2.2 Conference models…………….…………………………………….…..7
2.3 SIP Extension…...………………………………………………….………….9
Chapter 3 Protocol Design.............................................................................................10
3.1 MSIP Analysis and Improvement......................................................................10
3.2 SPPMC Design .................................................................................................11
3.3 SPPMC Signaling and Media Topology............................................................13
3.4 SPPMC Scenarios .……..………………….……………………….…...….....14
3.4.1 Conference Establishment …………………………………….………..14
3.4.2 Conference Joining...………….………….……….………………....….17
Chapter 4 Implementation …………………………….………………………....…...19
4.1 Implementation Model ……….……….…………….…..……………....……19
4.2 Program Architecture ………………………………………………………....20
4.3 Member Information Processing ……………………………………………..25
4.4 Audio Mixing………………….….………………………….………………..27
Chapter 5 Demonstrations & Results ……………………………………….……..…30
5.1 Unicast Audio Conference Establishment Scenario Demo …………..………31
5.2 Unicast Audio Conference Joining Scenario Demo ………………………….36
5.3 Multicast Audio Conference Establishment Scenario Demo .………………..37
5.4 Bandwidth and Time Consuming Measurements ……………………….……39.
5.4.1 Conference Forming Time Measurement……………………………….39
5.4.2 Network Bandwidth Consuming Measurement…………………………39
5.5 Network Bandwidth throughput estimate for RTP Audio stream……………..42
Chapter 6 Conclusion and discussion ………………………………………………...44
Chapter 7 Reference …………………………………………………………………..46
[1] Bormann, C., Kutscher, D., Ott, J., and Trossen, D. “Simple conference control protocol service specification.” Internet Draft, Internet Engineering Task Force, Mar. 2001.

[2] Elin Wedlund, and Schulzrinne, H. “Mobility support using SIP.” 1999

[3] Handley, M., and Jacobson, V. “SDP: session description protocol.” Request for Comments 2327, Internet Engineering Task Force, Apr. 1998.

[4] International Telecommunication Union, “Packet based multimedia communications system.” Recommendation H.323, Telecommunication Standardization Sector of ITU, Geneva, Switzerland, Feb. 1998.

[5] Kutscher, D., Ott, J., and Bormann, C. "Requirements for Session Description and Capability Negotiation." Internet Draft, draft-kutscher-mmusic-sdpng-req-01.txt, November 2000.

[6] Rosenberg, J., and Schulzrinne, H. “Models for multi party conferencing in SIP.” Internet Draft, draft-ietf-sipping-conferencing-models-00.txt, Nov. 2001.

[7] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A., Peterson, J., Sparks, R., Handley, M. and Schooler, E. "SIP: Session Initiation Protocol.” RFC 3261, June 2002.
[8] Schulzrinne, H., Casner, S., Frederick, R., and Jacobson, V. “RTP: A Transport Protocol for Real-TimeApplications.” Request for Comments 3350, July 2003.

[9]Singh, K., Nair, G., and Schulzrinne, H. "Centralized Conferencing using SIP." Proc. of the 2nd IP-Telephony Workshop (IPTel'2001), April 2001.

[10] Rosenberg, J., and Schulzrinne, H. “Guidelines for Authors of Extensions to the Session Initiation Protocol (SIP).” Internet-Draft, draft-ietf-sip-guidelines-07.txt, Oct. 2003.

[11] Uyless Black. “Voice Over IP, Second Edition.” VoIP Gateways and IP Call Processing Protocols. New Jersey: Prentice-Hall, 2002. 187-218.

[12] Zhi-Hao Wu, and Hsiao-kuang Wu “Application Layer multipoint conference Using SIP.”, June 2003.

[13] http://www.cisco.com/en/US/products/hw/phones/ps379/ps1854/index.html

[14] http://www.vovida.org/

[15] http://www.4front-tech.com/opensound.html

[16] http://www.ntop.org/ntop.html
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