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研究生:江萬添
研究生(外文):Wan-Tien Chiang
論文名稱:內建式IP交換機之設計與實現
論文名稱(外文):Design and Implementation of a Built-in IP Switch
指導教授:賴國華
指導教授(外文):Robert Lai
學位類別:碩士
校院名稱:元智大學
系所名稱:資訊工程學系
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:2001
畢業學年度:89
語文別:中文
論文頁數:75
中文關鍵詞:多媒體通信嵌入式系統電話閘道
外文關鍵詞:H.323VOIPITG
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儘管目前在H.323系列標準方面的研究繁多,但大部分學術界研究僅針對理論的觀點出發,很少考慮到實作的可行性,本篇論文的目的即在考量當今網際網路限制以及媒體資料的即時性,以實作的角度進行研究,提出一套H.323多媒體通訊系統的完整設計。
論文中提出一種內建式IP交換機設計方法與裝置架構。首先,在傳統電路式交換機直接內建VOIP功能;藉由此方法,取代外加VOIP方式以增進QoS。我們的VOIP採用嵌入式系統(Embedded System)架構,通信協定採用ITU所訂定H.323多媒體通信標準,並且對交換機與IP控制方式進行設計。
另外利用網路性能檢測機制提高IP電話服務品質,使用一個能夠接收和分析RTCP封包的監控程序,並結合其他網路管理程序,可以根據網路的即時運行情況,主動調整語音壓縮機制,改變資料傳輸量以提升QoS;並由模擬結果證實此方法確實可行。

In this thesis, we have developed a communication system base on H.323 multimedia communication protocol for transmitting multimedia data on the band-limited Internet. Instead of building an additional VOIP module outside of the traditional circuit switch, we directly implemented it into an embedded switching system for the sake of improving the QoS.Additionally, we also proposed a networking performance diagnostic mechanism, which is a combination of the procedure that can receive and analyse RTCP packet and other network management procedures,to improve the service quality. Furthermore, according to the real-time network situation, the mechanism can also automatically adjust voice compression rate to increase the QoS. Finally, the simulation results show that our system is effective.

表目錄III
圖目錄IV
第一章緒論1
1.1 背景1
1.2 範疇4
1.3 論文組織5
第二章相關研究6
2.1 VOIP現況6
2.2 VOIP信號方式9
2.3 IP電話標準11
2.4 VOIP 對QOS的要求14
第三章系統設計與實現21
3.1 系統結構21
3.2 RAS模組25
3.2.1 網守搜尋(Gatekeeper Discovery)25
3.2.2 端點註冊(Endpoint Registration)26
3.2.3 端點定位(Endpoint Location)27
3.2.4 接入許可與退出(Admissions and Disengage)27
3.2.5 頻寬調整(Bandwidth Change)28
3.2.6 狀態與資源(Status and Resource)29
3.3 呼叫信令模組30
3.3.1 無Gatekeeper的基本呼叫建立過程30
3.3.2 兩端點註冊同一個Gatekeeper30
3.3.3 出局啟動程序(Outgoing Setup Procedure)31
3.3.4 入局啟動程序(Incoming Setup Procedure)32
第四章服務品質控制33
4.1 IP傳輸對QOS 支持33
4.2 排隊規律(QUEUING DISCIPLINE)35
4.3 語音播放機制38
4.4 自我調整能力(ADAPTIVE)機制40
4.5 驗證模型(VALIDATION MODEL)43
第五章結論51
5.1 研究成果51
5.2 未來工作52
參考文獻53
附錄A RAS消息一覽表55
附錄B RAS模組流程圖56
附錄C 呼叫信令消息一覽表70
附錄D 呼叫信令模組流程圖71

[1] ITU-T Recommendation H.323, “Infrastructure of audiovisual services — Systems and terminal equipment for audiovisual services,” 1999.
[2] ITU-T Recommendation H.225, “Call signalling protocols and media stream packetization for packet-based multimedia communication systems,” 1997.
[3] ITU-T Recommendation H.245, “Control protocol for multimedia communication,” 2000.
[4] ITU-T Recommendation Q.931, “Digital subscriber signalling system no.1(DSS1)—ISDN user-network interface layer 3 specification for basic call contro,” 1993.
[5] M. Handley and V. Jacobson, “ SDP: Session Description Protocol,” Internet RFC2326,April 1998.
[6] Maryann P.Maher, “Session Announcement Protocol: Version 2,” Internet Draft,May 1998.
[7] H. Schularinne, “Real Time Streaming Protocol(RTSP),” RFC2326,April 1998.
[8] M. Handley, H. Schulzrinne, E. Schooler, and J. Rosenberg, “SIP: Session Initiation Protocol,” Internet RFC2543, March 1999.
[9] M. Arango and C Huitema ,“Media Gateway Control Protocol(MGCP),” IETF draft,Feb 1999.
[10] C. Huitema and F. Andreasen, “Media Gateway Control Protocol(MGCP) Call Flow Test Case,” IETF draft,Feb 1999.
[11] Cisco. Enterprise IP Packet Telephony Solution Guide. White paper. http://www.cisco.com.
[12] ITU-T Recommendation E.164,“Telephone Network and ISDN Operation Numbering Routing and Mobile Service Numbering Plan For Then ISDN Era,” 1991.
[13] Uyless Black, Voice Over IP ,Prentice Hall PTR,1999.
[14] Walter J.Goralski,Matthew C.Kolon.,IP Telephony,McGraw-Hill,2000.
[15] ITU-T Recommendation G.711,“Pulse Code Modulation (PCM) of Voice Frequencies,” 1988.
[16] ITU-T Recommendation G.723 , “Dual Rate Speed Coder For Multimedia Communications Transmitting at 5.3 and 6.3 kbit/s,” March 1996.
[17] ITU-T Recommendation G.729 , “Coding of Speed at 8kit/s using conjugate structure algebraic Code excited Linearprediction,” March 1996.
[18] ITU-T Recommendation G.729A , “A Reduced Complexity 8kbit/s CS-ACELP Speech Codec,” Nov 1996.
[19] ITU-T Recommendation G.728 , “Coding of Speech at 16kbit/s using Low Delay Code Excited Linear Prediction,” May 1992.
[20] 方盈,TCP/IP通信協定-入門與應用,四版,台北,博碩,2000.
[21] Hong Liu and Petros Mouchtaris. “ Voice Over IP Signaling: H.323 and Beyond,“ IEEE Communications Magazine, October 2000,pp 142-148.
[22] ITU-T Recommendation Z.100 , “CCITT Specification and description language (SDL),” March 1993.
[23] Cox, R. V.,Hassle,B.G., Lacuna, A.,Shahraray, B., and Rabiner,L.,”On the Applications of Multimedia Processing to Communication,” Proceeding of the IEEE, May 1998,Vol.86,No.5,.
[24] S. B. Moon,P.Skelly,and D. Towsley,”Estimation and removal of clock skew from network delay measurements,” in Proc. Infocom ’99,IEEE,New York,1999.
[25] Chinmay Padhye and Kenneth J. Christensen,”A New Adaptive FEC Loss Control Algorithm for Voice Over IP Applicatoin,”,IEEE, 2000,pp307-313.
[26] A Demers,S. Kashav,and S.Shenker, “Analysis and Simulation of a Fairing Queueing Algorithm,” ACM SIGCOMM,1989,pp1-12.

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