(3.238.235.155) 您好!臺灣時間:2021/05/16 08:06
字體大小: 字級放大   字級縮小   預設字形  
回查詢結果

詳目顯示:::

: 
twitterline
研究生:黃俊欽
研究生(外文):Chun-Chin Huang
論文名稱:適用於ATM骨幹網路架構的改良型VoIP交錯式順序封包格式之研究與VoIP終端機的實作
論文名稱(外文):An improving VoIP Interleaved packet format for ATM backbone network and the implementation of a VoIP terminal
指導教授:李維聰李維聰引用關係
指導教授(外文):Wei-Tsong Lee
學位類別:碩士
校院名稱:逢甲大學
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:84
中文關鍵詞:交錯式順序網路電話
相關次數:
  • 被引用被引用:1
  • 點閱點閱:344
  • 評分評分:
  • 下載下載:52
  • 收藏至我的研究室書目清單書目收藏:2
本篇論文主要討論在一個以ATM為骨幹網路架構的VoIP網路環境下,提出一個改良的交錯式順序封包格式,並且和ATM Forum所提出的RTP標頭壓縮(Header Com -pressing)技術相結合,並加入提前錯誤修正(Forward Error Correction)技術的觀念。
ATM的RTP標頭壓縮技術主要是為了減少封包的負載,因此不必傳送多餘的封包標頭資訊。至於交錯式順序方法主要的觀念則是會重新排列訊框傳送出的順序,所以即使某些訊框遺失了,這個方法會在接收端重組回原來訊框的順序,進而將錯誤分散,使得封包遺失的影響降到最低。另外我們再加上提前錯誤修正的觀念,利用ATM細胞中無用的填補資料來傳輸額外的資訊。透過這額外的資訊,用來補償遺失的訊框資料。
而我們將這些方法的優點整合,套用在ATM骨幹網路上,形成一個更適合VoIP應用程式的封包格式,使得這個新的封包格式與ATM網路能夠互相運作得非常順利。
最後,我們實作一個VoIP終端機,而我們可以利用這個終端機在同一個區域網路上與另一台PC或是終端機互相通話。
This paper focuses on proposing an improving interleaved packet format over VoIP network architecture with ATM backbone. And we combine the RTP header compressing technology and Forward Error Correction (FEC) idea with our new method.
The main purpose of the RTP header compressing technology over ATM is in order to reduce the packet header overhead. Therefore, we don’t transmit the redundant packet header. With regard to the important idea of interleaving method is permuting the order of a sequence of frames. Even if some frames missing, this method would recover the original sequence of frames and randomize the distribution of errors after reception to decreasing the effect of packet loss. Besides, we include the concept of FEC using the padding information in ATM cells to transmit the redundant information. Through the redundant information, we can use it to compensate the missing frames.
We integrate those advantages of above methods to form a new packet format which more suitable VoIP applications. To make this new packet format interworking with ATM network very smoothly.
Finally, we implement a VoIP terminal. We use the VoIP terminal to communicate with another PC or VoIP terminal.
中文摘要Ⅰ
英文摘要Ⅱ
目錄Ⅲ
圖目娽Ⅶ
表目錄Ⅸ
第一章簡介1
第二章VoIP的介紹6
2.1 VoIP的發展及影響6
  2.1.1 VoIP的發展6
2.2.2 VoIP的影響 7
2.2 VoIP常見的網路架構8
2.3 H.323通信協定 9
  2.3.1 H.323的系統架構10
2.3.2終端機 11
2.3.3閘道管理者13
2.3.4閘道器 15
2.3.5多方會談控制單元15
2.3.6H.323相關協定17
2.3.7一個H.323連線的建立20
2.4 MGCP通信協定 21
2.4.1媒體閘道器22
2.4.2信號閘道器23
2.4.3媒體閘道控制器23
2.4.4MGCP相關通信協定24
2.5 其它相關協定25
2.5.1語音編解碼演算法25
2.6 H.323-to-H.323 Gateway Over ATM28
  2.6.1 VoIP封包在ATM的IP/UDP標頭省略程序29
2.7 VoIP封包在ATM的RTP標頭壓縮技術31
2.7.1 RTP標頭格式 31
2.7.2在ATM的RTP標頭壓縮技術32
2.7.3在ATM的RTP標頭壓縮過程34
第三章減少VoIP封包遺失影響的技術36
3.1提前錯誤修正技術36
3.2交錯式順序方法38
3.2.1 方法介紹38
3.2.1 封包格式40
第四章改良的交錯式順序方法43
4.1問題43
4.2適合的演算法及陣列大小45
4.3化簡酬載標頭46
4.4加入提前錯誤修正訊框48
4.5解決時間戳記差值問題50
4.6討論及比較52
4.6.1 封包遺失的比較54
4.6.2時間延遲的比較56
第五章VoIP終端機的實作57
5.1動機 57
5.1.1 使用傳統電話的習慣57
5.1.2可攜性57
5.2 終端機的架構58
5.3 CPU模組60
5.3.1 E86MON作業系統 62
5.3.2在終端機中負責的功能62
5.4 LAN模組 63
5.4.1 NE2000網路卡記憶體配置情形63
5.4.2在終端機中負責的功能64
5.5 DSP模組65
5.5.1在終端機中負責的功能66
5.6 語音轉換處理67
5.6.1 在終端機中負責的功能68
5.7 電話介面電路68
5.7.1在終端機中負責的功能71
5.8 程式部份71
5.8.1撰寫網路介面Driver71
5.8.2週邊晶片控制73
5.8.3溝通信號協定73
5.9 測試環境架構77
第六章結論與未來工作78
參考文獻80
感謝詞84
作者簡介85
[1] ITU-T Recommendation H.323, “Packet-based Multimedia Communications Systems,” Geneva, Switzerland, Sep. 1999.
[2] M. Arango, et al., “Media Gateway Control Protocol (MGCP), “ IETF RFC 2705, Oct. 1999.
[3] 倪正耀、周勝鄰及周瑞宏,“H.323 概論”,電腦與通訊73期,p18-23,1998
[4] 陳慶昌、陳雲龍及柳文儀,“VoIP技術與應用”,財團法人資訊工業策進會網路通訊雜誌,Dec,2000
[5] ITU-T Recommendation G.723.1, “Dual Rate Speech Coder for multimedia communications transmitting at 5.3 and 6.3 kbit/s,” March,1996.
[6] Handley, ISI/ Schulzrinne, Columbia University/Schooler, Catltech, “SIP : Session Initiation Protocol,” IETF draft, May,1998.
[7] M. Handley, ISI/ V. Jacobson, LBNL, “SDP : Session Description Protocol,” RFC2327, Apr.,1998.
[8] ITU-T Recommendation G.729, “Coding of Speech at 8 kbps Using Conjugate-Structure Algebraic-Code-Excited Linear-Prediction (CS-CELP). Geneva, CH:ITU, March, 1996.
[9] ITU-T Recommendation G.728, “Coding Speech at 16 kbps Using Low-Delay Code Excited Linear Prediction. Geneva, CH:ITU, September 1992.
[10] Daniel Minoli, Emma Minili, “Delivering Voice over IP Networks,” 1998.
[11] ATM Forum Technical Committee AF-SAA-0124.000, “Gateway for H.323 Media Transport over ATM.”
[12] Carlos M. Pazos, Marek R. Kotelba, and Andrew G. Malis, “Real-Time Multimedia over ATM : RMOA,” IEEE Communication Magazine, April 2000.
[13] IETF RFC 1889, RTP : A Transport Protocol for Real-Time Applications, January 1996.
[14] IETF RFC 2508, Compressing IP/UDP/RTP Headers for Low-Speed Serial Links.
[15] J-C. Bolot and A. Garicia, “Control Mechanisms for Packet Audio in the Internet,” Proceedings of IEEE INFOCOM, pp. 232-239, March 1996.
[16] Chinmay Padhye, Kenneth J. Christensen and Wilfrido Moreno, “A New Adaptive
FEC Loss Control Algorithm for Voice Over IP Applications,” Performance, Computing, and Communications Conference, 2000. IPCCC ''00.
[17] Colin Perkins, Jon Crowcroft, “Effects of Interleaving on RTP Header Compression,” IEEE INFOCOM 2000.
[18] C. S. Perkins, “RTP payload format for interleaved media,” IETF Audio/Video Transport Working Group, February 1999, Work in progress.
[19] C. S. Perkins, “RTP payload for Redundant Audio data,” RFC2198.
[20] AMD, AMD 186 ED/EDLV DATA SHEET.
[21] AMD, SD186ED Demo Board Hardware.
[22] DSP GROUP, CT8022 VOIP/VON G.723.1, G.729AB TrueSpeech Co-Processor.
[23] National Semiconductor, TP3054/TP3057 Enhanced Serial Interface CODEC/Filter COMBO Family, 1995.
[24] REALTEK Semi-Conductor, Realtek Full-Duplex Ethernet Controller with Plug and Play Function, 1995.
[25] AMD, Am79R70 Ringing Subscriber Line Interface Circuit, Jan 1998.
[26] AMD, Am79R70 Ringing SLIC Device Call Processing Considerations, 1997.
[27] AMD, Am79R70 Ringing SLIC User’s Guide, 1997.
[28] ITU Rec. G.711, Pulse Code Modulation (PCM) of Voice Frequencies, 1998.
[29] AMD, E86MON Software User’s Manual, 1998.
[30] Bill Goodman, “Internet Telephony and Modem Delay,” IEEE Network, May/June, 1999.
[31] Daniel Minoli, Emma Minoli, “Delivering Voice over IP Netwroks,” Wiley Computer Publishing, 1998.
[32] Malathi Veeraraghavan, “Internetworking telephony, IP and ATM networks,” Telecommunications Symposium, 1998. ITS ''98 Proceedings. SBT/IEEE International , 1998 , Page(s): 251 -256 vol.1
[33] Christian Huitema, Petros Mouchtaris, “An architecture for Residential Interent Telephony Service,” IEEE Internet Computing Volume: 3 3 , May-June 1999 , Page(s): 73 —82
[34] Jonathan Rosenberg, Henning Schulzrinne, “Internet Telephony Gateway Location,” INFOCOM ''98. Seventeenth Annual Joint Conference of the IEEE Computer and Communications Societies. Proceedings. IEEE
Volume: 2 , 1998 , Page(s): 488 -496 vol.2
[35] Thomas J. Kostas, Michael S. Borella, “Real-Time Voice Over Packet-Switched Networks,” IEEE Network Volume: 12 1 , Jan.-Feb. 1998 , Page(s): 18 —27
[36] 郭文山及陳岱妮,ATM網路的TCP/IP技術指南,儒林出版社,1999.
[37] 黃能富,區域網路與高速網路,維科出版社,1996.
[38] 陳克任,現代類比暨數位通訊,儒林出版社,2000.
[39] 李安哲,ATM的奧祕,松格出版社,1997.
QRCODE
 
 
 
 
 
                                                                                                                                                                                                                                                                                                                                                                                                               
第一頁 上一頁 下一頁 最後一頁 top