研究生: |
洪吉亮 Ji-Liang Hong |
---|---|
論文名稱: |
以單一延伸裝置整合多重網路電話服務 Using an Extension Phone to Integrate Multiple VoIP Networks |
指導教授: |
石維寬
Wei-Kuan Shih |
口試委員: | |
學位類別: |
碩士 Master |
系所名稱: |
電機資訊學院 - 資訊工程學系 Computer Science |
論文出版年: | 2006 |
畢業學年度: | 94 |
語文別: | 中文 |
論文頁數: | 44 |
中文關鍵詞: | 網際電話 、行動電話 |
外文關鍵詞: | VoIP, Mobile Phone |
相關次數: | 點閱:3 下載:0 |
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所謂的VoIP即是要跳脫既有PSTN對於資源使用的思維,利用既有的IP網路打造出更經濟的語音服務。VoIP不但有價格上的優勢,還有多功能、移動性等等好處,然而其本身亦有通話品質、相容性、結構複雜、安全性等問題。近來,藉由P2P網路和多種軟體技術,大幅提昇通話品質及頻寬使用率,並有助於可靠度和安全性。但是利用上述技術卻導致相容度和複雜度的問題更加地嚴重。基於現狀,我們需要一個簡單、有彈性、可移植的方法來處理此一困境。
在研究過現存系統的優缺之後,本研究提出一個新的架構,此架構包含兩個元件:Station及Extension,Station必須有足夠的運算能力和網路頻寬去加入多個不同的VoIP網路,並且負責維護與Extension端間的連線;Extension則是一個簡單的裝置,他僅有一張全雙工音效卡和網路介面而已。
Extension端的功能在於收集使用者的聲音與提供使用者控制介面。聲音部分分別是將音效卡收到的聲音經由網路導向Station,與把從Station傳來的聲音經由音效卡發出給使用者。Station端的功能則在於傳遞聲音至語音服務、語音服務網路維護與系統訊息分派。聲音部分則分別是將Phone Service收到的聲音經由網路導向Extension,與把從Extension傳來的聲音經由Phone Service出給通話對方。
經由實作的結果發覺此法確實是可以整合多種不同的VoIP系統,使用者可以簡單地存取其需要的資源而不用顧慮其差異,且Extension端僅需要少量的資源即可運行,並且具有在更進一步縮減的可能性,而不論是Extension端還是Station端擴充都是可行的。
The so-called VoIP is to replace the resources management policy of PSTN and use Internet to make a more economic phone service. VoIP has advantage at prices, functionality and mobility. Even so, there are questions, such as QoS, compatibility, complexity and security ,etc. in its. Recently, the QoS and the utilization of bandwidth has be substantially improved by P2P framework and various software technologies, and such approach is also very helpful to reliability and security. But the problem of incompatibility and complexity has become more serious with using above technologies. Base on the current situation, we need a simple, flexible and portable method to solve this predicament.
After studying the existing systems, this thesis introduces a new architecture. There are two main components in this architecture: the Station and the Extension. The Station must have adequate computing power and network bandwidth to join multiple VoIP networks and be responsible to maintain the connection between he and the Extension. The Extension is a simple device which only has a full-duplex soundcard and a network interface.
The function of the Extension is to collect the user’s voice and to provide user interface. The voice part is separately transmit the sound received from soundcard to the Station via the network and transmit the voice received from the Station to the users via the soundcard. The function of the Station is to transmit voice to Phone Services, maintain the network of Phone Services and dispatch system messages. The voice part is separately transmit the voice received from Phone Services to the Extension via the network and transmit the voice received from the Extension to another user via Phone Services.
According evaluation result, this method can really integrate many kinds of different VoIP networks, and the user can easily access its resource he needed but don’t need care about its difference. Additionally, the Extension need only small amount of resources to work and have possibility to reduce further. Then, no matter the Extension or the Station are all extensible.
[1] K. EGEVANG, The IP Network Address Translator (NAT), RFC 1631, Network Working Group, May 1994
[2] International Telecommunication Union, “Packet-based multimedia communications systems v2”, ITU-T Recommendation H.323, February 1998
[3] M. Handley, “SIP: Session Initiation Protocol”, RFC 2543, March 1999
[4] H. Schulzrinne, “RTP: A Transport Protocol for Real-Time Applications”, RFC 3550, July 2003
[5] P. Saint-Andre, Ed., “Extensible Messaging and Presence Protocol (XMPP): Core”, RFC-3920, October 2004
[6] Scott Ludwig, “JEP-0166: Jingle”, [Online document] March 2006, Available at HTTP: http://www.jabber.org/jeps/jep-0166.html
[7] Scott Ludwig, “JEP-0167: Jingle Audio Media Description Format”, [Online document] March 2006, Available at HTTP: http://www.jabber.org/jeps/jep-0167.html
[8] International Telecommunication Union, “Pulse code modulation (PCM) of voice frequencies”, ITU-T Recommendation G.711, November 1988
[9] International Telecommunication Union, “Dual rate speech coder for multimedia communications transmitting at 5.3 and 6.3 kbit/s”, ITU-T Recommendation G.723, March 1996
[10] International Telecommunication Union, “Coding of speech at 8 kbit/s using conjugate-structure algebraic-code-excited linear prediction (CS-ACELP)”, ITU-T Recommendation G.729, March 1996
[11] S. Andersen, “Internet Low Bit Rate Codec (iLBC)”, RFC 3951, December 2004
[12] Greg Herlein, “RTP Payload Format for the Speex Codec”, draft-herlein-speex-rtp-profile-02
[13] Portable Operating System Interface, ISO/IEC 9945-1: 2003 (IEEE Std. 1003.1: 2001)
[14] D. Box. Essential COM, Addison-Wesley, 1998
[15] Daniel P. Bovet, Understanding the Linux Kernel, O’Reilly, 2000
[16] J. Myers, “Post Office Protocol - Version 3”, RFC 1939, May 1996
[17] J. Postel, “File Transfer Protocol (FTP)”, RFC 959, October 1985
[18] R. Fielding, “Hypertext Transfer Protocol -- HTTP/1.1”, RFC 2616, June 1999
[19] 4Front, OSS v3 Programmer's guide, [Online book] Nov 2000, Available at HTTP: http://www.opensound.com/pguide/oss.pdf
[20] Jeff Tranter , Linux Multimedia Guide , O'Reilly 1996
[21] Ralf H□lzer, “Linux Cryptoloop HOWTO”, [Online document] Jan 2004, Available at HTTP: http://www.linux.com/howtos/Cryptoloop-HOWTO/index.shtml
[22] CramFs: cram a file system onto a small ROM or compressed ROM file system, http://sourceforge.net/projects/cramfs/
[23] Microsoft Corporation, “Waveform Audio” , MSDN Library, [Online library], Available at HTTP: http://msdn.microsoft.com/library/default.asp?url=/library/en-us/multimed/htm/_win32_waveform_audio.asp
[24] Microsoft Corporation, “DirectSound”, MSDN Library, [Online library], Available at HTTP: http://msdn.microsoft.com/library/default.asp?url=/library/en-us/directx9_c/IDirectSound8.asp
[25] Bob Quinn, Windows Sockets Network Programming, Addison-Wesley, December 1995
[26] Microsoft Corporation, “Winsock2”, MSDN Library, [Online library], Available at HTTP: http://msdn.microsoft.com/library/default.asp?url=/library/en-us/winsock/winsock/windows_sockets_start_page_2.asp
[27]Microsoft Corporation, “Windows Messages”, MSDN Library, [Online library], Available at HTTP: http://msdn.microsoft.com/library/default.asp?url=/library/en-us/winui/winui/windowsuserinterface/windowing/messagesandmessagequeues/aboutmessagesandmessagequeues.asp
[28] Skype Technologies SA , Skype Public API 2.0 Reference Guide, [Online book], Available at HTTP: https://developer.skype.com/Docs/ApiDoc
[29]Microsoft Corporation, “Windows Messenger API” , MSDN Library, [Online library], Available at HTTP: http://msdn.microsoft.com/library/default.asp?url=/library/en-us/winmessenger/winmessenger/messenger_entry.asp
[30]Microsoft Corporation, “MSN Messenger Activity SDK” , MSDN Library, [Online library], Available at HTTP: http://msdn.microsoft.com/library/default.asp?url=/library/en-us/messengerp2p/introduction.asp
[31]Microsoft Corporation, “MSN Messenger Protocol”, [Online document] March 2003, Available at HTTP: http://www.hypothetic.org/docs/msn/index.php
[32]User-mode Linux Kernel, http://user-mode-linux.sourceforge.net/
[33]netperf: Rick Jones' <raj@cup.hp.com> network performance benchmarking package, http://www.netperf.org/
[34]NetPIPE: Quinn O. Snell, Armin R. Mikler and John L. Gustafson, Ames Laboratory/Scalable Computing Lab, Ames, Iowa 50011, USA, A Network Protocol Independent Performance Evaluator, http://www.scl.ameslab.gov/netpipe/paper/full.html
[35] Matteo Frigo and Steven G. Johnson, “FFTW”, MIT, http://www.fftw.org/