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  • 學位論文

VoIP 電話與傳統電話結合之研究及實作

A Study and Practice in the Integration of VoIP and PSTN Phones

指導教授 : 白宏達
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摘要


隨著硬體晶片、軟體壓縮技術、以及平台建置的不斷發展,透過網際網路傳送語音訊號(Voice over Internet Protocol, VoIP)的網路電話變得越來越普及。雖然網路電話有著低成本的好處,但是傳統類比電話還是有些無法取代的優勢,例如不需外接電源。如何整合兩者,並同時擁有各自的優點,是一項很值得研究的課題。 本研究首先將針對VoIP裝置之硬體,作深入的研究。接下來,提出VoIP網路裝置的架構及優缺點,並綜合嵌入式系統介面整合技術及概念,提供網路電話與傳統電話功能的整合構想。 最後,利用一硬體平台截取轉換傳統電話狀態訊號說明為例,論證網路電話主控制器是可監控傳統電話之行為狀態,並達到兩者主從之間的控制整合概念。使網路及傳統電話硬體平台能夠作最佳的整合,以兼具兩者的優點,作為未來整合晶片設計之基礎架構。

關鍵字

網路電話

並列摘要


With the ongoing development of hardware chips, software compression technology and platform establishment, an internet phone which transmits Voice over Internet Protocol (VoIP), has become increasingly popular. Although the internet phone has the advantage of being low cost, a traditional analog phone still holds other advantages which cannot be replaced; for example, traditional analog phone does not require an external power source. Therefore, the study to integrate these two types of phones while keeping their individual advantages; is worthwhile. This study will first focus on the hardware of VoIP devices and do an in-depth research. Next, we will study the structure of VoIP devices and their advantages and disadvantages; combine the integration techniques and concepts of the embedded system interface and then propose the concept to integrate the functions of the internet phone and the traditional analog phone. Finally, as an example, we will use a hardware platform to intercept and transform the status signal of a traditional analog phone, which will demonstrate the concept that the main controller of a VoIP phone can be used to monitor the status condition of a traditional analog phone, and furthermore, establish the control concept of the master-slave devices. With this, we can provide the best integration of the hardware platform between the internet phone and the traditional analog phone; obtain the optimization of both and use them as the base structure for future integrated chip design.

並列關鍵字

internet phone

參考文獻


[1] ITU-T Rec.H.323 “Packet-Based Multimedia Communications System” June 2006.
[2] H Liu and P Mouchtaris, “Voice over IP signaling: H.323 and beyond” IEEE Communications Magazine pp.142-148, October. 2000.
[3] RFC 2543, “SIP: Session Initiation Protocol” IETF, March 1999.
[4] RFC 3261, “SIP: Session Initiation Protocol” IETF, June 2002.
[5] RFC 2327, “Session description protocol” IETF, April 1998.

被引用紀錄


王文義(2014)。利用整合網路電話及實體電話提升客服中心員工效能之研究〔碩士論文,國立臺中科技大學〕。華藝線上圖書館。https://doi.org/10.6826/NUTC.2014.00123
呂禮有(2015)。應用互動式鑽石模型探究VoIP導入對企業成效之研究-以C公司為例〔碩士論文,國立中央大學〕。華藝線上圖書館。https://www.airitilibrary.com/Article/Detail?DocID=U0031-0412201512042555
陳巍中(2017)。VoIP通話品質與流量之研究-以SIP及Red5為例〔碩士論文,國立臺中科技大學〕。華藝線上圖書館。https://www.airitilibrary.com/Article/Detail?DocID=U0061-1701201718142800

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