Full metadata record
DC Field | Value | Language |
---|---|---|
dc.contributor.author | 鍾永國 | en_US |
dc.contributor.author | Chung, Yung-Kuo | en_US |
dc.contributor.author | 羅濟群 | en_US |
dc.contributor.author | Lo, Chi-Chun | en_US |
dc.date.accessioned | 2014-12-12T02:39:22Z | - |
dc.date.available | 2014-12-12T02:39:22Z | - |
dc.date.issued | 2013 | en_US |
dc.identifier.uri | http://140.113.39.130/cdrfb3/record/nctu/#GT079864525 | en_US |
dc.identifier.uri | http://hdl.handle.net/11536/73970 | - |
dc.description.abstract | 電腦電話整合(Computer Telephony Integration,CTI)經過這幾年蓬勃發展,提供眾多解決方案。整合通訊(Unified Communication,UC)目的建立企業內部各種不同通訊設備整合應用的平台,藉此提高工作效率。VoIP (Voice over on Internet Protocol)集成語音、傳真、數據和視頻等多種通信功能,應用在企業的總公司與分公司之間,可以為企業降低通信成本增加營運效率。瀏覽器技術的創新應用,具備呈現多媒體的豐富內容元素,網路使用者多以Browser-Based為操作介面中心,使用網路資源。傳統瀏覽器無法實現的功能,必須透過外掛程式安裝ActiveX、Java Applet、Flash等plug-ins元件來滿足特定功能。WebRTC(Web Real-Time Communication )不需安裝plug-ins外掛元件,利用HTML5制定之新通訊協議,可實現視訊音訊即時通訊能力,徹底擺脫外掛程式安全陰影,帶來更多無限可能性的創新應用。本論文研究透過使用單一瀏覽器介面及新興標準通訊協議,從普及的VoIP語音應用為基礎,擴展以瀏覽器基於WebRTC即時通訊能力,兼俱音視訊電話系統之實作方法。整合sipML5網頁開發框架(Framework)、IP-PBX開放原始碼Freeswitch系統功能,實作以SIP(Session Initiation Protocol)為通訊方式之音訊視訊的電話系統。 | zh_TW |
dc.description.abstract | Computer Telephony Integration (CTI) offers many solutions after several years of vigorous development. The purpose of Unified Communications is to establishing a variety of different communications devices within the enterprise application integration platform, in order to enhance work efficiency. Voip(Voice over on Internet Protocol) set Multiple communication functions and applications in between corporate headquarters and branch offices, enterprises can reduce communication costs and increase operational efficiency. It reduce communication costs and increase operational efficiency. Computer telephony integration of resources, unified communications applications and VoIP Cost savings, from the three perspective, how to enhance the effective use of resources, is the enterprise needs of integration of resources. The innovative applications of browser technology include rich multimedia content elements. Internet users use network resources with Browser-Based as the operator Interface. Traditional browser must been installed ActiveX, Java Applet, Flash and other plug-ins to meet specific functions, otherwise, it can not achieve the function. Web Real-Time Communication use new websocket communication of HTML5 can enable video audio real-time communication, it need not to install plug-ins plug components and bring more innovation applications. The research of this paper based on the VoIP Voice, by using a single browser interface and a new standard communication protocol to extend real-time audio and video communication capabilities. Use IP-PBX open source Freeswitch system functions, sipML5 web development framework, implemented with SIP (Session Initiation Protocol) and audio video phone system. | en_US |
dc.language.iso | zh_TW | en_US |
dc.subject | 瀏覽器 | zh_TW |
dc.subject | 音訊 | zh_TW |
dc.subject | 視訊 | zh_TW |
dc.subject | WebRTC | en_US |
dc.subject | Html5 | en_US |
dc.subject | Websocket | en_US |
dc.subject | SIP | en_US |
dc.subject | VoIP | en_US |
dc.subject | Freeswitch | en_US |
dc.subject | SIPml5 | en_US |
dc.title | 基於WebRTC之音視訊電話系統實作研究 | zh_TW |
dc.title | The Study of Implementing Audio and Video WebPBX System based on WebRTC | en_US |
dc.type | Thesis | en_US |
dc.contributor.department | 管理學院資訊管理學程 | zh_TW |
Appears in Collections: | Thesis |