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  • 學位論文

基於SIP通訊協定之嵌入式網路視訊電話設計與實現

The Design and Implementation of an SIP-base Embedded Video Phone

指導教授 : 王振興 王永鐘
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摘要


在目前網路上的各種應用服務中,網路電話(VOIP)是近年來最熱門的一項服務。現今網路電話大致以SIP通訊協定為主流,因此本研究是將SIP使用者代理程式與網路視訊電話系統做整合,並實作在嵌入式開發平台上。然而網路視訊電話雖帶給人們較多便利,但同時也加重了網路系統處理時的負擔。網路節點會因同時湧入大量串流資料,而造成封包遺失,延遲變長等問題,這些問題對於需即時性的網路視訊電話來說影響更為嚴重,故在本研究中,針對此問題整合出一套具QoS的流量控制機制,該機制能有效的估測出網路頻寬,進而控制視訊畫面的傳輸量,達到即時且平順的畫面品質。 而本論文的嵌入式開發平台是使用Linux作為系統核心,配合WinBond W90221PA -RISC 32-bit做為嵌入式處理器,SIP協定方面是使用SIP代理伺服器和嵌入式使用者代理程式來完成網路電話的功能,並結合硬體影像壓縮(H.263)與硬體聲音壓縮(G.723),藉此降低多媒體的資料量,另外透過RTP即時傳輸協定,來達成即時性的雙向傳輸。最後經實測證實,本嵌入式視訊電話系統,與其他網路電話軟體做比較,本系統確實能達到擁有VOIP軟體之優點又同時俱備傳統電話的便利性。

並列摘要


Among the various internet application services, Voice over IP (VOIP) become the most popular application service in recent years. Currently, VOIP primarily follows SIP protocol. Therefore, this research is addressing on how SIP user agent program integrates into video IP Phone system and then implement it on the embedded development platform. The internet telephone is very convenient to people, but it does cause network system transmission burden. In this research, We propose a traffic control mechanism to resolve this issue. This mechanism effectively estimates internet bandwidth and subsequently controls the vedio data transmission to achieve instant and smooth video quality. In this research, Linux is used as the operation system of the embedded development platform, and combined with WinBond W90221PA-RISC 32-bit as the embedded processor. SIP proxy server and embedded user agent program are used to implement internet telephone functions. We apply H.263 Video Codec and G.723.1 Audio Codec to reduce multimedia data size. RTP transmission protocol is implemented to achieve immediate two-way transmission. Finally experiment proves that this system truly can achieve the merit of VOIP software and at the same time bears with the convenience of the traditions telephone.

並列關鍵字

embedded system SIP RTP H.263 bandwidth estimate

參考文獻


[13] 創傑科技股份有限公司,SVP 1.0 Demo Kit User Guide, 2003年8月。
[5] M. Handley, H. Schulzrinne, E. Schooler and J. Rosenberg, “SIP: Session Initiation Protocol,” RFC 2543, 1999.
[7] 李文宏,以SIP為基礎之網路視訊電話實作,碩士論文,國立台北科技大學電機工程系所,台北,2006。
[12] 許毓辰,嵌入式網路視訊電話之實現,碩士論文,國立台北科技大學電機工程系所,台北,2005。.
[16] C. Zhu, “RTP Payload Format for H.263 Video Streams,” RFC 2190, 1997.

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