本論文以文獻 [1] 所提出的 SIP 視訊串流系統為基礎，針對該系統進行通訊協定標準化與最佳化，以提升系統效能與穩定性。並進一步將可連線之用戶端延伸至網路位址轉換 (Network Address Translation, NAT) 網路內，以實現 NAT 穿越功能。另一方面也針對使用者介面進行修正，讓使用者能更容易地對本系統進行操作與設定。

　　本研究中我們使用了 JAIN-SIP 來做為會談起始協議 (Session Initiation Protocol, SIP) 與會談描述協議 (Session Description Protocol, SDP) 用戶端程式的實現，並搭配 OpenSIPS 建立之 SIP 伺服器及網頁伺服器等服務，將
VLC (VideoLAN Client) 串流伺服器軟體於直播用戶端上所擷取之串流資料透過即時傳送協議 (Real-time Transport Protocol, RTP) 與即時串流協議 (Real Time Streaming Protocol, RTSP) 轉送至一般觀眾手中。

　　藉由這些研究，我們希望讓一般人能用最簡便的方式－無論何時何地、無論桌上型或筆記型電腦，只要搭配網路攝影機 (Webcam) 以及開放原始碼
軟體(Open Source Software, OSS)，便可輕鬆建立自己的影音串流頻道。

This thesis is based on SIP Video Streaming System which proposed in references [1]. We standardized protocols and optimized in this system that increase system efficacy and stability, and made the NAT (Network Address Translation) traversal capacity that means a client can connect to this system inside NAT. On the other side, we also made a significant modification on user interface that can easily operate and setup this system by normal users.

We use JAIN-SIP to implement the client of SIP (Session Initiation Protocol) and SDP (Session Description Protocol) with a SIP server which setting up by OpenSIPS, web server and etc. These services can forward the streaming data which is captured by the live sharer from the VLC (VideoLAN Client) streaming server through RTP (Real-time Transport Protocol) and RTSP (Real Time Streaming Protocol) to any audience.

By means of this thesis, we expect that user can make own streaming channel only using webcam and open source software at any time, any place, no matter what desktop or laptop he used.

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