| Summary: | 碩士 === 國立成功大學 === 電機工程學系碩博士班 === 93 === Recently, growing the foundation of internet-based, wide band network changing the behavior of human being, and redefine the style of network multimedia, wide band network enable the apply of multimedia over internet become practical technology, for example : Videoconference, VOD(Video On Demand) etc. , among these application, the technology of widely use and successfully apply is the video and audio over the internet , we call all these is VoIP(Voice Over Internet Protocol), now is rapidly change the voice communication manner by traditional phone .
In order to understand rapidly the technology of delivering real-time video and audio data packet through the network, and develop the video telephone in the future, in order to achieve the goal of video phone, we study and analyze the open source of audio conferencing software and video conference, then integrated the audio and video software successfully.
In this thesis we study the Real-time Transport Protocol (RTP) and the Real-time Control Protocol (RTCP) for transmitting real-time packets, and the Message Bus(MBus), and the packet loss, repair mechanism of internet multimedia. we present the architecture and the execution flow of the video and audio, finally, we present the execution time of each function block making up video phone, and the number of packet loss in the internet.
|
|---|
