| Summary: | 碩士 === 淡江大學 === 資訊工程學系 === 87 === Nowadays, the Internet adopts TCP/IP, which grants no assurance of Quality of Service. Furthermore, the individual network bandwidth differs and software as well as hardware devices diverge, both facts cause packet delay jitter and packet loss. If the network environment is the same still, we must provide proper mechanisms in order to transfer real-time audio in the Internet with acceptable quality.
This thesis provides three mechanisms. Firstly, the adaptive audio playout mechanism gets proper playout delay time by on-line statistics of the packet delay time and packet loss rate of the recent time intervals in order to make the audio stream playback continuous with certain quality. Secondly, the error recovery mechanism makes the main information of the loss packets to be recovered from the following packets with redundant information so as to reduce the degree of content loss. Thirdly, the rate control mechanism selects the combination strategy of the main information and redundant information of the packets based on the network congestion state and the network loss rate of the packets in order to conquer the issue of the need of the network bandwidth.
This project effectively solves the problems of packet loss out of delivering over the Internet and of the combination strategy of the packets under several codecs. Network system developers can apply the mechanisms provided here to Internet Telephony, Internet Broadcast, and Video Conferencing.
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