摘要
隨著全球資訊網(World-Wide-Web)的蓬勃發展，短短數十年間，人們用於傳遞訊息之形式已演進為結合視訊(Video)及語音(Audio)之即時性多媒體互動服務。然而，網路系統的負擔及複雜度卻伴隨著此種高流量訊務的到來而增加，同時也導致封包遺失、封包毀損及封包傳送逾時等問題愈趨嚴重，進而影響視訊及語音服務的即時性與流暢性，更可能因流量過大而拖垮整個網路系統的運作。 為了解決上述問題，本論文使用一套於應用層控制網路視訊及語音資料傳輸之通訊協定，並實做於嵌入式開發平台上，透過設計有效之初始網路頻寬估測機制及動態速率調整機制，且配合硬體影像壓縮(H.263 CODEC)及硬體語音壓縮(G723.1 CODEC)，期望能有效地控制視訊及語音的傳輸資料量。經實測驗證後，此嵌入式網路視訊電話系統無論處於任何網路環境之下，皆可透過頻寬估測及速率調整機制而得到平順且即時之視訊品質，達到適應性速率控制的功能。
並列摘要
With the growing up of the popularity of WWW users in recent decades, the communication format for human being has changed from text message to real time multimedia services which include video and audio streams. However, problems of complexity and burden of network structure, packet loss, packet damage, and packet delay are rising with the growing video and audio traffics, and furthermore will destroy the basic function as a result of bursting network traffic. In order to solve the problem described before, we adopt a control mechanism for network video and audio data transmission and it was be implemented on embedded platform with H.263 Video Codec and G.723.1 Audio Codec. We expect this mechanism can work efficiently on controlling the video and audio transmission data size via the initial bandwidth estimation and dynamic throughput adjustment. After a series of verification, the result comes to a conclusion of the embedded video phone system with rate control can work correctly in any kind of network environment and performs the video quality smoothly.