- 學位論文
具動態頻寬估測之網路視訊電話實做
Implementation of Video Phone with Dynamic Bandwidth Estimation
摘要
由於寬頻網路的普及化，使得與多媒體相關的網路應用更加盛行，其中之一為相當熱門的網路視訊電話應用。現實網路環境無法保證在任何時刻皆可提供穩定且順暢的頻寬傳送，且有封包的遺失、毀損、延遲等問題，因此如何在網路上穩定且順暢的傳送視訊語音資料成為發展網路電話的一個重要問題。 針對上述問題，本論文實作具動態頻寬估測機制的網路電話，並結合H.263 Codec與Speex Codec進行視訊語音的壓縮。在進行視訊電話時，視訊接收端每隔一段時間會收到視訊傳送端的探測封包，經由探測封包內的資訊去估測出目前可利用的頻寬，再將估測結果回饋給視訊傳送端，視訊傳送端利用此回饋資訊去調整視訊語音的編碼模式與封包傳送速度，藉此降低封包的遺失、毀損、延遲的機率，如此可以使視訊語音傳送品質維持穩定且順暢，讓使用者有較佳的網路電話體驗。並經由不同網路環境做測試，如ADSL、Cable、廣域網路、區域網路等連接方式，此機制可調整出適應的視訊傳送品質與頻寬量，達到適應性速度控制的功能。
並列摘要
Since the popularization of broadband networks make related to the internet and multimedia applications more prevalent. One of these is a popular video phone applications. Real network environment can not guarantee to provide stable and smooth bandwidth transmission at any moment. Furthermore, there are some problems of packet loss, packet damage, packet delay. How to have stable and smooth transmission of video and voice data becomes an important issue for the video phone of the network. In this paper, the implement of video phone with dynamic bandwidth estimation combines H.263 Video Codec and Speex Audio Codec for video and voice compression. When dealing a call , the receiver will receive detective packets of sender for each period. By the information of probe packet to estimate the available bandwidth, the receiver send results back to the sender so that the sender can modulate modes of encoding and bit rates of the packets to reduce the packet loss rate, packet damage rate, packet delay rate. We can maintain stable transmission of video and voice. Users will have a better video phone service. After testing for available network environment, such as ADSL, Cable, WAN, and LAN, this mechanism can readjust the adaptive video transmission quality and bandwidth capacity to achieve the adaptive rate control.