Adaptive Rate Control for Aggregated VoIP Traffic

F. Sabrina, J. Valin
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引用次数: 14

Abstract

This paper presents a novel mechanism for dynamically adapting the quality of congestion controlled voice over IP (VoIP) applications on the Internet in real time. The system uses our proposed variable bit rate speech codec called Speex, which can dynamically adjust the encoding bit rate (and hence the speech quality) based on both the feedback information about the network congestion and the instantaneous speech properties. Our extensive NS2 simulation results prove that the proposed system indeed provides highest quality speech while maximising the bandwidth utilisation and reducing the network congestion.
聚合VoIP流量的自适应速率控制
本文提出了一种实时动态适应网络上IP拥塞控制语音(VoIP)应用质量的机制。该系统使用我们提出的可变比特率语音编解码器Speex,它可以根据网络拥塞和瞬时语音特性的反馈信息动态调整编码比特率(从而调整语音质量)。我们广泛的NS2仿真结果证明,所提出的系统确实提供最高质量的语音,同时最大限度地提高带宽利用率和减少网络拥塞。
本文章由计算机程序翻译,如有差异,请以英文原文为准。
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