Enhancing low speed network performance for voice over internet protocol

Y. Bandung, C. Machbub, A. Langi, S. Supangkat
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引用次数: 1

Abstract

In this paper, we optimize low speed networks access for designing an efficient Voice over Internet Protocol network, especially in 64 kbps, 128 kbps, and 256 kbps bandwidth capacities. We draw some analytical approach based on Extended E-model to quantify some levels of quality of service. Using the model, those levels are determined by network parameters such as delay, jitter, packet loss level, network utilization, and bandwidth capacity and by implementation configuration such as voice coder, packetization scheme, and size of jitter buffer. Our objective is to find maximum number of calls in some given bandwidth capacities while maintaining a certain level of the quality of service. Using numeric estimation, we found that the optimum solution for some given bandwidth capacities can be achieved by applying G.723.1 5.3 kbps voice coder, packet loss level less than 1%, jitter less than 80 ms, and utilization network less than 80%.
增强互联网协议语音的低速网络性能
在本文中,我们优化了低速网络接入,以设计一个高效的基于互联网协议的语音网络,特别是在64kbps, 128kbps和256kbps的带宽容量下。本文提出了一种基于扩展e模型的分析方法来量化服务质量的各个层次。使用该模型,这些级别由网络参数(如延迟、抖动、丢包级别、网络利用率和带宽容量)和实现配置(如语音编码器、分组方案和抖动缓冲区大小)决定。我们的目标是在一定带宽容量下找到最大的呼叫数,同时保持一定水平的服务质量。通过数值估计,我们发现对于某些给定的带宽容量,采用G.723.1 5.3 kbps语音编码器,丢包率小于1%,抖动小于80 ms,网络利用率小于80%可以实现最佳解决方案。
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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