VoIP会话质量的整体研究-控制的旋钮

R. V. Prasad, V. Rao, H. N. Shankar, P. Pawełczak, R. Muralishankar, I. Niemegeers
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引用次数: 0

摘要

VoIP数据包在互联网上传输时,会经历丢失和可变延迟。网络的效果不仅取决于后台流量,还取决于VoIP报文本身的参数,如VoIP报文大小、报文生成间隔等。虽然较大大小的数据包经历更多的丢失,但它们经历较少的延迟抖动,因此在播放缓冲区中处理它们很容易。为了全面研究各种网络条件对VoIP会话的影响,我们提出了一个完整的端到端研究,考虑了底层网络的各种状态。我们提出了一个G.711编码数据包的案例研究,以20和40毫秒的间隔产生比较。当网络加载时,传输20ms数据的数据包更好,而当网络不饱和时,传输40ms数据的数据包更受欢迎。这会影响抖动和损失,从而影响质量。我们用平均意见得分来解释这种权衡。
本文章由计算机程序翻译,如有差异,请以英文原文为准。
A Holistic Study of VoIP Session Quality - The Knobs that Control
VoIP packets, when transported over the Internet, experience loss and variable delay. The effect of the network not only depends on the background flows but also on the parameters of VoIP packets itself, such as VoIP packet size and the packet generation intervals. While higher sized packets experience more losses, they experience less delay jitter and handling them is thus easy at the playout buffer. To investigate the effect of various network conditions on VoIP session holistically, we present a complete end to end study considering various states of the underlying network. We present as a case study of G.711 coded packets generated at 20 and 40 ms intervals for comparison. While packets carrying 20 ms data are better when the network is loaded, 40 ms packetization is favored when the network is not saturated. This affects the jitter and loss thus affecting the quality. We explain this trade-off using mean opinion scores.
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