Jitter Buffer Analysis

Boris Oklander, M. Sidi
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引用次数: 26

Abstract

VoIP is rapidly growing and widely used real-time voice service. On the path through the packet-switched networks, the regularity of VoIP stream is impaired by routing, queuing, scheduling and serialization effects, consequently resulting in loss and delay jitter of packets. Achieving high quality real-time voice requires smoothing the delay jitter at the receiver which is generally done by means of jitter buffer mechanism. Although being an important VoIP element, the exact analysis of the jitter buffer is rather rare. In this paper we model the jitter buffer mechanism and carry out the analysis of its performance. The relations between such quantities as initial playout delay, delay jitter and loss are established. Then, these relations are used together with the voice quality evaluation methodologies MOS and E-model, to optimally choose the controlling parameters of the jitter buffer, in order to increase the perceived quality of the voice. The analytic results developed in this work are applicable for incorporation in playout algorithms to achieve better voice quality.
抖动缓冲分析
VoIP是一种发展迅速、应用广泛的实时语音业务。在通过分组交换网络的路径上,VoIP流受到路由、排队、调度和序列化效应的影响,导致报文丢失和时延抖动。实现高质量的实时语音需要平滑接收端的延迟抖动,通常通过抖动缓冲机制来实现。虽然是一个重要的VoIP元素,但对抖动缓冲的精确分析是相当罕见的。本文对抖动缓冲机制进行了建模,并对其性能进行了分析。建立了初始播放延时、延时抖动、损耗等物理量之间的关系。然后,将这些关系与语音质量评价方法MOS和E-model相结合,优化选择抖动缓冲器的控制参数,以提高语音感知质量。本工作的分析结果可应用于播放算法中,以获得更好的语音质量。
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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