VOIP动态缓出时延算法研究分析

在以包为单位进行数据传输合、语音应用程序(VOIP,Voice Over Internet Protocol)中,为了补偿数据包在网络传输中不可预知的网络传输延迟,在接收端首先必须把接收到的数据包缓存起来,缓存一定的时间再播放出来,以减少通话的抖动,得到比较满意的通话质量。文章主要研究动态缓出时延算法,力求使这个缓出时延尽可能小,同时尽可能减少包的丢失率。文章提出了一个有效动态缓出时延算法,该算法主要跟踪最近到达的数据包的网络传输时延求出其近似分布函数,并利用这些信息和延迟峰的侦测算法预测下一个语音峰的缓出时延。实验结果表明利用该算法可以在缓出时延和包丢失率之间达到最佳平衡,是一种理想、有效的算法。

A Packet Audio Playout Delay Adjustment Algorithm Analysis

In packet audio applications,packets are buffered at a receiving site and their playout delayed in order to compensate for variable network delays and obtain desirable communication quality.This paper considers the problem of adaptively adjusting the playout delay in order to keep the delay as small as possible,while at the same time avoiding excessive "loss "due to the arrival of packets at the receiver after their playout time has already passed.lt presents a new adaptive delay adjustment algorithm that tracks the network delay of recently received packets and efficiently maintains delay percentile information.This information,together with a "delay spike"detection algorithm is used to dynamically adjust talkspurt playout delay.It shows that this algorithm exceeds existing delay adjustment algorithms over a number of measured audio delay traces and performs close to the theoretical optimum over a range of parameter values of interest.