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VoIP Playout Buffer Adjustment using Adaptive Estimation of Network Delays

机译:使用网络延迟的自适应估计进行VoIP播放缓冲区调整

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The poor quality of Voice over IP can be improved by adaptive playout buffering at the receiver. This technique dynamically adapts the playout deadline to network conditions, thus minimizing both late packet loss and buffering time. A standard playout buffer strategy uses an estimate (Exponentially Weighted Moving Average) of the mean and variance of network delay to set the playout deadline. This estimation is characterized by a fixed, constant weighting factor. We show that tuning of this parameter so that the strategy works very well for all network conditions is not feasible. Therefore we propose to extend this standard buffer strategy by replacing the fixed, constant weighting factor with a dynamic one. In our solution, the weighting factor is dynamically adjusted according to the observed delay variations. When these variations are high (which implies that the network conditions are changing), the parameter is set low, and vice-versa. This allows rapid adaptation to network variations and reduces the frequency of late packets (or buffering time). Simulations and experimental results show that with our strategy, the trade-off between buffering delay and late packet loss at the receiver is improved significantly.
机译:IP语音质量较差,可以通过在接收器处进行自适应播放缓冲来解决。这项技术可根据网络条件动态调整播出期限,从而将延迟的数据包丢失和缓冲时间降到最低。标准播出缓冲策略使用网络延迟的均值和方差的估计值(指数加权移动平均值)来设置播出期限。该估计的特征在于固定的,恒定的加权因子。我们表明,对该参数进行调整以使该策略在所有网络条件下均能很好地工作是不可行的。因此,我们建议通过用动态变量代替固定的,恒定的加权因子来扩展此标准缓冲策略。在我们的解决方案中,加权因子是根据观察到的延迟变化动态调整的。当这些变化很大时(这意味着网络状况正在改变),则将参数设置为低,反之亦然。这样可以快速适应网络变化,并减少后期数据包的频率(或缓冲时间)。仿真和实验结果表明,采用我们的策略,显着改善了接收器的缓冲延迟和延迟丢包之间的折衷。

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