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On the impact of loss and delay variation on Internet packet audio transmission

机译:丢失和延迟变化对Internet分组音频传输的影响

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The quality of audio in IP telephony is significantly influenced by various factors, including type of encoder, delay, delay variation, rate and distribution of packet loss, and type of error concealment. Hence, the performance of IP telephony systems is highly dependent on understanding the contribution of these factors to audio quality, and their impact on adaptive transport mechanisms such as error and buffer control. We conducted a large-scale audio transmission experiment over the Internet in a 12-month-period in order to evaluate the effects and the correlation of such parameters on audio transmission over IP. We have noticed that the correlation of loss and delay is not linear, but stronger correlation is observed as the delay approaches certain thresholds. We have made a number of new observations on various delay thresholds that are significant for loss prediction for adaptive audio transmission over IP networks. We also have made new observations to assess the audio quality of PCM μ-law and G.728 codecs under different loss and delay conditions. The paper provides a number of recommendations for implementing efficient adaptive FEC mechanisms based on our measurement observations and analysis.
机译:IP电话中的音频质量受各种因素的影响很大,包括编码器的类型,延迟,延迟变化,丢包率和分布以及错误隐藏的类型。因此,IP电话系统的性能高度依赖于理解这些因素对音频质量的影响,以及它们对诸如错误和缓冲区控制之类的自适应传输机制的影响。为了评估这些参数对IP上音频传输的影响和相关性,我们在12个月内进行了一次大规模的Internet音频传输实验。我们已经注意到损失和延迟的相关性不是线性的,但是随着延迟接近某些阈值,观察到了更强的相关性。我们已经对各种延迟阈值进行了许多新观察,这些阈值对于IP网络上自适应音频传输的损耗预测非常重要。我们还做出了新的观察,以评估在不同丢失和延迟条件下PCMμ律和G.728编解码器的音频质量。本文根据我们的测量观察和分析,为实现有效的自适应FEC机制提供了许多建议。

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