This work studies the jitter buffer management algorithm for Voice over IP in WebRTC. In particular, it details the core concepts of WebRTC’s jitter buffer management. Furthermore, it investigates how jitter buffer management algorithm behaves under network conditions with packet bursts. It also proposes an approach, different from the default WebRTC algorithm, to avoid distortions that occur under such network conditions. Under packet bursts, when the packet buffer becomes full, the WebRTC jitter buffer algorithm may discard all the packets in the buffer to make room for incoming packets. The proposed approach offers a novel strategy to minimize the number of packets discarded in the presence of packet bursts. Therefore, voice quality as perceived by the user is improved. ITU-T Rec. P.863, which also confirms the improvement, is employed to objectively evaluate the listening quality.

中文翻译：

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改进了 WebRTC 的抖动缓冲区管理

本工作研究WebRTC中IP语音的抖动缓存管理算法。特别是，它详细介绍了 WebRTC 抖动缓冲区管理的核心概念。此外，它还研究了抖动缓冲区管理算法在具有数据包突发的网络条件下如何表现。它还提出了一种不同于默认 WebRTC 算法的方法，以避免在此类网络条件下发生失真。在数据包突发的情况下，当数据包缓冲区变满时，WebRTC 抖动缓冲区算法可能会丢弃缓冲区中的所有数据包，为传入的数据包腾出空间。所提出的方法提供了一种新颖的策略，以最小化在存在数据包突发的情况下丢弃的数据包数量。因此，提高了用户感知的语音质量。ITU-T 建议书 P.863，也证实了改进，