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Perceived voice quality is a key metric in VoIP applications. The quality is mainly affected by IP network impairments such as delay, jitter and packet loss. Playout buffer at the receiving end can be used to compensate for the effects of jitter based on a tradeoff between delay and loss. Adaptive smoothing algorithms are capable of adjusting dynamically the smoothing time based on the network parameters to improve voice quality. In this article, we introduce an efficient and easy perceived quality method for buffer optimization to archive the best voice quality. This work formulates an online loss model which incorporates buffer sizes and applies the Lagrange multiplier approach to optimize the delay-loss problem. Distinct from the other optimal smoothers, the proposed optimal smoother is suitable for any codec and carries the lowest complexity. Simulation experiments validate that the proposed adaptive smoother archives significant improvement in the voice quality. [ABSTRACT FROM AUTHOR]

Perceived voice quality is mainly affected by IP network impairments such as delay, jitter and packet loss. Adaptive smoothing buffer at the receiving end can compensate for the effects of jitter based on a tradeoff between delay and loss to archive a best voice quality. This work formulates an online loss model which incorporates buffer sizes and applies the ITU-T E-model approach to optimize the delay-loss problem. Distinct from the other optimal smoothers, the proposed optimal smoother suitable for most of codecs carries the lowest complexity. Since the adaptive smoothing scheme introduces variable playback delays, the buffer re-synchronization between the capture and the playback becomes essential. This work also presents a buffer re-synchronization algorithm based on silence skipping to prevent unacceptable increase in the buffer preloading delay and even buffer overflow. Simulation experiments validate that the proposed adaptive smoother archives significant improvement in the voice quality. [ABSTRACT FROM AUTHOR]