Optimal Adaptive Voice Smoother for New Generation VoIP Services

Abstract

VoIPs, one of emerging technologies, offer high quality of real time voice services over IP-based broadband networks. Perceived voice quality is a key metric for VoIP applications. It is mainly affected by IP network impairments such as delay, jitter and packet loss. Playout buffer at the receiving end can compensate for the effects of jitter based on a tradeoff between delay and loss. Adaptive smoothing algorithms are capable of adjusting dynamically the smoothing size by introducing a variable delay based on the network parameters to avoid the quality decay problem. This dissertation introduces efficient and feasible perceived quality methods 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 method, and ITU-T E-model approaches 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.

Key words: Adaptive voice smoother, VoIP, buffer re-synchronization, delay/loss trade off, E-model