Abstract

Jitter is one of the premier impairments which inflict a heavy damage on the quality of service in wireless VoIP. To solve this issue, this paper proposes a new quality-based jitter buffer algorithm. An adaptive windowing algorithm is introduced to dynamically adjust the window size which indicates the numbers of packets used to estimate the future network delay and loss rete. When receiving a voice packet, the receiver firstly uses the variable-size window algorithm to update the window size. Then, the histogram of delay is established according to the remaining packets in window. Finally, E-Model is applied to evaluate the speech quality based on delay histogram. By searching for the maximum speech quality, the optimal buffer delay is obtained. Owe to the variable-size statistical window, both the accuracy of network delay prediction and the ability to deal with spikes have further improved. The experiment results show that the whole VoIP communication under our proposed algorithm not only suffers the smallest average delay and lowest packet loss, but also achieves the highest speech quality.

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