'Institute of Electrical and Electronics Engineers (IEEE)'
Abstract
The quality of Voice over Internet Protocol (VoIP) calls is highly influenced by transmission impairments such as delay, packet loss and jitter, with jitter being manifested as one of the deleterious effects affecting its quality. A jitter buffer is usually employed at the receiver side to mitigate its effects by adapting its parameters in a trade-off between delay and packet loss. This paper proposes a novel de-jitter algorithm that adaptively changes the size of the playout buffer depending on the network states, in order to effectively handle the packet loss and delay, whereas E-model is used to quantify speech quality. Based on the statistics of the received packets, the adaptive playout buffer algorithm dynamically adjusts the weighting factor (α) and the safety factor (β) for regulating the delay and trade-off loss, thus maximizing the quality for VoIP
