Adaptive playout algorithm using packet expansion for the VoIP

Abstract

Internet telephony means placing telephone call over the Internet which providing "best effort" service only instead of Public Switched telephone networks. Because of this, Internet telephony cannot guarantee QoS. In this paper, we propose a new algorithm to reduce the packet loss and to compensate the jitter which are factors affection QoS called the Frame Expansion for Adaptive Playout Time algorithm. This algorithm is suitable for soft real-time applications which can tolerate a certain amount of packet loss and delay. This is because it has low packet loss and delay regardless of some playout delay caused by expanding the received frame size using Synchronized Overlap and Add algorithm in receiver side. In order to analyze and evaluate the performance of proposed algorithm, we carry out a simulation of delayed packet loss rate and playout delay, which is computed by taking the difference between the playout time and the arrival time on receiver side. From the simulation, our proposed algorithm can reduce the probability of packet loss considerably, and improve the quality of the voice compared with existing playout buffering algorithm. © Springer-Verlag Berlin Heidelberg 2003.