Enhancing the Qos of a Voip Call Using an Adaptive Jitter Buffer Playout Algorithm with Variable Window Size

Transmitting real-time voice over the Internet is a technological challenge. Variation in network characteristics introduces jitter to the propagating voice packets. Jitter hampers voice quality and makes the VoIP call uncomfortable to the user. Often buffers are used to store the received packets for a short time before playing them at equal spaced intervals to minimize jitter. Choosing optimum buffering time is essential for reducing the added end-to-end delay and number of discarded packets. In this paper, some established adaptive jitter buffer playout algorithms have been studied and a new algorithm has been proposed. The network used for analyzing the algorithms has been simulated using OPNET modeler 14.5.A. Further studies have been conducted for finding the optimum sliding window size for the proposed algorithm. The proposed algorithm kept jitter within a tolerable limit along with significant reduction of delay and loss compared to other algorithms analyzed in this paper.