Abstract
The QoS standard of a VoIP session degrades if its stringent time requirements are not met. Low end-to-end delay of the voice packets and low packet loss must be maintained. Jitter between voice packets must also be within tolerable limits. Jitter hampers voice quality and makes the VoIP call uncomfortable to the user. Very often, buffers are used to store the received packets for a short time before playing them at equal spaced intervals to minimize jitter. However, this introduces the problem of added end-to-end delay and 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 the analysis of the algorithms has been simulated using OPNET modeler 14.5.A. The proposed algorithm kept jitter within a tolerable limit along with drastic reduction of delay and loss compared to other algorithms analyzed in this paper.
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Mukhopadhyay, A., Chakraborty, T., Bhunia, S., Misra, I.S., Sanyal, S.K. (2011). An Adaptive Jitter Buffer Playout Algorithm for Enhanced VoIP Performance. In: Wyld, D.C., Wozniak, M., Chaki, N., Meghanathan, N., Nagamalai, D. (eds) Advances in Computing and Information Technology. ACITY 2011. Communications in Computer and Information Science, vol 198. Springer, Berlin, Heidelberg. https://doi.org/10.1007/978-3-642-22555-0_24
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DOI: https://doi.org/10.1007/978-3-642-22555-0_24
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