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Enhancing the QOS of A Voip call Using an Adaptive Jitter Buffer Playout Algorithm with Variable Window Size

ISSN : 1839-5678
Journal from gdlhub / 2017-08-14 11:52:34
Oleh : Atri Mukhopadhyay, Tamal Chakraborty, Iti Saha Misra and Salil Kumar Sanyal, International Journal of Mobile Network Communications & Telematics
Dibuat : 2012-07-04, dengan 1 file

Keyword : VoIP, QoS parameters, Adaptive Algorithm, Congested Network, Jitter Buffer
Subjek : Enhancing the QOS of A Voip call Using an Adaptive Jitter Buffer Playout Algorithm with Variable Window Size
Url : http://airccse.org/journal/ijmnct/papers/2312ijmnct06.pdf
Sumber pengambilan dokumen : Internet

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.

Deskripsi Alternatif :

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.

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