Study of enhanced VoIP performance under congested wireless network scenarios

A. Mukhopadhyay, T. Chakraborty, S. Bhunia, I. S. Misra, S. Sanyal
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引用次数: 11

Abstract

Stringent QoS maintenance in a wireless environment for VoIP communication is a major challenge. VoIP inherently generates constant bit rate traffic and is highly sensitive to network delay. However, unpredictable network congestion makes a VoIP session so degraded that its QoS goes below a tolerable limit. Accordingly, a suitable solution is needed to adapt varying network conditions satisfying minimum QoS while maintaining transparency to the end user. We have extensively studied the effect of variable voice packet payload size with changing number of voice sample frames in the payload of the RTP packets. For this, we have used OPNET Modeler 14.5.A to observe the performance in terms of MOS, End-to-end delay through extensive simulations. Results are provided with valid discussions. Based on the observation, we propose an Adaptive VoIP (AdVoIP) algorithm that may be used in any real life VoIP network for further enhancement of the performance. The adaptive algorithm may use RTCP Receiver Reports to assess the network conditions in real wireless scenarios.
拥塞无线网络场景下增强VoIP性能的研究
在VoIP通信的无线环境中严格的QoS维护是一个主要挑战。VoIP固有地产生恒定的比特率流量,并且对网络延迟高度敏感。然而,不可预测的网络拥塞使VoIP会话降级到其QoS低于可容忍的限制。因此,需要一个合适的解决方案来适应不同的网络条件,在满足最低QoS的同时保持对最终用户的透明性。我们广泛地研究了语音包有效载荷大小随RTP包有效载荷中语音样本帧数的变化而变化的影响。为此,我们使用了OPNET Modeler 14.5。通过大量的仿真来观察MOS、端到端延迟方面的性能。结果提供了有效的讨论。在此基础上,我们提出了一种自适应VoIP (AdVoIP)算法,该算法可用于任何现实生活中的VoIP网络,以进一步提高性能。自适应算法可以利用RTCP接收方报告来评估真实无线场景下的网络状况。
本文章由计算机程序翻译,如有差异,请以英文原文为准。
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