Voice over IP (VoIP) supports us to make free of charge or very cheap soft-phone and hard-phone calls locally and globally. VoIP is moving on Wireless Local Area Networks (WLANs) based on IEEE 802.11 standards. For audio speech quality in packet switch applications, the main concerns are end to end delay and packet loss. This paper presents a VoIP-Telephony deployment model using Opnet Simulator. We have configured an experimental setup and analyzed various methods to improve voice quality. Experimental results using IEEE 802.11a/b were analyzed for voice traffic. Experimental results with varying packet loss using Jperf and distributed coordination function (DCF-Voice) presented.
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Nisar et al. (2010) studied this question.
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