Our goal is to extend recommendations ITU-T G.114 and G.131 to cover distorted phone calls transported (partly) over a packet-based network. We assume the user terminals to be optimally tuned and focus on how network parameters-delay, packet loss, jitter, and so on-influence voice quality. We then discuss how those parameters can be quantified and incorporated into a model that lets us predict the quality of any packet-based phone call.
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Janssen et al. (2002) studied this question.
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