Abstract
In this work we address the problem of statistically multiplexing a variable number of telephone calls via a limited number of channels. Terminals operate with voice activity and silence detectors, and the speech is encoded to a bit rate which is system state dependent. Growing from zero, calls in the system are admitted with a maximum bit rate (maximum quality) until the cutoff fraction of talkspurt (a front end clipping) reaches a certain threshold. At this point the voice bit rate of all transmitting terminals is reduced. If due to traffic fluctuations, the number of calls in progress decreases, then all voice terminals are allowed to operate at the maximum bit rate again. In order to avoid annoying effects to listeners, both the percentage of voice information that is lost and the mean number of changes in the bit rate per mean call holding time are constrained. The first constraint is strongly dependent on the encoding bit rate, and the second one is controlled by using a hysteresis threshold when switching from one bit rate to another. In this work we have used three encoding bit rates, high, medium and low. A birth–death Markov process is used to model the system, which provided exact numerical evaluations for the percentage of time the system operates in each encoding bit rate and for the mean number of changes in the bit rate. Metrics are defined to measure the percentage of both types of voice information that are lost (not transmitted), cutoff or front end clipping and uniform dropping. Finally, an illustrative example is reported.
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Casares Giner, V. Variable Bit Rate Voice Using Hysteresis Thresholds. Telecommunication Systems 17, 31–62 (2001). https://doi.org/10.1023/A:1016699902484
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DOI: https://doi.org/10.1023/A:1016699902484