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The ideal design for VoIP should be able to maximize the number of concurrent users while each admitted user enjoys the satisfactory quality of voice communication. In a recent packet-based wireless system, the quality-of-service (QoS) provisioning for VoIP raises several challenging issues including correlated delay performance among users and the rate-reliability tradeoff in managing multiple transmission rates on top of the inherent difficulty originating from the dependency of QoS on end-to-end delay. For these reasons, it is rarely found in the literature the work which presents the admission criteria for VoIP in an explicit manner. In this paper, our major contributions include 1) deriving the link-level QoS requirement from the E-model; 2) presenting the admission criteria in an explicit manner; 3) proposing a transmission rate adaptation algorithm specialized for VoIP application; and 4) illustrating the feasibility of our proposed schemes by getting the necessary information from the measurement. In addition to the theoretical justification, through the intensive simulation study in the Rayleigh fading environment, we prove that our joint designs composed of the rate adaptation, measurement-based admission control algorithm, and its extension to supporting the cell-boundary users indeed work well. Also, we evaluate the impact of various parameters including the average SNR, the Doppler rate, and the wireline delay.