We elaborate on provisioning adaptive rate to IP telephony applications through variable bit-rate coding algorithms and use a selective dropping mechanism to make the deadline misses distributed evenly during the congestion period. This paper presents simulation results outlining the behavior of rate-adaptive voice communications over a Differentiated Service IP network. An algorithm is proposed for driving the transmission rate of voice sources on the basis of estimations of the network conditions, measured in terms of packet delays and losses, using the average queue size in random early detection (RED) as an effective mechanism to control the congestion in the network. The performance of the introduced algorithm is verified by comparing the performance of simple drop-tail (FIFO) queuing, and RED in a DiffServ enabled network with NS2.