We develop a joint playout buffer and Forward Error Correction (FEC) adjustment scheme for Internet Telephony, which incorporates the impact of end-to-end delay on the perceived audio quality. We show that it provides better quality than the adjustment schemes for playout buffer and FEC that were previously published. This is important because of a threshold effect when the end-to-end delay of interactive audio is around 150 ms. We represent the perceived audio quality as a function of both the end-to-end delay and the distortion of the voice signal. We develop a joint rate/error/playout delay control algorithm that optimizes this measure of quality and is TCP-Friendly. It uses a channel model for both loss and delay. We validate our approach by simulation and show that (1) our scheme allows a source to increase its utility by avoiding an increase of the playout delay when it is not really necessary, (2) it performs better than direct combinations of existing algorithms in the cases where end-to-end delay is important and (3) adaptive delay aware FEC adjustment brings significant improvements only if it is coupled with an adaptive playout adjustment.