A joint playout buffer and forward error correction (FEC) adjustment scheme are developed 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.

We use active and passive traffic measurements to identify the issues involved in the deployment of a voice service over a tier-1 IP backbone network. Our findings indicate that no specific handling of voice packets (i.e. QoS differentiation) is needed in the current backbone but new protocols and mechanisms need to be introduced to provide a better protection against link failures. We discover that link failures may be followed by long periods of routing instability, during which packets can be dropped because forwarded along invalid paths. We also identify the need for a new family of quality of service mechanisms based on fast protection of traffic and high availability of the service rather than performance in terms of delay and loss.

Consider a network with an arbitrary topology and arbitrary communication delays, in which congestion control is based on additive-increase and multiplicative-decrease. We show that the source rates tend to be distributed in order to maximize an objective function called F/sub A//sup h/ ("F/sub A//sup h/ fairness"). We derive this result under the assumption of rate proportional negative feedback and for the regime of rare negative feedback. This applies to TCP in moderately loaded networks, and to those TCP implementations that are designed to interpret multiple packet losses within one RTT as a single congestion indication and do not rely on re-transmission timeout. This result provides some insight into the distribution of rates, and hence of packet loss ratios, which can be expected in a given network with a number of competing TCP or TCP-friendly sources. We validate our findings by analyzing a multiple-bottleneck scenario, and comparing with previous results (Floyd, 1991, Mathis et al, 1997) and an extensive numerical simulation with realistic parameter settings. We apply F/sub A//sup h/ fairness to gain a more accurate understanding of the bias of TCP against long round-trip times.