Delay aspects in Internet telephony

In this work, we address the transport of high quality voice over the Internet with a particular concern for delays. Transport of interactive audio over IP networks often suffers from packet loss and variations in the network delay (jitter). Forward Error Correction (FEC) mitigates the impact of packet loss at the expense of an increase of the end-to-end delay and the bit rate requirement of an audio source. Furthermore, adaptive playout buffer algorithms at the receiver compensate for jitter, but again this may come at the expense of additional delay. As a consequence, existing error control and playout adjustment schemes often have end-to-end delays exceeding 150 ms, which significantly impairs the perceived quality, while it would be more important to keep delay low and accept some small loss. We develop a joint playout buffer and FEC adjustment scheme for Internet Telephony that incorporates the impact of end-to-end delay on perceived audio quality. To this end, we take a utility function approach. 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 which 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 increasing the playout delay when it is not really necessary and (2) it provides better quality than the adjustment schemes for playout and FEC that were previously published. We use this scheme in the framework of non-elevated services which allow applications to select a service class with reduced end-to-end delay at the expense of a higher loss rate. The tradeoff between delay and loss is not straightforward since audio sources may be forced to compensate the additional losses by more FEC and hence more delay. We show that the use of non-elevated services can lead to quality improvements, but that the choice of service depends on network conditions and on the importance that users attach to delay. Based on this observation, we propose an adaptive service choosing algorithm that allows audio sources to choose in real-time the service providing the highest audio quality. In addition, when used over the standard IP best effort service, an audio source should also control its rate in order to react to network congestion and to share the bandwidth in a fair way. Current congestion control mechanisms are based on packets (i.e., they aim to reduce or increase the number of packets sent per time interval to adjust to the current level of congestion in the network). However, voice is an inelastic traffic where packets are generated at regular intervals but packet size varies with the codec that is used. Therefore, standard congestion control is not directly applicable to this type of traffic. We present three alternative modifications to equation based congestion control protocols and evaluate them through mathematical analysis and network simulation

Realisation of an Adaptive Audio Tool

Real-time audio over the best effort Internet often suffers from packet loss. At this time, Forward Error Correction (FEC) seems to be an efficient way to attenuate the impact of loss. Nevertheless to ensure efficiency of FEC, the source rate must be continuously controlled to avoid congestion. In this paper, we describe a realisation of adaptive FEC subdued to a TCP-friendly rate control

Global Fairness of additive-increase and multiplicative-decrease with heterogeneous round-trip times

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 $F_A^h$ (``$F_A^h$ 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 the parking lot scenario, and comparing with previous results \cite{floyd-91-b,mathis-97-a}, and an extensive numerical simulation with realistic parameter settings. We apply $F_A^h$ fairness to gain a more accurate understanding of the bias of TCP against long round trip times

End-to-end congestion control for tcp-friendly flows with variable packet size

Current TCP-friendly congestion control mechanisms adjust the packet rate in order to adapt to network conditions and obtain a throughput not exceeding that of a TCP connection operating under the same conditions. In an environment where the bottleneck resource is packet processing, this is the correct behavior. However, if the bottleneck resource is bandwidth, and flows may use packets of different size, resource sharing depends on packet size and is no longer fair. For some applications, such as Internet telephony, it is more natural to adjust the packet size, while keeping the packet rate as constant as possible. In this paper we study the impact of variations in packet size on equation-based congestion control and propose methods to remove the resulting throughput bias. We investigate the design space in detail and propose a number of possible designs. We evaluate these designs through simulation and conclude with some concrete proposals. Our findings can be used to design a TCP-friendly congestion control mechanism for applications that adjust packet size rather than packet rate, or applications that are forced to use a small packet size

Realisation of an Adaptive Audio Tool

Real-time audio over the best effort Internet often suffers from packet loss. At this time, Forward Error Correction (FEC) seems to be an efficient way to attenuate the impact of loss

A Note on the Fairness of TCP Vegas

We study the fairness of TCP Vegas. The latter is an alternative to the commonly used TCP Reno, and uses measures of the round trip time as feedback on congestion. We consider two cases that depend on the value of the two parameters alpha and beta controlling the window sizes` update. Our main conclusion is that TCP Vegas is unfair in several points. First, when alpha = beta, if the propagation delays are correctly estimated, TCP Vegas is known to be fair. However we show that any over-estimation of the propagation delay of a given connection results in an increase of its rate and hence leads to unfairness. This rate increase augments with the over-estimation factor. Moreover, the rate oscillations, whose amplitude increases with the rate value, are not sufficient to provide an accurate estimation of the propagation delay. Second, when alpha < beta, TCP Vegas is unfair even if the propagation delays are correctly estimated. In this case, the rate of a connection converges to a stable value that depends on the arrival order of all connections so that earliest established connections get more bandwidth. Also, in a more realistic scheme, later connections see their propagation delay over-estimated and thus they gain larger portion of the bandwidth. These two effects tend to counterbalance each other but the second tends to dominate. Future use of TCP Vegas in the context of TCP-friendly applications, should therefore rely on alpha = beta, but will require the propagation delays to be correctly estimated. Yet, this seems to be quite hard to achieve

Impact of link failures on VoIP performance

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