TCP-friendly transmission of voice over IP

In the last few years an increasing amount of attention has been paid to technologies for the transmission of voice over IP (VoIP). At present, the UDP transport protocol is used to provide this service. However, when the same bottleneck link is shared with TCP flows, and in the presence of a high network load and congestion, UDP sources capture most of the bandwidth, strongly penalizing TCP sources. To solve this problem some congestion control should be introduced for UDP traffic as well, in such a way that this traffic becomes " TCP-friendly ". In this perspective, several TCP-friendly algorithms have been proposed in the literature. Among them, the most promising candidates for the immediate future are RAP and TFRC. However, although these algorithms were introduced to support real-time applications on the Internet, up to now the only target in optimizing them has been that of achieving fairness with TCP flows in the network. No attention has been paid to the applications using them, and in particular, to the quality of service (QoS) perceived by their users. The target of this paper is to analyze the problem of transmitting voice over IP when voice sources use one of the above-mentioned TCP-friendly algorithms. With this aim, a VoIP system architecture is introduced and the characteristics of each its elements are discussed. To optimize the system, a multirate voice encoder is used to be feasible to work over a TCP layer, and a modification of both RAP and TFRC is proposed. Finally, in order to analyze the performance of the proposed system architecture and to compare the modified RAP and TFRC with the original algorithms, the sources have been modeled with an arrival process modulated by a Markov chain, and the model has been used to generate traffic in a simulation study performed with the ns-2 network simulator.