Abstract
Although the widespread of broadband network has boosted the deployment of voice over IP (VoIP) systems, most part of the telephone service is still provided on the traditional PSTN lines. It can be expected that the transition from traditional telephone to VoIP systems proceed, and that the network traffic due to the phone calls will become significant. Hence, it is still important to select the best protocols, both for traffic forwarding and for encoding to ensure the required quality. In this paper we present the results of a series of experiments to evaluate the protocols involved in VoIP service. On a laboratory testbed we reproduce a common situation of two interconnected branches and compare the performance achievable by selecting different network protocols like SIP and IAX combined with current voice codecs like G.711 and G.729.
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