The current Internet provides a single class best effort service. From an application's point of view, this service amounts in practice to providing channels with time-varying characteristics such as delay and loss distributions. One way to support real time applications such as interactive audio given this service is to use control mechanisms that adapt the audio coding and decoding processes based on the characteristics of the channels, the goal be gin to maximize the quality of the audio delivered to the destinations. In this paper, we describe and analyze a set of such control mechanisms. They include a jitter control mechanism and a combined error and rate control mechanism. These mechanisms have been implemented and evaluated over the Internet and the MBone. Experiments indicate that they make it possible to establish and maintain reasonable quality audioconferences even across fairly congested connections.