Excessive packet loss rates can dramatically decrease the audio quality perceived by users of Internet telephony applications. Previous results suggest that error control schemes using forward error correction (FEC) are good candidates for decreasing the impact of packet loss on audio quality. However, the FEC scheme must be coupled to a rate control scheme. Furthermore, the amount of redundant information used at any given point in time should also depend on the characteristics of the loss process at that time (it would make no sense to send much redundant information when the channel is loss free), on the end to end delay constraints (destination typically have to wait longer to decode the FEC as more FEC information is used), on the quality of the redundant information, etc. However, it is not clear given all these constraints how to choose the "best" possible redundant information. We address this issue, and illustrate the approach using an FEC scheme for packet audio standardized in the IETF. We show that the problem of finding the best redundant information can be expressed mathematically as a constrained optimization problem for which we give explicit solutions. We obtain from these solutions a simple algorithm with very interesting features, namely (i) the algorithm optimizes a subjective measure (such as the audio quality perceived at a destination) as opposed to an objective measure of quality (such as the packet loss rate at a destination), (ii) it incorporates the constraints of rate control and playout delay adjustment schemes, and (iii) it adapts to varying loss conditions in the network (estimated online with RTCP feedback). We have been using the algorithm, together with a TCP-friendly rate control scheme and we have found it to provide very good audio quality even over paths with high and varying loss rates. We present simulation and experimental results to illustrate its performance.

Much of the work on distributed virtual environments (DVEs) has focused on efficient schemes and algorithms for the description and visual rendering of such environments. However, there has been comparatively little effort on senses other than sight, and in particular on live voice. This is surprising because adding voice to DVEs enhances the sense of immersion in the environment, and can in fact completely change the way entities interact. For example, adding voice in a distributed game impacts tactics, team building, cheating, etc. We examine issues related to adding voice between participants in DVEs. We consider in particular a special kind of DVE, namely distributed games over the Internet. We consider all stages of voice manipulation, including voice generation (with emphasis on echo cancellation), voice transmission (with emphasis on RTP and packetization), and voice restitution (with emphasis on spatial rendition and on synchronization between voice and visual cues). We also consider implementation issues, and illustrate these with the MiMaze game and the FreePhone audio tool, both developed at INRIA.