We consider the problem of distributing audio data over networks such as the Internet that do not provide support for real-time applications. Experiments with such networks indicate that audio quality is mediocre in large part because of excessive audio packet losses. In this paper, we show using measurements over the Internet as well as analytic modeling that the number of consecutively lost audio packets is small unless the network load is very high. This indicates that open loop error control mechanisms based on forward error correction would be adequate to reconstruct most lost audio packets.