Video conferencing can be used for demanding applications such as playing music together over networks, and other distributed multimedia. This application is very demanding with respect to end-to-end delay, and uncompressed audio and video transmission has been used in demonstrations in order to minimize delays. The purpose of this project is to explore possibilities between this raw format with its low latency and high bandwidth requirement, and standard video conferencing formats with higher latencies but efficient compression. Central questions will be the dependence on the application of the involved compromises, which includes packet loss handling, end-to-end delay, and audio quality. Experimental software for audio streaming, as well as video streaming, is developed. Subjective assessments are used in this project to evaluate the perceived quality.

Project coordinator: Professor Peter Svensson