Joe will present during the first talk of the academic year.
Many things that people do together rely on time synchronisation. When using videoconferencing software this all falls apart because of audio delays caused by network latency. Networking people try to reduce latency by improving protocols and connectivity, but there are fundamental limits such as the speed of light, audio and network buffering etc.In this talk I suggest that rather than incrementally improve performance through networking and audio processing advances, we could instead harness the fact that what we're transmitting is human behaviour and thus may be predictable in advance. With this approach, we can plausibly aim for zero latency, where audio communication appears similar to if the participants were in the same place. I'll talk through a bunch of possible ways to do this...
University of Nottingham School of Computer Science Nottingham, NG8 1BB
email: mrl@cs.nott.ac.uk