The quality of real-time voice communication over besteffort networks is mainly determined by the de1a.y and loss characteristics observed along the network path. Excessive playout buffering at the receiver is prohibitive and significantly delayed packets have to be discarded and considered as late loss. We propose to improve the tradeoff among delay, late loss rate, and speech quality using multi-stream transmission of real-time voice over the Internet, where multiple redundant descriptions of the voice stream are sent over independent network paths. Scheduling the playout of the received voice packets is based on a novel multi-stream adaptive playout scheduling technique that uses a Lagrangian cost function to trade delay versus loss. Experiments over the Internet suggest largely uncorrelated packet erasure and delay jitter characteristics for different network paths which leads to a noticeable path diversity gain. We observe significant reductions in mean end-to-end latency and loss rates as well as improved speech quality when compared to FEC protected single-path transmission at the same data rate. In addition to our Internet measurements, we analyze the performance of the proposed multi-path voice communication scheme using the ns network simulator for different network topologies, including shared network links. Packet path diversity, multi-stream transmission, multi-path transmission, adaptive playout scheduling, multiple description coding, forward error correction, voice over IP.
Finally, authors in REF have shown substantial improvement for real-time voice communication over the Internet using path diversity together with MDC, and sophisticated playback schedule.
Real-time voice communication over the internet using packet path diversity
MULTIMEDIA '01
Computer Science