Video communication is often afflicted by various forms of losses, such as packet loss over the Internet. This paper examines the question of whether the packet loss pattern, and in particular the burst length, is important for accurately estimating the expected mean-squared error distortion. Specifically, we (1) verify that the loss pattern does have a significant effect on the resulting distortion, (2) explain why a loss pattern, for example a burst loss, generally produces a larger distortion than an equal number of isolated losses, and (3) propose a model that accurately estimates the expected distortion by explicitly accounting for the loss pattern, inter-frame error propagation, and the correlation between error frames. The accuracy of the proposed model is validated with JVT/H.26L coded video and previous frame concealment, where for most sequences the total distortion is predicted to within ±0.25 dB for burst loss of length two packets, as compared to prior models which underestimate the distortion by about 1.5 dB. Furthermore, as the burst length increases, our prediction is within ±0.7 dB, while prior models degrade and underestimate the distortion by over 3 dB.

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The quality of real-time voice communication over besteffort networks is mainly determined by the delay 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. KeywordsPacket path diversity, multi-stream transmission, multi-path transmission, adaptive playout scheduling, multiple description coding, forward error correction, voice over IP.

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