Low-Latency Network-Adaptive Error Control for Interactive Streaming

We introduce a novel network-adaptive algorithm that is suitable for alleviating network packet losses for low-latency interactive communications between a source and a destination. Network packet losses happen in a bursty manner as well as an arbitrary manner, where the former is usually due to network congestion and the latter can be caused by unreliable wireless links. Our network-adaptive algorithm estimates in real time the best parameters of a recently proposed streaming code that corrects both arbitrary losses (which cause crackling noise in audio) and burst losses (which cause undesirable jitters and pauses in audio) using forward error correction (FEC). The network-adaptive algorithm updates the coding parameters in real time as follows: The destination estimates appropriate coding parameters based on its observed packet loss pattern and then the parameters are fed back to the source for updating the underlying code. In addition, a new explicit construction of practical low-latency streaming codes that achieve the optimal tradeoff between the capability of correcting arbitrary losses and the capability of correcting burst losses is provided. Simulation evaluations based on real-world packet loss traces reveal that our proposed network-adaptive algorithm combined with our optimal streaming codes achieves significantly higher reliability compared to uncoded and non-adaptive FEC schemes over UDP (User Datagram Protocol).