  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. Our network-adaptive algorithm estimates in
real-time the best parameters of a recently proposed streaming code that uses
forward error correction (FEC) to correct both arbitrary and burst losses,
which cause a crackling noise and undesirable jitters, respectively in audio.
In particular, the destination estimates appropriate coding parameters based on
its observed packet loss pattern and sends them back to the source for updating
the underlying code. Besides, 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 statistical losses
and real-world packet loss traces reveal the following: (i) Our proposed
network-adaptive algorithm combined with our optimal streaming codes can
achieve significantly higher performance compared to uncoded and non-adaptive
FEC schemes over UDP (User Datagram Protocol); (ii) Our explicit streaming
codes can significantly outperform traditional MDS (maximum-distance separable)
streaming schemes when they are used along with our network-adaptive algorithm.
