Thumbnail Image

Adaptive FEC-Based Error Control for Interactive Audio in the Internet

Excessive packet loss rates can dramatically decrease the audio quality perceived by users of Internet telephony applications. Recent results suggest that error control schemes using forward error correction (FEC) are good candidates for decreasing the impact of packet loss on audio quality. With FEC schemes, redundant information is transmitted along with the original information so that the lost original data can be recovered at least in part from the redundant information. Clearly, sending additional redundancy increases the probability of recovering lost packets, but it also increases the bandwidth requirements and thus the loss rate of the audio stream. This means that the FEC scheme must be coupled to a rate control scheme. Furthermore, the amount of redundant information used at any given point in time should also depend on the characteristics of the loss process at that time (it would make no sense to send much redundant information when the channel is loss free), on the end to end delay constraints (destination typically have to wait longer to decode the FEC as more FEC information is used), on the quality of the redundant information, etc. However, it is not clear how to choose the ”best” possible redundant information given all the constraints mentioned above. We address this issue in the paper, and illustrate our approach using a FEC scheme for packet audio recently standardized in the IETF. We find that the problem best redundant information can be expressed mathematically as a constrained optimization problem for which we give explicit solutions. We obtain from these solutions a simple algorithm with very interesting features: i) it optimizes a subjective measure of quality (such as the perceived audio quality at a destination) as opposed to a non subjective measure (such as the packet loss rate at a destination), ii) it incorporates the constraints of rate control and playout delay adjustment schemes, and iii) it adapts to varying (and estimated on line with RTCP) loss conditions in the network. We have been using the algorithm, together with a TCP-friendly rate control scheme, for a few months now and we have found it to provide very good audio quality even with high and varying loss rates. We present simulation and experimental results to illustrate its performance.
Research Projects
Organizational Units
Journal Issue
Publisher Version
Embedded videos