Selecting an audio redundancy codec combination for error control in Internet telephony

In the current Internet, the quality of service of audio application is not guaranteed due to variable loss and delay. Forward error correction (FEC)-based error control mechanisms which send the redundant information with the main information can be used to minimize the effects of packet loss. However, increasing the amount of redundancy may waste bandwidth and increase the end-to-end delay. The existing FEC-based error control mechanisms select the codec combination only based on the network loss rate. These mechanisms do not consider the end-to-end delay by increasing the amount of redundancy. This may result in exceeding the acceptable maximum delay. We propose a selection mechanism of the audio codec combination for error control. In simulations, we confirm that the proposed algorithm reduces the end-to-end delay and the packet loss together.