Voice over IP has a problem with the high packet loss present on the Internet, reducing speech quality. For interactive voice applications forward error correction has been suggested as a solution not demanding improved quality of service. The goal was to produce a speech decoder that use GSM enhanced full rate encoded data as primary source and as redundant source, a LPC-10 like vocoder. This decoder was designed and implemented and then the quality was measured with SNR, perceptual speech quality measure and comparative mean opinion score listening tests. The speech quality was primarily improved for single packet losses but also for longer bursts. Distortions which there were no time to correct prevents showing the full potential of the concept. The speech quality improvement potential of this scheme is significant and worth exploiting despite the increased delay.