OPTIMIZATION OF SOURCE AND CHANNEL CODING FOR VOICE OVER IP (FriAmPO1)

Author(s) : Yicheng Huang (National University of Singapore, Singapore) Jari Korhonen (National University of Singapore, Singapore) Ye Wang (National University of Singapore, Singapore)

Abstract : Voice over Internet Protocol (VoIP) applications must typically choose a tradeoff between the bits allocated for Forward Error Correcting (FEC) and that for the source coding to achieve the best speech quality at a given packet loss rate. In this paper, we present a new scheme to optimize the speech quality subject to the bandwidth constraints and the packet loss rate. The scheme adopts Adaptive Multi-Rate (AMR) speech codec along with a FEC scheme based on Exclusive OR (XOR) operations. Retransmission is also taken into account if the Round Trip Time (RTT) is within a certain limit. We use a simplified E-Model as objective metric. Extensive subjective listening tests show that our scheme improves the perceptual speech quality significantly compared to the non-adaptive baseline speech transmission system.

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