Optimization of source and channel coding for voice over IP

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. Subjective listening tests show that our scheme improves the perceptual speech quality significantly compared to the non-adaptive baseline speech transmission system.

• Type: Conference
• Year: 2005
• Where: ICMCS
• Authors: Yicheng Huang, Jari Korhonen, Ye Wang
• Keywords: ICMCS 2005, Packet Loss Rate, Perceptual Speech Quality, Speech Quality