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2005
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Optimization of source and channel coding for voice over IP

10 years 10 months ago
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.
Yicheng Huang, Jari Korhonen, Ye Wang
Added 24 Jun 2010
Updated 24 Jun 2010
Type Conference
Year 2005
Where ICMCS
Authors Yicheng Huang, Jari Korhonen, Ye Wang
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