The Research Of AMR Based VoIP System

Traditionally, fixed bit-rate speech coding is used in VoIP system. This technology can not adaptively switch its bit-rate due to network condition. So we select AMR speech coder.The AMR speech coder consists of the multi-rate speech coder, a source controlled rate scheme including a voice activity detector and a comfort noise generation system, and an error concealment mechanism to combat the effects of transmission errors and lost packets. The multi-rate speech coder is a single integrated speech codec with eight source rates from 4.75 kbit/s to 12.2 kbit/s, and a low rate background noise encoding mode. The speech coder is capable of switching its bit-rate every 20 ms speech frame upon command. Compared with fixed bit-rate speech coder AMR can provide more flexibility.This paper focuses on how to efficiently merge VoIP system and Adaptive Multi-Rate (AMR) technology. The general knowledge of VoIP system and AMR were introduced in the first chapter. Based on the analyses of these two technologies, we proposed a method to merge them into one system in order to overcome the drawback of fixed bit-rate voice coding in VoIP system. By simulation we proved that VoIP system based on AMR speech coder can achieve higher performance gain than that based on fixed bit-rate coder.Besides, we do optimization on comfort noise generation system, voice active detector and error concealment mechanism to enhance the performance of the system. Simulation shows that the optimized AMR algorithm in VoIP system reduces the time latency to about 80% compared to standard AMR based VoIP system. Therefore, our proposed optimal AMR speech coder can work well in VoIP system and the performance gain is huge compared with fixed bit-rate speech coder.