패킷크기와 코덱에 기반한 VoIP 음성품질 개선

Abstract

VoIP that provide voice service using internet as a new opportunity and is getting into the spotlight business We transmit compressed voice data by various kinds codec such as PCM and ADPCM on internet in this treatise, offer tone quality that is improved about limit latency that analyze effect that change of packet size and limit latency gets in tone quality of receiving area and is given about suitable CODEC selection plan study . An experiment to success was achieved between multiplex points through Public internet. Transmission delay and packet damage etc. that The transmiter encodes voice signal by PCM and ADPCM from microphone and transmited this using UDP packet, and happens in net in ethics side rough packet by voice signal play . Used MOS estimation method for tone quality estimation, measurement method measures random after Call Forwarding set to Seoul call center and VQT with PAMS (Perceptual Analysis Measurement System) Algorithm, could know that voice quality becomes different by voice codec Data Rate and Datagram size as result that use and measure Audio Direction, Reference Audio.