VOICE CODEC QUALITY COMPARISON AND INTERCONNECTION TESTING BETWEEN ASTERISK SERVER AND PSTN CONNECTION

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Nowadays information technology, especially the Internet developed very rapidly, which is actually a Internet computers connected to each other. Telephony technology is also developed very fast and there is some alternative to use VoIP beside analog telephone because the cost is cheaper. VoIP also use codec that can compress voice data but the quality is still good. This research design an open source system of Asterisk server because company need of VoIP that can support traditional analog telephony system. Beside design an open source system, some codec technology is also tested, which are G.711 as commonly codec and also G.729 and G.723.1 as propiteary codecs, offering less bandwidth and more clearly sound than G.711. G.729 and G.723.1 is limited for one user only so it can be tested only for one user. After codec testing is arranged then an interconnection system of PSTN or analog telephony system is also tested. Using Linksys SPA-3102 interconnection to analog telephony is also tested and worked for one client.