Using Linksys SPA-3102 interconnection to analog telephony is also tested and worked for one client. After codec testing is arranged then an interconnection system of PSTN or analog telephony system is also tested. G.729 and G.723.1 is limited for one user only so it can be tested only for one user. 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. This research design an open source system of Asterisk server because company need of VoIP that can support traditional analog telephony system. VoIP also use codec that can compress voice data but the quality is still good. Telephony technology is also developed very fast and there is some alternative to use VoIP beside analog telephone because the cost is cheaper. Nowadays information technology, especially the Internet developed very rapidly, which is actually a Internet computers connected to each other. It is recommended that the VoIP server should not be loaded continuously at this maximum processor load for optimum performance and longevity of life. At the maximum concurrent calls handling capacity, the processor of the VoIP server was observed to have 100% loading or utilization measure. While that of UDP bandwidth measurement gives 50Mbps, jitter value of 1.794ms and packet loss of 0.59% for IPv4 traffic and 49.7Mbps bandwidth, 0.592ms jitter with packet loss of 0.22% for IPv6 traffic.
The result from experimental values shows that the achievable bandwidth otherwise called TCP throughput (for the 100Mbps LAN) for IPv4 and IPv6 traffics in dual stack configuration is 86.8Mbps and 87.4Mbps respectively.
It can be deduced that the concurrent calls capacity of a VoIP server is dependent on a number of factors such as the VoIP readiness / bandwidth of the network and hardware specifications of the VoIP server. Voipmonitor, another open source software, was used to measure the bandwidth capacity and maximum concurrent calls handling capacity of the Asterisk Server over fast Ethernet Local Area Network. Network readiness in IPv4 and IPv6 environments was evaluated using Iperf while VoIP calls were simulated with open source SIPp (Session Initiation Protocol performance-tester). This paper presents an empirical study of achievable bandwidth/concurrent calls capacity of an open source Voice over Internet Protocol (VoIP) Server deployed on an Intranet with open source tools.
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