Empirical Study of Achievable Bandwidth Capacity of VoIP Infrastructure in an Intranet with Open Source Tools

Abstract

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. 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). 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. 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. 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. 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. At the maximum concurrent calls handling capacity, the processor of the VoIP server was observed to have 100% loading or utilization measure. It is recommended that the VoIP server should not be loaded continuously at this maximum processor load for optimum performance and longevity of life.