Signaling performance of SIP based VoIP: a measurement-based approach

VoIP must offer the same level of performance and reliability as PSTN, if it were to replace the traditional PSTN. An important indicator of the performance of VoIP is the signaling or call set up delay. Most of the prior research efforts concerning the signaling performance consider long distance calls routed over public wide area networks. Due to the many outstanding issues associated with using VoIP over public networks, however, VoIP is considered to be a viable option primarily in the case of corporate intranets and tunnels. When VoIP is used over corporate intranets and tunnels, the influence of factors such as workload of end user devices, and the number of proxies along the routing path may be prominent and this influence needs to be determined. Also, corporate VoIP service is likely to be offered across heterogeneous network infrastructures using solutions from multiple vendors. As a result, it is necessary to be able to measure the signaling performance of VoIP independent of the underlying networking technology and the vendor-specific implementation. In this paper we demonstrate the feasibility of open, standard JAIN (Java APIs for integrated networks) SIP APIs, which are originally developed to foster rapid service creation across heterogeneous networks, for the purpose of measuring signaling performance of VoIP. Since the performance measurement capability is based on open, standard APIs it can be used uniformly for VoIP implementations from multiple vendors operating on heterogeneous network infrastructures. We quantify the impact of workload on end-user devices on the signaling performance of VoIP via extensive experimentation. Our results can be used to estimate the signaling performance for a given level of load prior to deployment of VoIP.