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Planning Internet-based applications services such as telephony, requires a variety of components to ensure the quality of services, for instances, user identification, authentication, authorization, registration, accounting, routing, gateway discovery, directory service, signaling, etc. Two protocols have emerged to provide these functions, H.323 series of recommendations by ITU-T, and session initiation protocol (SIP) by multi-party multimedia session control working group, IETF. SIP is a call processing protocol. It is used to create sessions carry session descriptions which allow participants to agree on a set of compatible media types. SIP supports user mobility by proxying and redirecting requests to the user's current location. Users can register their current location as well. The session may be multimedia conference, or point-point telephone call. SIP is not dependent upon any particular conference control protocol, such as H323, and it does not define any method of transporting the session traffic. All the major SIP traffics, management and services will be handled by SIP proxy, redirect and registrar servers. The servers play very important roles in the SIP network. The resources for servers might become a bottleneck for the SIP network, although the SIP is lightweight for being a signaling protocol and companies and organizations have just developed some SIP related services, it still has many virgin lands to be dig out. But before an SIP network service is planned and implemented, one major concern the SIP service providers care the most, is how good (or bad) the service performance is going to be under various circumstance. In other words, they are trying to get more understanding about the network performance impacts as a whole, when certain type of traffic surge due to a particular event for instance, or when a portion of route gets blocked due to a network problem, etc. This research tries to establish a network planning model that simulates normal- operations of an SIP network. Based on this model, a variety of multimedia traffic is supported. In addition, the research implements a convenient Web-based tool to analyze the performance of respective network elements in a simulated SIP environment and under different configurations and scenario setups. The tool supports VoIP (voice over Internet protocol) traffic types initially. The user can input the desired traffic types, volumes and distributions, and configure the simulation scenario through the friendly input format from any client.