Implementing Intelligent Network Services in VoIP Application with SIP, TRIP and ENUM

Internet telephony must offer the standard telephony services. However, the transition to Internet-based telephony services also provides an opportunity to create new services more rapidly and with lower complexity than in the existing public switched telephone network (PSTN). The session initiation protocol (SIP) is a signaling protocol that creates, modifies and terminates associations between Internet end systems. SIP implements services such as call forwarding and transfer, placing calls on hold, camp-on and call queuing by a small set of call handling primitives. The session initiation protocol (SIP) uses DNS procedures to allow a client to resolve a SIP URI into the IP address, port, and transport protocol of the next hop to contact. Generally, these are the problems of mapping the name of a destination into an address, and to find the best route to the destination in a combined IP and PSTN network. Two protocols are being developed by the Internet Engineering Task Force (IETF) to solve these problems. The telephony routing over IP (TRIP) protocol solve the gateway location problem by distributing routing information between entities on the IP network. The telephony numbering mapping (ENUM) provides a solution to the terminal location problem based on DNS. This paper describes the problem of implementing intelligent network services in VoIP application with SIP (signalling protocol), TRIP (routing protocol over IP) and ENUM