ISDN User Part Traffic Optimization in the SS7 Network

Abstract

The state of the art Long Term Evolution (LTE) mobile core networks predominantly use Session Initiation Protocol (SIP) and Real-time Transport Protocol (RTP) for transporting voice across networks. Voice-over-LTE (VoLTE) designed for LTE network based on an IP Multimedia System (IMS)) framework is the key to enriched voice communication services for the next generation networks. Emerging technologies like Rich Communication Suite, Web Real-Time Communication (WebRTC) will impart a new dimension to voice communications and harmonise voice and value added services. Many of the mobile networks and voice carriers evolved their network from traditional Time-Division Multiplexing (TDM) / Asynchronous Transfer Mode (ATM) to Internet Protocol (IP). Several interworking protocols have appeared for interexchange of Voice traffic between TDM and IP. SIP to ISDN User Part (ISUP) interoperability mechanisms have been devised. But, in spite of all the endeavours of the mobile networks move to the next generation, it will probably take some time for the networks to completely migrate from UMTS to LTE. As both the technologies will coexist, Circuit Switched Fallback (CSFB) and Single Radio Voice Call Continuity (SRVCC) are two of the most preferred options in the interim to switch from 4G to 3G when the coverage is lost. 3G networks rely still on circuit switching. It can be inferred that circuit switched networks piggybacking on ISUP for voice communication will continue. In this paper, we examine the ISUP signalling aspects to optimize the load sharing and streamline the performance of the traditional voice core networks. We will study the dependencies of the Circuit Identification Code (CIC) selection process on ISUP load sharing to derive our conclusions.