Providing Carrier Grade voice Services with Session Initiation Protocol

摘要

SIP is defined as a protocol that enables end-to-end voice calls as well as for establishing multiparty, multimedia communications in IP-based networks. Presently, SIP is the most widely deployed intra-carrier VoIP protocol but it is also extensively utilized within many carrier networks for transporting voice/video calls over short and long distances. For all of these reasons, SIP can lay a claim to being the global standard for software based voice communication over IP.Furthermore, an important driving force for IP telephony is cost savings for consumers and corporations with large data networks. The high cost of long-distance and international voice calls presents both a challenge and an opportunity and must be taken into account. A significant portion of this cost originates from regulatory taxes imposed on long distance voice calls within the legacy networks. Such surcharges do not apply to long-distance circuit networks carrying data traffic; thus, for a given bandwidth, making a data call is much less expensive than a voice call. The objective of this research is to acknowledge SIP based communications as way to provide a better, reliable, cost effective, resource efficient and service flexible method for communications. The results will show certain vulnerabilities or weaknesses of the method, but also point solutions.This thesis explains the VoIP/SIP based telephony network with call routing and admission control for real time traffic flows, also considering the priority usage perspective. To accomplish the main objectives, of proving the advantages of VoIP over traditional voice communications, we will analyze concepts such as Assured Services SIP, Multi-Level Precedence, admission control and bandwidth broker network elements. Moreover, we will touch Signaling System 7 with Session Border Controller as well as a small comparison to H.323 protocol.