Providing Carrier Grade voice Services with Session Initiation Protocol

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dc.contributor Aalto-yliopisto fi
dc.contributor Aalto University en
dc.contributor.advisor Luoma, Marko
dc.contributor.author Martin-Perez, Javier
dc.date.accessioned 2014-05-13T08:43:20Z
dc.date.available 2014-05-13T08:43:20Z
dc.date.issued 2014-05-05
dc.identifier.uri https://aaltodoc.aalto.fi/handle/123456789/13034
dc.description.abstract 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. en
dc.format.extent 67+9
dc.format.mimetype application/pdf en
dc.language.iso en en
dc.title Providing Carrier Grade voice Services with Session Initiation Protocol en
dc.type G2 Pro gradu, diplomityö en
dc.contributor.school Sähkötekniikan korkeakoulu fi
dc.subject.keyword Voice over Internet Protocol en
dc.subject.keyword Quality of Service en
dc.subject.keyword signalling en
dc.subject.keyword Session Initiation Protocol en
dc.subject.keyword routing en
dc.subject.keyword availability en
dc.subject.keyword Session Border Controller en
dc.identifier.urn URN:NBN:fi:aalto-201405131789
dc.programme.major Networking Technology fi
dc.programme.mcode S3029 fi
dc.type.ontasot Diplomityö fi
dc.type.ontasot Master's thesis en
dc.contributor.supervisor Kantola, Raimo
dc.programme TLT - Master’s Programme in Communications Engineering fi
dc.location P1 fi


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