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Author:Martin-Perez, Javier
Title:Providing Carrier Grade voice Services with Session Initiation Protocol
Publication type:Master's thesis
Publication year:2014
Pages:ix + 67      Language:   eng
Department/School:Sähkötekniikan korkeakoulu
Main subject:Networking Technology   (S3029)
Supervisor:Kantola, Raimo
Instructor:Luoma, Marko
Electronic version URL: http://urn.fi/URN:NBN:fi:aalto-201405131789
OEVS:
Electronic archive copy is available via Aalto Thesis Database.
Instructions

Reading digital theses in the closed network of the Aalto University Harald Herlin Learning Centre

In the closed network of Learning Centre you can read digital and digitized theses not available in the open network.

The Learning Centre contact details and opening hours: https://learningcentre.aalto.fi/en/harald-herlin-learning-centre/

You can read theses on the Learning Centre customer computers, which are available on all floors.

Logging on to the customer computers

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  • Other customers log on using a shared username and password.

Opening a thesis

  • On the desktop of the customer computers, you will find an icon titled:

    Aalto Thesis Database

  • Click on the icon to search for and open the thesis you are looking for from Aaltodoc database. You can find the thesis file by clicking the link on the OEV or OEVS field.

Reading the thesis

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  • You cannot save the thesis file on a flash drive or email it.
  • You cannot copy text or images from the file.
  • You cannot edit the file.

Printing the thesis

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  • Other customers can use the printer u90203-psc3. All printing is subject to a charge to non-University members.
Location:P1 Ark Aalto  1050   | Archive
Keywords:Voice over Internet Protocol
Quality of Service
signalling
Session Initiation Protocol
routing
availability
Session Border Controller
Abstract (eng):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.
ED:2014-05-18
INSSI record number: 49024
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