Dissertations / Theses: 'Internet telephony'

1

Perri, Richard. "Internet Telephony." Thesis, Monterey, Calif. : Springfield, Va. : Naval Postgraduate School ; Available from National Technical Information Service, 1999. http://handle.dtic.mil/100.2/ADA374104.

Full text

APA, Harvard, Vancouver, ISO, and other styles

2

Forsell, Erik. "Internet Telephony : An Internet Service Provider's Perspective." Thesis, KTH, Mikroelektronik och Informationsteknik, IMIT, 2005. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-92287.

Full text

Abstract:

The aim of this Masters Thesis is to propose to SYSteam Nät AB, a local Internet Service Provider (ISP) in Uppsala, Sweden, how to implement IP telephony in their existing ITinfrastructure as a service to their customers. Thus the perspective of the thesis will be that of a local Internet Service Provider. Three general areas are covered in the thesis: Market and Business Model, Technology, and Economics. Important issues for SYSteam Nät AB as an established local broadband Internet Service Provider are to both retain present customers and to attract new customers. Some believe that offering value added services such as IP telephony could do this. Implementation of IP telephony can be done in different ways to fulfil SYSteam Nät’s requirements. The analysis leads to a proposal of how SYSteam Nät could implement IP telephony. This involves many multi-faceted business, technical, and financial issues; each aspect is examined in this thesis.
Avsikten med detta examensarbete är att komma med ett förslag till SYSteam Nät AB, en lokal Internetleverantör i Uppsala, om hur de, som en service till sina kunder, kan implementera IP telefoni i sin existerande IT-infrastruktur. Detta betyder att jag behandlar frågeställningen med en Internetleverantörs perspektiv.Tre huvudområden behandlas i examensarbetet: Marknad och Affärsmodell, Teknik och Ekonomi. Som en etablerad lokal leverantör av Internet via bredband är det viktigt för SYSteam Nät AB att både behålla nuvarande kunder och att attrahera nya kunder. En del tror att man skulle kunna åstadkomma detta genom att erbjuda värdehöjande tjänster som IP telefoni. Implementering av IP telefoni, som svarar mot SYSteam Näts krav, kan utföras på olika sätt. Analysen, som leder till ett förslag hur SYSteam Nät skulle kunna implementera IP telefoni, involverar många mångfasetterade frågeställningar av affärs-, teknisk- och ekonomisk natur. Var och en av dessa aspekter behandlas i rapporten.

APA, Harvard, Vancouver, ISO, and other styles

3

Hoene, Christian. "Internet telephony over wireless links." [S.l.] : [s.n.], 2006. http://deposit.ddb.de/cgi-bin/dokserv?idn=979194121.

Full text

APA, Harvard, Vancouver, ISO, and other styles

4

Andersson, Magus. "Distribution Channels for Internet Telephony." Thesis, University of Skövde, Department of Computer Science, 1998. http://urn.kb.se/resolve?urn=urn:nbn:se:his:diva-205.

Full text

Abstract:

There are a great number of models and theories described by authors like Kotler & Armstrong (1996) and Hutt & Speh (1989) about distribution channels. However the models and theories are quite abstract and not branch specific.

The size of the Internet is growing in an incredible speed and new possibilities with the Internet are being invented every day. The new technologies make it possible to use the Internet in many different businesses (Statskontoret, 1997). One such business is the telecommunication business and the technique of Internet telephony.

With these both statements above I ask myself: How can Internet telephony be distributed?

The question is if existing distribution channels on the market are suitable for the Internet telephony products, and what channels must telecommunication companies use to be competitive on the new combined tele- and datacommunication market.

The purpose of this research is to contribute to an increasing knowledge in how Internet telephony can be distributed. Further the purpose is to identify and evaluate different options of distribution and find important factors that affect the choice and management of the distribution channel.

Because of a lack of information in literature describing distribution channels for Internet telephony products, I decided to do a qualitative study in order to get a hold of relevant information. Since I wanted to study a such new and specific area as distribution channels for Internet telephony products and as I wanted to have the possibility to come back with supplementary questions, I chose a case study with complementary interviews as my method.

This research has shown that there are a number of options to distribute Internet telephony products. There are also a number of intermediaries which are relatively new on the market and are not described in detail in any literature. New kinds of value- added resellers and system integrators with good competence are quite common on the tele- and datacommunication market today. They do not only resell products but also add value in form of hardware, software or service. When doing a choice of distribution channel both manufacturers and channel members want to have a close long term relationship. This can be done with common objectives and strategies or with partnerships in different forms. In evaluating a choice of distribution channel, criterias like cost, control and adaptivity, should be taken in concern. A multichannel or horizontal distribution approach is also quite common on the tele- and datacommunication market. Manufacturers need to use different channels to reach different customer segments. To manage a good relationship in a channel the manufacturer have to motivate, handle conflicts and support the channel members.

APA, Harvard, Vancouver, ISO, and other styles

5

Marsh, Ian. "Quality aspects of Internet telephony." Doctoral thesis, Stockholm : Skolan för elektro- och systemteknik, Kungliga Tekniska högskolan, 2009. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-10572.

Full text

APA, Harvard, Vancouver, ISO, and other styles

6

Sears, Andrew Lester 1973. "Directory services for Internet telephony : creating a spanning layer over the Internet and telephone networks." Thesis, Massachusetts Institute of Technology, 1997. http://hdl.handle.net/1721.1/43529.

Full text

Abstract:

Thesis (M.S.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1997.
Includes bibliographical references (leaves 57-58).
by Andrew Lester Sears.
M.S.

APA, Harvard, Vancouver, ISO, and other styles

7

Chow, Wing Yan. "Packet loss recovery in internet telephony." HKBU Institutional Repository, 2007. http://repository.hkbu.edu.hk/etd_ra/831.

Full text

APA, Harvard, Vancouver, ISO, and other styles

8

Tse, Lily. "Feasibility study of VoIP integration into the MYSEA environment." Thesis, Monterey, Calif. : Springfield, Va. : Naval Postgraduate School ; Available from National Technical Information Service, 2005. http://library.nps.navy.mil/uhtbin/hyperion/05Sep%5FTse.pdf.

Full text

Abstract:

Thesis (M.S. in Computer Science)--Naval Postgraduate School, September 2005.
Thesis Advisor(s): Cynthia E. Irvine, Thuy D. Nguyen. Includes bibliographical references (p. 179-180). Also available online.

APA, Harvard, Vancouver, ISO, and other styles

9

Evloguieva, Evelina. "Light-weight SIP protocol for Internet telephony services." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 2000. http://www.collectionscanada.ca/obj/s4/f2/dsk1/tape3/PQDD_0020/MQ55052.pdf.

Full text

APA, Harvard, Vancouver, ISO, and other styles

10

Evloguieva, Evelina. "Light-weight SIP protocol for internet telephony services." Thesis, McGill University, 1999. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=30117.

Full text

Abstract:

The technology that governs the Telecommunications today is based on the Intelligent Networks (IN). But we can see that the Internet Telephony is emerging rapidly and that it has good chances to become the basis for the next generation telecommunication networks. There are two major competing standards for Internet Telephony - H.323 protocol stack and SIP. This thesis focuses on SIP and the Value Added Services in the SIP based Internet Telephony. It describes and analyzes two existing SIP based approaches to the VAS implementation and presents new hybrid SIP-IN approach based on the concept of reusing the existing IN nodes. The major part of the study is devoted to the design of a lightweight protocol, built as SIP extension, providing for VAS in the hybrid SIP-IN environment. To illustrate the hybrid SIP-IN approach to VAS implementation and the SIPext protocol operation the execution of the Freephone and the Call Distribution services is described. Finally the functional modules supporting the SIPext communication in a hybrid SIP-IN architecture are outlined.

APA, Harvard, Vancouver, ISO, and other styles

11

Chin, Chi Kwan. "Design and control of an Internet telephony system." HKBU Institutional Repository, 2000. http://repository.hkbu.edu.hk/etd_ra/239.

Full text

APA, Harvard, Vancouver, ISO, and other styles

12

Eliasson, Erik. "Secure Internet telephony : design, implementation and performace measurements /." Stockholm : Telecommunication Systems Laboratory, Electronic, Computer and Software Systems, KTH, Royal Institute of Technology, 2006. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-4080.

Full text

APA, Harvard, Vancouver, ISO, and other styles

13

Celikadam, Turgut. "Design And Development Of An Internet Telephony Test Device." Master's thesis, METU, 2003. http://etd.lib.metu.edu.tr/upload/1223148/index.pdf.

Full text

Abstract:

The issues involved in Internet telephony (Voice over Internet Protocol (VoIP) device) can be best understood by actually implementing a VoIP device and studying its performance. In this regard, an Internet telephony device, providing full duplex voice communication over internet, and a user interface program have been developed. In the process, a number of implementation issues came into focus, which we have touched upon in this thesis. Transport layer network protocols are discussed in the concept of real time streaming applications and Real Time Protocol (RTP) is modified to use as transport layer protocol in developed VoIP device. Adaptive playout buffering algorithms are studied and compared with each other by trace driven simulation experiments with objective measures. A method to solve clock synchronization problem in streaming internet applications is presented. One way and round trip delay measurement functionalities are added to the VoIP device, so that device can be used to investigate the network characteristics.

APA, Harvard, Vancouver, ISO, and other styles

14

Glitho, Roch H. "A Mobile Agent Based Service Architecture for Internet Telephony." Doctoral thesis, KTH, Microelectronics and Information Technology, IMIT, 2002. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-3320.

Full text

Abstract:

Internet Telephony defined as real time voice or multimediacommunications over packet switched networks dates back to theearly days of the Internet. ARPA's Network SecureCommunications project had implemented, as early as December1973, an infrastructure for local and transnet real time voicecommunication. Two main sets of standards have emerged: H. 323from the ITU-T and the session initiation protocol (SIP) fromthe Internet Engineering Task Force (IETF). Both includespecifications for value added services. Value added services,or more simply services, are critical to service providers'survival and success. Unfortunately, the service architecturesthat come with the ITU-T and the IETF sets of standards arerather weak. Although they are constantly evolving,alternatives and complements need to be researched. This thesiswhich is made up of a formal dissertation and 6 appendices,proposes a novel mobile agent based service architecture forInternet Telephony. The architecture addresses the issues noneof the existing architectures solves in a satisfactory manner.Furthermore it adds mobile agents to the panoply of servicecreation tools. The appendices are reprints of articlespublished in refereed magazines/journals or under considerationfor publication. The formal dissertation is a summary of thepublications. A consistent and comprehensive set ofrequirements are derived. They are TINA-C flavored, but adaptedto Internet Telephony. They are used to critically reviewrelated work and also used to motivate the use of mobile agentsas the pillars of a novel architecture. The components of thisnovel architecture are identified. The key component is themobile service agent. It acts as a folder and carriesservice(s) to which the end-user has subscribed. Mobile serviceagents need to be upgraded when new versions of service logicare available and when end-users make changes to service data.This thesis proposes a novel upgrading framework. The currentInternet infrastructure comprises a wide range of hosts. Mobileagent platforms are now available for most of thesehosts/clients including memory/processing power constrainedPDAs. Our mobile service agents need to adapt to hostvariability when roaming. A novel adaptivity framework is alsoproposed. These two frameworks are general and can be appliedto any other mobile agent which meets a basic set ofassumptions. A key advantage of a mobile agent based servicearchitecture is that it enables the developement of mobileagent based services. The thesis proposes a novel mobile agentbased multi-party session scheduler. The feasibility and theadvantages of the architecture proposed by this thesis havebeen demonstrated by a prototype on which measurements havebeen made. Future work includes the addition of a securityframework to the architecture, and refinenements to theupgrading and adaptivity frameworks. More mobile agent basedservices, especially mobile multi agent based services willalso be developed.

APA, Harvard, Vancouver, ISO, and other styles

15

Beyh, S. "Computer and communication engineering : internet protocol telephony in construction." Thesis, University of Salford, 2004. http://usir.salford.ac.uk/26582/.

Full text

Abstract:

A construction project traditionally involves intensive communication flows between the site operations (workers, gangers, engineers, foremen, etc.), the site office, the company and the Supply Chain. Typically on the jobsite, a temporary site office is set up in order to conduct the operations of the construction project phases. The site office is equipped with traditional telecommunication means such as phone, fax and Internet connection. The site personnel are provided with a multitude of mobile, satellite and wireless telecommunication devices where appropriate, such as PDA, GSM and satellite phones/fax, and walkie-talkies. Technically, these legacy systems, once put together, could be able to provide adequate communication resources to the construction project teams. But one of the main issues emerging from the use of the abovementioned traditional telecommunication systems is that their cost can be found in some cases to be very high. On the other hand, in the absence of providing the necessary communication means available through the traditional telecommunication systems to the personnel on the move for whatsoever reason could be very harmful and, may negatively affect the execution of the construction works and the project lifecycle as a whole. This situation could be overcome if alternative solutions are put in place to reduce cost and improve communications. Therefore, this study has investigated a new communication paradigm known as IP (Internet Protocol) Telephony, which could possibly provide the site office, as well as the entire project team members with adequate, cheaper and more effective communications means at the jobsite. IP Telephony refers to communication services such as voice, video, facsimile, and/or voice-messaging applications that are transported via the Internet, rather than the Public Switch Telephone Network (PSTN). The basic steps involved in originating an IP Telephony call are the conversion of the analogue voice signal into digital format and the compression/translation of the signal into IP packets for transmission over the Internet. This communication paradigm eliminates the need for separate infrastructures for voice and data networks as these services can be implemented over a single data infrastructure. Furthermore, while, from the technical point of view IP Telephony Technology could be ready to satisfy the business case in general, its development within the construction sector has not been observed due to several barriers that have been investigated in this work as being part of the development of an integrated framework that aimed at enabling the use of Internet Protocol Telephony in construction. This research aimed at developing a generic integrated framework for enabling the use of Internet Protocol (IP) Telephony in construction. The process involved in the development of this framework included the conduct of intensive literature around the traditional telecommunication systems used by construction firms in the United Kingdom as well as the investigation of the current situation of IP Telephony technology in terms of availability of commercial services and applications used by the construction industry. The field investigations were obtained through appropriate surveys and interviews conducted with construction firms, telecommunication operators and Internet Protocol (IP) Telephony equipment vendors respectively. The research further looked at the issues related to the transfer of such a technology into the construction industry and investigated the main barriers preventing its implementation in construction sites' environments. These investigations represented an important part in the development of the "Internet Protocol Telephony on Construction Sites (IPTCS) Framework" which represents the focus of this research. The various modes of communications are described under this common framework which is expected to benefit in premier-lieu the construction industry by driving construction firms to look at IP Telephony technology as an adequate and cost effective alternative to their communication means for empowering their mobile personnel on construction sites and in the office alike. It could also motivate telecommunication operators, IP Telephony application developers and equipment vendors to establish specific solutions suitable for construction sites environments according to the industry's needs and requirements.

APA, Harvard, Vancouver, ISO, and other styles

16

Ho, Grant (Grant Ian) 1974. "Internet Telephony : optimizing protocols, packet recovery, and packet size." Thesis, Massachusetts Institute of Technology, 1999. http://hdl.handle.net/1721.1/79968.

Full text

APA, Harvard, Vancouver, ISO, and other styles

17

Culleton, Albert. "Evaluation of VOIP technologies as a replacement for traditional PSTN based PBX systems." [Denver, Colo.] : Regis University, 2006. http://165.236.235.140/lib/ACulletonPartI2006.pdf.

Full text

APA, Harvard, Vancouver, ISO, and other styles

18

Wallace, David T. Vegter Henry M. "Exploitation of existing Voice over Internet Protocol Technology for Department of the Navy application /." Monterey, Calif. : Springfield, Va. : Naval Postgraduate School ; Available from National Technical Information Service, 2002. http://library.nps.navy.mil/uhtbin/hyperion-image/02sep%5FWallace.pdf.

Full text

Abstract:

Thesis (M.S. in Information Technology Management)--Naval Postgraduate School, September 2002.
Thesis advisor(s):Dan Boger, Rex Buddenberg. Includes bibliographical references (p. 101-102). Also available online.

APA, Harvard, Vancouver, ISO, and other styles

19

Tasyumruk, Lutfullah. "Analysis of voice quality problems of Voice Over Internet Protocol (VoIP)." Thesis, Monterey, Calif. : Springfield, Va. : Naval Postgraduate School ; Available from National Technical Information Service, 2003. http://library.nps.navy.mil/uhtbin/hyperion-image/03sep%5FTasyumruk.pdf.

Full text

APA, Harvard, Vancouver, ISO, and other styles

20

Kaplan, Shaun. "Micro-controller based Internet phone." Thesis, Cape Technikon, 2004. http://hdl.handle.net/20.500.11838/1096.

Full text

Abstract:

Thesis (MTech (Electrical Engineering))--Cape Technikon, Cape Town, 2004
This work describes research towards the development of a micro-controller based, standalone Internet telephone to be used as an alternative to conventional line telephones. Our definition of 'stand-alone' refers to the unit's capability to perform its function wholly without the need for an attached computer. The unit should be low cost and capable of allowing two users to communicate using the units. Bandwidth usage should be kept low to allow the unit to be used over dial up connections which are prevalent in South Africa. The units should be easy to use as the anticipated users may be unskilled. A module containing a 16-bit micro-controller, an Ethernet controller, flash memory and RAM was chosen as the controller. The module came with a real-time operating system and a TCPlIP stack. The session initiation protocol (SIP) was selected to perform the signalling. SIP uses the session description protocol (SDP) to negotiate the attributes of the media session to be established. The real-time transport protocol (RTP) was implemented to transport encoded audio between the end points. The RTP control protocol (RTCP) was implemented to provide basic quality of service parameters. The ITU-T recommendation G.729 annex A was the voice codec selected. Codec ICs were used to encode and decode the audio. The implementations were designed specifically for a two user, direct communication environment. That is two phone units were developed that communicated directly with each other and not through intermediary servers.

APA, Harvard, Vancouver, ISO, and other styles

21

Leida, Brett A. (Brett Alan) 1968. "A cost model of Internet service providers : implications for Internet telephony and yield management." Thesis, Massachusetts Institute of Technology, 1998. http://hdl.handle.net/1721.1/10052.

Full text

APA, Harvard, Vancouver, ISO, and other styles

22

Cecere, Grazia. "Economics of internet and telecommunications : an analysis of development and diffusion of internet telephony." Paris 11, 2009. http://www.theses.fr/2009PA111010.

Full text

APA, Harvard, Vancouver, ISO, and other styles

23

Floros, Georgios. "Analysis of Internet Telephony and the H.323 multimedia standard." Thesis, Monterey, Calif. : Springfield, Va. : Naval Postgraduate School ; Available from National Technical Information Service, 1999. http://handle.dtic.mil/100.2/ADA370794.

Full text

Abstract:

Thesis (M.S. in Computer Science) Naval Postgraduate School, September1999.
"September 1999". Thesis advisor(s): Gilbert M. Lundy. Includes bibliographical references (p. 107-108). Also available online.

APA, Harvard, Vancouver, ISO, and other styles

24

Evripidis, Romanidis. "Lawful Interception and Countermeasures : In the era of Internet Telephony." Thesis, KTH, Kommunikationssystem, CoS, 2008. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-91683.

Full text

Abstract:

Lawful interception and the way it is performed have played a significant role in the effectiveness of this type of communication monitoring. Although the secrecy of interception and the related equipment are supposed to provide correct information to a law enforcement agency, there are some countermeasures that can be taken by the subject that can seriously undermine the collection of correct and accurate data. This thesis project attempts to identify the problems that exist for interception of telephony (be it fixed, mobile, or via the Internet). Moreover, there are some suggestions for improvements how lawful interception should be performed in order to avoid possible attacks that could decrease the credibility of the intercepted data. Numerous publications (in print or distributed on the Internet) have described weaknesses in the current state of the art lawful interception when using equipment that can be purchased in the market. This thesis presents improvements in how LI can be conducted in order to avoid these vulnerabilities. Additionally, there is a description of the key escrow systems and the possibility of avoiding one of their most significant vulnerabilities. The main problem of the lawful interception is the rapid changes in telecommunications and the complicated architecture of the telecommunication networks, as both make monitoring vulnerable to specific countermeasures. An analysis of how lawful interception can take place and current countermeasures for lawful interception of Internet telephony are vital in order to identify the problems in carrying out such intercepts today and to make suggestions for improvements. This topic is especially relevant given the current Swedish “FRA lagen” regarding interception of electronic communication going into, out of, and through Sweden. Not only is it important to understand how lawful interception can be performed or prevented, but it is also important to understand how information obtained from lawful interception could be purposely misleading or falsified.

APA, Harvard, Vancouver, ISO, and other styles

25

Chan, Ken Yat-Wan. "Methods for designing Internet telephony services with fewer feature interactions." Thesis, University of Ottawa (Canada), 2003. http://hdl.handle.net/10393/26457.

Full text

Abstract:

This thesis describes a practical formal approach to designing Internet (IP) Telephony services with fewer feature interactions using SDL and Message Sequence Charts (MSC). Feature interactions are undesirable side effects caused by interactions between features and/or their environment. These undesirable side effects are known to affect the construction of reliable software systems. The IETF 'Session Initiation Protocol' (SIP) is chosen as the IP telephony signaling protocol for the case study of this thesis. Although IP Telephony services do not have some of the traditional feature interactions, the service designers are confronted with new feature interaction problems. We have proposed an extension to the classical feature interaction classification system. The main contribution of this thesis is our formal SDL model of SIP and its sample services that are derived from informal SIP specification. We apply use case analysis to some informal extent in our design, and specify SIP services more precisely by creating an "Abstract User interface". (Abstract shortened by UMI.)

APA, Harvard, Vancouver, ISO, and other styles

26

Jiang, Dongmei. "Internet telephony services for presence with SIP and extended CPL." Thesis, University of Ottawa (Canada), 2004. http://hdl.handle.net/10393/26668.

Full text

Abstract:

Internet Telephony is the next generation of telephony with many new features and low cost. Because of the explosion of new features, it has become critical to control and manage these features. The main challenge in Internet Telephony is service programming. The Call Processing Language (CPL) is a solution for end users to describe and control their services in Internet Telephony. Current CPL focuses on call processing services only. It is not adequate for the definition of many types of new services, such as the combination of telephony services with email, instant messaging, presence etc. This thesis extends CPL to describe new Internet Telephony services including presence services and call processing services related to presence. In the thesis, the presence system is systematically described in a three-layer architecture. End user's presence services and system basic services are clearly separated in the architecture. Presence information, as the basis of presence services, is extended from traditional "online" and "offline" indicators to include broader meaning, such as location, phone line status, role and availability status etc. Through CPL extensions for presence, user's new presence services and new presence related call processing services are illustrated by using various examples. A simulation system is implemented to demonstrate the Internet Telephony services specified in extended CPL. End users can create and modify their own services via the Graphic User Interfaces (GUIs) and access their services at any location through the Internet. The simulation system is verified with various test cases.

APA, Harvard, Vancouver, ISO, and other styles

27

Zuniga, Rodriguez Ricardo Francisco. "A comparative analysis of internet protocol telephony in Latin America /." Thesis, McGill University, 2001. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=32819.

Full text

Abstract:

Recent technological advances (such as digitization) and economic shifts (such as privatization and liberalization) have radically changed the telecommunications landscape.
Internet Telephony is one of the most important developments that has resulted from digitization. It raises several legal-regulatory issues at both domestic and international levels.
This thesis analyzes the existing legal-regulatory framework for Internet Telephony in Latin America, and examines its implications and potential developments in the region.
In order to provide a more solid foundation for this analysis, the first chapter provides an overview of the traditional regulatory framework, and describes the technological and economic underpinnings of Internet Telephony.
The second chapter studies the regulatory frameworks for Internet Telephony that have been adopted by Canada, the United States and the European Union. These frameworks could serve as models and provide further guidance to analyze the approaches taken to regulate Internet Telephony in Latin America.
Finally, the third chapter examines the three existing legal-regulatory approaches to Internet Telephony that have been adopted in Latin America and its potential implications.

APA, Harvard, Vancouver, ISO, and other styles

28

Deng, Xianglin. "Security of VoIP : Analysis, Testing and Mitigation of SIP-based DDoS attacks on VoIP Networks." Thesis, University of Canterbury. Computer Science and Software Engineering, 2008. http://hdl.handle.net/10092/2227.

Full text

Abstract:

Voice over IP (VoIP) is gaining more popularity in today‟s communications. The Session Initiation Protocol (SIP) is becoming one of the dominant VoIP signalling protocol[1, 2], however it is vulnerable to many kinds of attacks. Among these attacks, flood-based denial of service attacks have been identified as the major threat to SIP. Even though a great deal of research has been carried out to mitigate denial of service attacks, only a small proportion has been specific to SIP. This project examines the way denial of service attacks affect the performance of a SIP-based system and two evolutionary solutions to this problem that build on each other are proposed with experimental results to demonstrate the effectiveness of each solution. In stage one, this project proposes the Security-Enhanced SIP System (SESS), which contains a security-enhanced firewall, which evolved from the work of stage one and a security-enhanced SIP proxy server. This approach helps to improve the Quality-of-Service (QoS) of legitimate users during the SIP flooding attack, while maintaining a 100 percent success rate in blocking attack traffic. However, this system only mitigates SIP INVITE and REGISTER floods. In stage two, this project further advances SESS, and proposes an Improved Security-Enhanced SIP System (ISESS). ISESS advances the solution by blocking other SIP request floods, for example CANCEL, OK and BYE flood. JAIN Service Logic Execution Environment (JAIN SLEE) is a java-based application server specifically designed for event-driven applications. JAIN SLEE is used to implement enhancements of the SIP proxy server, as it is becoming a popular choice in implementing communication applications. The experimental results show that during a SIP flood, ISESS cannot only drop all attack packets but also the call setup delay of legitimate users can be improved substantially compared to and unsecured VoIP system.

APA, Harvard, Vancouver, ISO, and other styles

29

Adams, Pieter. "The Strategic Migration of Telephony into an Internet Protocol World : a South African Perspective." Diss., University of Pretoria, 2005. http://hdl.handle.net/2263/24129.

Full text

Abstract:

Internet Protocol Telephony (IPT), also known as Voice over Internet Protocol (VoIP), has evolved from a niche technology to one that is adopted fairly well in developed countries. The aim of this research report was to determine whether IPT will also be a success in the Republic of South Africa, which is one of many developing countries. The technology was analysed and it was found that cost reduction, increased productivity and enhanced applications were the most valuable benefits the technology could offer. Particular interesting impediments of the technology were discovered and it was found that there existed both hard issues like security and quality problems, as well as softer issues like internal politics, that could hinder the global success of the technology. The adoption rate of South Africa was compared to that of industrial countries and it was found that South African organisations overall posed a wait-and-see attitude towards IPT. Various implementation models were discussed and it was found that a hybrid approach would be the most viable option for local organisations. The South African environment were analysed and it was discovered that the biggest obstacle for success in South Africa was the regulatory environment. But it was also found that the environment would soon change and that competitors, including Black Economic Empowerment companies, should use the opportunities available. Social factors like HIV/AIDS and theft as well as economic factors like the exchange rate could hamper the competitiveness of local companies using IPT. IPT technology can only be a success in South Africa if it is intensely supported by Government, implemented in the correct manner and adopted aggressively by the local market.
Dissertation (MBA)--University of Pretoria, 2006.
Graduate School of Management
unrestricted

APA, Harvard, Vancouver, ISO, and other styles

30

Lo, Chor Wing. "Voice over IP performance using expedited forwarding and assured forwarding on differentiated services enabled network /." View Abstract or Full-Text, 2002. http://library.ust.hk/cgi/db/thesis.pl?COMP%202002%20LO.

Full text

Abstract:

Thesis (M. Phil.)--Hong Kong University of Science and Technology, 2002.
Includes bibliographical references (leaves 57-59). Also available in electronic version. Access restricted to campus users.

APA, Harvard, Vancouver, ISO, and other styles

31

Lewis, Rosemary. "Operational benefit of implementing VoIP in a tactical environment." Thesis, Monterey, Calif. : Springfield, Va. : Naval Postgraduate School ; Available from National Technical Information Service, 2003. http://library.nps.navy.mil/uhtbin/hyperion-image/03Jun%5FLewis.pdf.

Full text

Abstract:

Thesis (M.S. in Information Systems and Operations)--Naval Postgraduate School, June 2003.
Thesis advisor(s): Dan C. Boger, Rex Buddenberg. Includes bibliographical references (p. 61-62). Also available online.

APA, Harvard, Vancouver, ISO, and other styles

32

Patton, Mark B. "A case study of Internet Protocol Telephony implementation at United States Coast Guard headquarters." Thesis, Monterey, Calif. : Springfield, Va. : Naval Postgraduate School ; Available from National Technical Information Service, 2005. http://library.nps.navy.mil/uhtbin/hyperion/05Mar%5FPatton.pdf.

Full text

Abstract:

Thesis (M.S. in Information Technology Management)--Naval Postgraduate School, March 2005.
Thesis Advisor(s): Dan C. Boger, R. Scott Coté. Includes bibliographical references (p. 134-138). Also available online.

APA, Harvard, Vancouver, ISO, and other styles

33

Krebs, Eric M. "An audio architecture integrating sound and live voice for virtual environments." Thesis, Monterey, Calif. : Springfield, Va. : Naval Postgraduate School ; Available from National Technical Information Service, 2002. http://library.nps.navy.mil/uhtbin/hyperion-image/02sep%5FKrebs.pdf.

Full text

Abstract:

Thesis (M.S. in Modeling, Virtual Environments and Simulation)--Naval Postgraduate School, September 2002.
Thesis advisor(s): Russell D. Shilling, Rudolph P. Darken. Includes bibliographical references (p. 175). Also available online.

APA, Harvard, Vancouver, ISO, and other styles

34

Tang, Jingrong. "Mobile agent based advanced service architecture for H.323 Internet Protocol Telephony." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 2000. http://www.collectionscanada.ca/obj/s4/f2/dsk1/tape4/PQDD_0016/MQ57740.pdf.

Full text

APA, Harvard, Vancouver, ISO, and other styles

35

Wang, Huan Adele. "Building value-added services using mobile agents in SIP based Internet telephony." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1999. http://www.collectionscanada.ca/obj/s4/f2/dsk1/tape3/PQDD_0029/MQ64476.pdf.

Full text

APA, Harvard, Vancouver, ISO, and other styles

36

Wang, Huan Adele 1967. "Building value-added services using mobile agents in SIP based internet telephony." Thesis, McGill University, 1999. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=30765.

Full text

Abstract:

Internet Telephony is attracting a great deal of attention in modern society. IETF SIP is a standard for Internet telephony, but SIP service architecture has some shortcomings, for instance, it does not support service mobility.
This thesis proposes a novel mobile agent based service architecture for Internet Telephony. The architecture relies on the use of two mobile agents for each subscriber: one agent contains originating services and the other contains terminating services. These two mobile agents act as capsules and carry the services to which the user has subscribed. The architecture brings the advantage associated with mobile agents. This includes network traffic reduction and flexibility in service provisioning. We start the thesis by introducing value-added services. The MA based architecture for IP telephony is then introduced. Conceptual model, implementation model, and service scenarios are all described in this architecture. Special attention is paid to service mobility. SIP is used as concrete example throughout the thesis and "Grasshopper" as basis for the implementation architecture.

APA, Harvard, Vancouver, ISO, and other styles

37

Tang, Jingrong Carleton University Dissertation Engineering Systems and Computer. "Mobile agent based advanced service architecture for H.323 Internet protocol telephony." Ottawa, 2000.

Find full text

APA, Harvard, Vancouver, ISO, and other styles

38

Kolan, Prakash Dantu Ram. "System and methods for detecting unwanted voice calls." [Denton, Tex.] : University of North Texas, 2007. http://digital.library.unt.edu/permalink/meta-dc-5155.

Full text

APA, Harvard, Vancouver, ISO, and other styles

39

Dechjaroen, Chaiporn. "Performance evaluation of Voice over Internet Protocol." Thesis, Monterey, Calif. : Springfield, Va. : Naval Postgraduate School ; Available from National Technical Information Service, 2002. http://library.nps.navy.mil/uhtbin/hyperion-image/02Dec%5FDechjaroen.pdf.

Full text

APA, Harvard, Vancouver, ISO, and other styles

40

Blum, Konrad. "Evaluating the applications of spatial audio in telephony." Thesis, Stellenbosch : University of Stellenbosch, 2010. http://hdl.handle.net/10019.1/4376.

Full text

Abstract:

Thesis (MScEng (Electrical and Electronic Engineering))--University of Stellenbosch, 2010.
ENGLISH ABSTRACT: Telephony has developed substantially over the years, but the fundamental auditory model of mixing all the audio from di erent sources together into a single monaural stream has not changed since the telephone was rst invented. Monaural audio is very di cult to follow in a multiple-source situation such as a conference call. Sound originating from a speci c point in space will travel along a slightly di erent path to each ear. Although we are not consciously aware of it, our brain processes these spatial cues to help us to locate sounds in space. It is this spatial information that allows us to focus our attention and listen to a single speaker in an environment where many di erent sources may be active at the same time; a phenomenon known as the \cocktail party e ect". It is possible to reproduce these spatial cues in a sound recording, using Head-Related Transfer Functions (HRTFs) to allow a listener to experience localised audio, even when sound is reproduced through a headset. In this thesis, spatial audio is implemented in a telephony application as well as in a virtual world. Experiments were conducted which demonstrated that spatial audio increases the intelligibility of speech in a multiple-source environment and aids active speaker identi cation. Resource usage measurements show that these bene ts are, however, not without a cost. In conclusion, spatial audio was shown to be an improvement over the monaural audio model traditionally implemented in telephony.
AFRIKAANSE OPSOMMING: Telefonie het ansienlik ontwikkel oor die jare, maar die basiese ouditiewe model waarin die klank van alle verskillende bronne bymekaar gemeng word na een enkelouditoriese stroom het nie verander sedert die eerste telefoon gebou is nie. Enkelouditoriese klank is baie moeilik om te volg in 'n meervoudigebron situasie, soos byvoorbeeld in 'n konferensie oproep. Klank met oorsprong by 'n sekere punt in die ruimte sal 'n e ens anderse pad na elke oor volg. Selfs is ons nie aktief bewus hiervan nie, verwerk ons brein hierdie ruimtelike aanduidinge om ons te help om klanke in die ruimte te vind. Dit is hierdie ruimtelike inligting wat ons toelaat om ons aandag te vestig en te luister na 'n enkele spreker in 'n omgewing waar baie verskillende bronne terselfdertyd aktief mag wees, 'n verskynsel wat bekend staan as die \skemerkelkiepartytjiee ek". Dit is moontlik om hierdie ruimtelike leidrade na 'n klank te reproduseer met behulp van hoofverwandeoordragfunksies (HRTFs) en om daardeur 'n luisteraar gelokaliseerde klank te laat ervaar, selfs wanneer die klank deur middel van oorfone gespeel word. In hierdie tesis word ruimtelike klank ge mplementeer in 'n telefonieprogram, sowel as in 'n virtuelew^ereld. Eksperimente is uitgevoer wat getoon het dat ruimtelike klank die verstaanbaarheid van spraak in 'n meerderebronomgewing verhoog en help met aktiewe spreker identi kasie. Hulpbrongebruiks metings toon aan dat hierdie voordele egter nie sonder 'n koste kom nie. Ter afsluiting, dit is bewys dat ruimtelike klank 'n verbetering tewees gebring het oor die enkelouditorieseklankmodel wat tradisioneel in telefonie gebruik het.

APA, Harvard, Vancouver, ISO, and other styles

41

Lloyd, Patrick. "An exploration of covert channels within voice over IP /." Online version of thesis, 2010. http://hdl.handle.net/1850/12241.

Full text

APA, Harvard, Vancouver, ISO, and other styles

42

Claassen, Hendre. "Validating the unified communications business case for Fruitways." Thesis, Stellenbosch : Stellenbosch University, 2014. http://hdl.handle.net/10019.1/97276.

Full text

Abstract:

Thesis (MBA)--Stellenbosch University, 2014.
ENGLISH ABSTRACT: The difficulty in building a justifiable business case is preventing Fruitways from adopting a unified communication solution. Subsequently the research set out to determine the business value that Fruitways can expect to derive from implementing a unified communication solution before committing to a full-scale implementation. The results of this research have put Fruitways in a position to make an informed decision on whether or not to adopt unified communication. In order to answer the research question a unified communications pilot project was initiated within the Fruitways group using the introduction of Microsoft Lync (a unified communications application) as the event under study. The research described and analysed a unified communications pilot project in a bounded system, Fruitways (the case). The case study made use of abductive reasoning in order to synthesise a case for the business value that Fruitways can expect to derive from implementing unified communication technology. The literature study showed that unified communications provide value to an organisation on four levels; namely, the personal, workgroup, enterprise and infrastructure level. However, participants in the pilot project indicated that they experienced value from unified communication only on three of the four levels as identified in the literature; these levels were: • Personal level. Participants indicated that they felt more productive with unified communications at their disposal. The rich choice of communication methods empowered participants to choose the most appropriate mode of communication given the communication need and context. • Workgroup level. The main drivers of value for participants on this level was found in the increased speed of group tasks, working more effectively across distance and evidence of more effective work practises forming. • Enterprise level. The study showed improved coordination between departments in Fruitways with the deployment of unified communication technology. Surprisingly there was no indication that Fruitways could expect to save significant infrastructure related costs by consolidating communication onto a single data network. The infrastructure related cost saving value proposition that is most often cited by vendors of unified communication systems was not clear for Fruitways. Although the study showed that unified communication presents value to Fruitways, the literature also pointed toward the importance that this value needs to be aligned with the strategic objectives of the business in order to constitute true business value. The business value that unified communication presents to Fruitways does indeed support its strategic objectives.

APA, Harvard, Vancouver, ISO, and other styles

43

Hitchcock, Jonathan. "Decorating Asterisk : experiments in service creation for a multi-protocol telephony environment using open source tools." Thesis, Rhodes University, 2006. http://hdl.handle.net/10962/d1006539.

Full text

Abstract:

As Voice over IP becomes more prevalent, value-adds to the service will become ubiquitous. Voice over IP (VoIP) is no longer a single service application, but an array of marketable services of increasing depth, which are moving into the non-desktop market. In addition, as the range of devices being generally used increases, it will become necessary for all services, including VoIP services, to be accessible from multiple platforms and through varied interfaces. With the recent introduction and growth of the open source software PBX system named Asterisk, the possibility of achieving these goals has become more concrete. In addition to Asterisk, a number of open source systems are being developed which facilitate the development of systems that interoperate over a wide variety of platforms and through multiple interfaces. This thesis investigates Asterisk in terms of its viability to provide the depth of services that will be required in a VoIP environment, as well as a number of other open source systems in terms of what they can offer such a system. In addition, it investigates whether these services can be made available on different devices. Using various systems built as a proof-of-concept, this thesis shows that Asterisk, in conjunction with various other open source projects, such as the Twisted framework provides a concrete tool which can be used to realise flexible and protocol independent telephony solutions for a small to medium enterprise.

APA, Harvard, Vancouver, ISO, and other styles

44

Zhong, Xin. "Speech coding and transmission for improved automatic recognition in communication networks." Diss., Available online, Georgia Institute of Technology, 2004:, 2003. http://etd.gatech.edu/theses/available/etd-04072004-180252/unrestricted/zhong%5Fxin%5F200312%5Fphd.pdf.

Full text

APA, Harvard, Vancouver, ISO, and other styles

45

Clayton, Bradley. "Securing media streams in an Asterisk-based environment and evaluating the resulting performance cost." Thesis, Rhodes University, 2007. http://eprints.ru.ac.za/851/.

Full text

APA, Harvard, Vancouver, ISO, and other styles

46

Julius, Elroy Peter. "Guaranteed delivery of multimodal semi-synchronous IP-based communication." Thesis, University of the Western Cape, 2005. http://etd.uwc.ac.za/index.php?module=etd&amp.

Full text

Abstract:

This thesis explored how hearing and deaf users are brought together into one communication space where interaction between them is a semi-synchronous form of message exchange. The focus of this thesis was the means by which message delivery between two e

APA, Harvard, Vancouver, ISO, and other styles

47

Newman, John M. "The effects of synchronous voice and video tools on acceptance of online communications by students in undergraduate technology courses /." abstract and full text PDF (free order & download UNR users only), 2007. http://0-gateway.proquest.com.innopac.library.unr.edu/openurl?url_ver=Z39.88-2004&rft_val_fmt=info:ofi/fmt:kev:mtx:dissertation&res_dat=xri:pqdiss&rft_dat=xri:pqdiss:3276959.

Full text

Abstract:

Thesis (Ph. D.)--University of Nevada, Reno, 2007.
"May, 2007." Includes bibliographical references (leaves 85-104). Online version available on the World Wide Web. Library also has microfilm. Ann Arbor, Mich. : ProQuest Information and Learning Company, [2007]. 1 microfilm reel ; 35 mm.

APA, Harvard, Vancouver, ISO, and other styles

48

Kamat, Narasinha. "A delay-efficient rerouting scheme for voice over ip traffic." [Gainesville, Fla.] : University of Florida, 2002. http://purl.fcla.edu/fcla/etd/UFE0000548.

Full text

APA, Harvard, Vancouver, ISO, and other styles

49

Bruun, Thomas. "Evaluation of Telecom Operator Enabled Internet Telephony by Creating a Proof-of-Concept Web Application." Thesis, Norges teknisk-naturvitenskapelige universitet, Institutt for datateknikk og informasjonsvitenskap, 2013. http://urn.kb.se/resolve?urn=urn:nbn:no:ntnu:diva-22978.

Full text

Abstract:

Since the beginning of the 21st century, telecommunication operators have witnessed the arrival of internet services that are increasingly taking over traditional voice and messaging markets. Some operators are realizing that they need to bring their traditional services to the internet, in order to keep up with the modern consumer. In this thesis, I create an application allowing phone calls to be placed from an internet browser using a customer's existing phone number. The application extends the existing service Talk+, currently developed as native phone applications by Telenor Comoyo AS. The planning and implementation is conducted by examining modern technologies available to web developers, and by utilizing these in a JavaScript application. The application is evaluated using a set of requirements derived from the existing implementation of Talk+ for iPhone. The result is a working browser telephony application for outgoing calls, with the same quality of voice and delay as on a cellular phone. The thesis therefore concludes that it is possible to create such an application with the technology available, but underlines that the technology is at an early stage, and that it lacks the support from several large browser vendors.

APA, Harvard, Vancouver, ISO, and other styles

50

Jacobs, Ashley. "Investigating call control using MGCP in conjuction with SIP and H.323." Thesis, Rhodes University, 2005. http://hdl.handle.net/10962/d1006516.

Full text

Abstract:

Telephony used to mean using a telephone to call another telephone on the Public Switched Telephone Network (PSTN), and data networks were used purely to allow computers to communicate. However, with the advent of the Internet, telephony services have been extended to run on data networks. Telephone calls within the IP network are known as Voice over IP. These calls are carried by a number of protocols, with the most popular ones currently being Session Initiation Protocol (SIP) and H.323. Calls can be made from the IP network to the PSTN and vice versa through the use of a gateway. The gateway translates the packets from the IP network to circuits on the PSTN and vice versa to facilitate calls between the two networks. Gateways have evolved and are now split into two entities using the master/slave architecture. The master is an intelligent Media Gateway Controller (MGC) that handles the call control and signalling. The slave is a "dumb" Media Gateway (MG) that handles the translation of the media. The current gateway control protocols in use are Megaco/H.248, MGCP and Skinny. These protocols have proved themselves on the edge of the network. Furthermore, since they communicate with the call signalling VoIP protocols as well as the PSTN, they have to be the lingua franca between the two networks. Within the VoIP network, the numbers of call signalling protocols make it difficult to communicate with each other and to create services. This research investigates the use of Gateway Control Protocols as the lowest common denominator between the call signalling protocols SIP and H.323. More specifically, it uses MGCP to investigate service creation. It also considers the use of MGCP as a protocol translator between SIP and H.323. A service was created using MGCP to allow H.323 endpoints to send Short Message Service (SMS) messages. This service was then extended with minimal effort to SIP endpoints. This service investigated MGCP’s ability to handle call control from the H.323 and SIP endpoints. An MGC was then successfully used to perform as a protocol translator between SIP and H.323.

APA, Harvard, Vancouver, ISO, and other styles

Dissertations / Theses on the topic 'Internet telephony'

Create a spot-on reference in APA, MLA, Chicago, Harvard, and other styles