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Dissertations / Theses on the topic 'IP technology'

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Published: 30 May 2022
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1

Landström, Sara. "TCP/IP technology for modern network environments." Doctoral thesis, Luleå tekniska universitet, Datavetenskap, 2008. http://urn.kb.se/resolve?urn=urn:nbn:se:ltu:diva-25736.

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To facilitate the merging of wireless access technologies and the traditional Internet, the core protocols for data communication should be robust and have low overhead. In this thesis, we propose refinements to the Transmission Control Protocol (TCP) that improve its cost efficiency over wireless links.TCP is unable to distinguish between congestion and error induced losses, reordered, or delayed segments. A reordering robust TCP would make it possible to simplify network elements, now performing explicit actions to prevent reordering, and open up for deployment of new technologies that naturally cause reordering. We propose TCP-Aix; a set of TCP modifications that improves the robustness of TCP to reordering and delay spikes. TCP-Aix decouples loss recovery and congestion control actions. We also present an algorithm called the winthresh algorithm for computing a duplicate acknowledgment threshold based on TCP buffer space and current send window size. The results show that TCP-Aix with the winthresh algorithm is able to maintain almost constant performance even in scenarios frequently displaying long reordering durations. It is also fair towards competing standards-compliant TCP flows.In wireless networks, where the links out of efficiency constraints are more error prone than wired links, the error and the reordering sensitivity of TCP have motivated link layer protocols that perform retransmissions and enforce in-order delivery. We investigate the potential gains of using a reordering robust TCP, like TCP-Aix, with a wireless link layer that allows out-of-order delivery, compared to using in-order delivery with a standardscompliant TCP. We found that the smoothness of TCP is strongly affected by the link layer configuration. In-order delivery leads to burstier traffic and larger network layer buffering needs, than out-of-order delivery and TCP-Aix. The interference and power consumption in foremost wireless networks make it important to reduce the communication overhead. The TCP receiver acknowledges each or every second segment. We study how to reduce the acknowledgment frequency while preserving throughput performance also in wireline networks where frequent acknowledgments generally are not problematic. To preserve throughput, the sender should use byte counting and be involved in determining the acknowledgment frequency. The results show that acknowledging four segments per send window is sufficient to maintain throughput performance also in wireline scenarios. This indicates that a lower acknowledgment frequency than provided through the delayed acknowledgment algorithm is possible today for general usage.A key service to the successful merging of traditional Internet technology and wireless cellular networks is Voice over IP (VoIP). Channels to be shared by both VoIP and TCP-based traffic is being considered for wireless cellular systems. It is challenging to provide VoIP over a shared wireless cellular channel, because VoIP is a low bitrate service with high demands on channel availability to bound the delay. The scheduling algorithm, controlling access to the channel, is central to achieve efficiency as well as to satisfy user demands. We study how a scheduler for a mix of VoIP and interactive (TCP-based) traffic should be designed for High Speed Downlink Packet Access (HSDPA). In particular, we find that slowly increasing the priority of a VoIP session relative TCP-based services is important to take advantage of the varying network conditions.

Godkänd; 2008; 20080520 (ysko)

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2

Kenney, Joshua D., Myron L. Moodie, Gary L. Ragsdale, and Thomas B. Grace. "Technology Trades in IP-Based Telemetry Networks." International Foundation for Telemetering, 2008. http://hdl.handle.net/10150/606196.

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ITC/USA 2008 Conference Proceedings / The Forty-Fourth Annual International Telemetering Conference and Technical Exhibition / October 27-30, 2008 / Town and Country Resort & Convention Center, San Diego, California
The integrated Network Enhanced Telemetry (iNET) project established a test article standards working group to define open standards for network components and interfaces for test articles in the aeronautical test environment. Its aim is to utilize the growth of Internet technologies for telemetry networks and ensure interoperability among network devices. This paper describes the technology background and the current technology trades of an IP-based network paradigm used in producing standards for test article networks. Specifically, the paper will include descriptions of selected network technologies as applied to test data and time distribution within test articles.
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3

Fan, Qiang. "IP telephony service creation using SIP Servlet technology." Mémoire, [S.l. : s.n.], 2006. http://savoirs.usherbrooke.ca/handle/11143/1308.

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4

Immonen, Mia. "SIGTRAN : Signaling over IP -- a step closer to an all-IP network." Thesis, KTH, Mikroelektronik och Informationsteknik, IMIT, 2005. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-92285.

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The mass popularization of telecommunication services in recent years have resulted in a heavily loaded signaling network. The Signaling System number 7 (SS7) is used in fixed and wireless networks and is needed for call control and services such as caller ID, roaming, and for sending SMS. The traditional SS7 networks are expensive to lease and to expand, hence a new suite of protocols have been designed to carry signaling messages over IP. This suite contains a transport protocol called Stream Control Transmission Protocol (SCTP) and various user adaptation layer protocols such as M2PA, M2UA, M3UA, and SUA. To transport the highly loss and delay sensitive signaling messages over IP, it is mandatory that the transport protocol meets the high performance requirements of SS7. Not before the IP-solution has been tested in detail, will it replace significant parts of the national telephone network. In this thesis, the failover duration in the case of link failure was tested using the feature of SCTP called multi-homing. The results suggest that carrying SS7 signaling traffic over IP is possible, since the failover duration does not exceed the required limit.
Under de senaste åren har telekommunikationstjänster blivit allt mer populära, vilket har lett till ett tungt belastat signaleringsnätverk. The Signaling System number 7 (SS7) används i fasta och trådlösa nätverk och behövs för att kontrollera telefonsamtal och för tjänster såsom caller ID, roaming och för att skicka SMS. De traditionella SS7- nätverken är dyra att hyra och att expandera, varför en ny grupp av protokoll har designats för att bära signaleringsmeddelanden över IP. De nya protokollen innehåller ett transportprotokoll som heter Stream Transmission Control Protocol (SCTP) och flera adaptionslagerprotokoll, bl.a. M2PA, M2UA, M3UA och SUA. För att transportera de förlust- och förseningskänsliga signaleringsmeddelandena över IP, måste transportprotokollet möta de höga krav som SS7 har. Inte förrän IP-lösningen har testats ingående, kommer den att ersätta betydelsefulla delar av det nationella telefonnätet. I detta examensarbete har failovertiden mätts då en nätverkslänk mellan två noder har utsatts för ett avbrott. Resultaten pekar på att det är möjligt att bära SS7-trafik över IP eftersom failovertidskraven inte överstigs.
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5

Sjöstedt, Magnus, and Oskar Bergquist. "IP Telephony : A Swedish Perspective." Thesis, KTH, Mikroelektronik och Informationsteknik, IMIT, 2003. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-93111.

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The aim of this Masters Thesis project is to give the Swedish National Post and Telecom Agency, Post- och Telestyrelsen (PTS), an updated view on I P Telephony relative to the Swedish consumer market. The basic questions were raised by PTS and the focus is on the relevant topics from the agency’s point of view. PTS is primarily interested in understanding what state IP Telephony is in and furthermore what IP Telephony can be used for in practice. What are the possibilities of different implementations and what will be their advantages and disadvantages? Despite being on the scene for many years, IP Telephony is now on the verge of a break through. With the creation of gateways between IP and PSTN, various IP Telephony devices, and with the introduction of SIP (described in RFC2543) as a standard signalling protocol, perhaps today, IP Telephony has a significant potential on the consumer market. Many argue that 2003 will be the year that IP Telephony blossoms. The fact that anyone can be their own operator, the role of the traditional operator versus the new operators, the separation of content and signalling transmission, as well as the efficient use of bandwidth are some of the topics covered in this report.</p> Due to the introduction of The Electronic Communications Act that comes into effect the 25th of July 2003, replacing The Telecommunications Act and The Radiocommunications Act, much of the focus in this report lies on regulatory issues. However, in order to get an insight in to the regulatory issues it is important to understand the underlying technology of IP Telephony as well as areas such as security, robustness, privacy, and emergency calls. Through a market analysis, an updated overview of the market for IP Telephony will be given, along with plausible future scenarios. This report will provide the reader with answers not by focusing on the theoretical details of the technology itself, but rather in terms of it’s practical use and limitations.
Syftet med denna rapport är att ge Post- och Telestyrelsen (PTS) en uppdaterad bild av den svenska IP-telefonimarknaden för privatpersoner. Då examensarbetet utfördes för PTS togs de grundläggande frågeställningarna fram av myndigheten och fokus för examensarbetet lades således på ämnen relevanta för PTS. Trots att IP-telefoni har varit på tapeten i flera år, har tekniken ännu inte haft sitt genombrott på privatmarknaden. Existensen av slussar mellan IP och PSTN, samt utvecklingen av IP-telefoniutrustning och standardiserade protokoll (såsom SIP beskriven i RFC2543), har givit IP-telefoni förutsättningen att ta en signifikant roll på den svenska privatmarknaden. Många anser att 2003 är året då IP-telefoni kommer att ha sitt genombrott. Utvecklingen och ändringen av den traditionella operatörsrollen, separationen av signaltrafik och samtalstrafik samt den effektiva och variabla användningen av bandbredd, är några av de faktorer som ger upphov till spännande frågeställningar. Introduktionen av lagen för elektronisk kommunikation som träder i kraft den 25 Juli 2003, och ersätter både telelagen och lagen om radiokommunikation, har gjort att fokus till stor del riktas på de regulativa frågorna inom området. För att sätta sig in i dessa regulativa frågor krävs dock god insikt i tekniken bakom IP-telefoni samt förståelse i områden såsom säkerhet, robusthet, QoS och nödsamtal. En marknadsanalys bidrar med en uppdaterad bild av marknaden för IP-telefoni samt möjliga framtidscenarion. Rapporten förser läsaren med svar genom att fokusera på teknikens praktiska användning och begränsningar snarare än att fokusera på de teoretiska detaljerna.
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6

Johansson, Henrik. "Realtidsstyrning av IP-kamera." Thesis, Automatic Control, 2009. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-19737.

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The final thesis named Tracking in Sensor Networks was created by the company Instrument control Sweden. The thesis work is to create a plug-in to an already developed software pro- gram, SkyView. Via the software program one should be able to control an IP-camera from AXIS, model 215 PTZ, with the appropriate protocol over a network.  An Application Programming Interface, API, was used to easily control the camera through SkyView. The interface is called VAPIX and is used to all cameras from AXIS. VAPIX sup- ports Hyper Text Transfer Protocol, HTTP, and Real Time Streaming Protocol, RTSP.  The video was sent through the camera with RTSP, thus an investigation resulted in that the protocol was best suited at streaming video. Steering commands to the camera was sent via HTTP. A small program was written, which was then integrated to SkyView. Features to be able to control the camera with the keyboard, mouse and a joystick were implemented. Zoom, bright- ness, focus and some more features were also added. The problem with the delay should be dealt with as soon as possible.

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7

García, Hijes Raúl. "Corporate Wireless IP Telephony." Thesis, KTH, Mikroelektronik och Informationsteknik, IMIT, 2005. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-92268.

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IP telephony is defined as the transport of telephony calls over an IP network. IP telephony exploits the integration of voice and data networks. However, enterprises are still reluctant to deploy IP telephony despite the potential increase in productivity and reduction of costs. The principal concerns are: can IP telephony provide the same level of performance in terms of security, reliability, and scalability as traditional telephony? If so, are its proclaimed benefits such as flexibility and mobility cost-effective? The aim of this thesis is to analyze how to deploy IP telephony in large corporations - while providing the necessary security and facilitating mobility. Through the different parts of this thesis, we will analyze the applicable technologies, along with their integration and management. We will focus on the essential requirements for an enterprise of scalability, reliability, flexibility, high-availability, and cost-effectiveness. The massive changes brought about due to the deregulation of telecommunications in nearly all countries, the increasingly global nature of business, and the progressively affordable and power technology underlying information and communication technologies have lead to increasing adoption of IP telephony by residential and commercial users. This thesis will examine these technologies in the context of a very large distributed corporation.
IP telefoni är definierat som transporten av telefon samtal genom ett IP nätverk. IP telefoni utnyttjar integrationen av tal och data nätverk. Dock är affärsföretag fortfarande motsträviga till att införa IP telefoni trots potentiell ökning i produktivitet och minskade kostnader. Huvud bekymren är: kan IP telefoni tillhandahålla samma nivå av prestanda med avseende på säkerhet, tillförlitlighet, och skalbarhet som traditionell telefoni? Och i så fall, är dom proklamerade fördelarna flexibilitet och rörlighet kostnadseffektiva? Målet för detta examensarbete är att analysera hur IP telefoni kan införas i stora affärsföretag - medan samtidigt tillhandahålla nödvändig säkerhet och främja rörlighet. Genom olika delar av detta examensarbete, analyserar vi tillämpliga teknologier, inklusive deras integrering och skötsel. Vi kommer att fokusera på de grundläggande kraven för ett affärsföretag gällande skalbarhet, tillförlitlighet, flexibilitet, hög tillgänglighet, och kostnadseffektivitet. Dom massiva förändringarna frambringade i och med avregleringen av telekommunikation i stort sett alla länder, affärsverksamhetens alltmer globala natur, och de progressivt kostnadseffektiva och kraftfulla underliggande teknologier bakom informations och kommunikations system har lett till ökande adoptering av IP telefoni av både privata och kommersiella användare. Detta examensarbete undersöker relevanta teknologier i samband med mycket stora utbredda affärsföretag.

Exchange student from Centro Politecnico Superior (University of Zaragoza, Spain).

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8

Yang, Yunfeng. "Low Power UDP/IP Accelerator for IM3910 Processor." Thesis, KTH, Skolan för informations- och kommunikationsteknik (ICT), 2012. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-92241.

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Due to their attractive flexibility and high productivity, general purpose processors (GPPs) are found to be spreading over large domain of applications. The growing complexity of modern applications results in high performance demands and as a response, several solutions have came to fulfill these demands. One of these solutions is to couple the GPP with a hardware accelerator to off-load critical functionalities. In this thesis, A UDP/IP hardware accelerator is build to and coupled with an existing GPP with DMA interface, namely IM3910 form Imsys Technology AB in Stockholm, Sweden. The main goal of this thesis is to investigate the semantics of coupling the accelerator with IM3910 and to characterize its area, performance and power consumption. Building this UDP/IP accelerator started from an initial version taken from "OpenCore" and then it was completed and optimized to suit the project needs. After verifying the accelerator at RTL, it was prototyped using Altera FPGA and connected to IM3910 through the DMA. In the last step, the accelerator was synthesized to gate-level netlist using 90nm technology library and characterized in-terms of area, performance and power consumption.
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9

Derakhshanno, Homayoun. "Voice over IP over GPRS." Thesis, KTH, Kommunikationssystem, CoS, 2008. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-91856.

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The Voice over IP (VoIP) technology has become prevalent today due to its lower cost than traditional telephony and its ability to support new value-added services. Additionally, the increasing availability of wireless internet access has led to research studies examining the combination of wireless network access with voice over IP. With the widespread availability of advanced mobile phones and Pocket PCs, the need for VoIP applications on these mobile platforms is tangible. To enable this, we need to evaluate the current wireless access technologies to see if they can support the necessary traffic and implement software to offer these VoIP services to users. In order to easily implement an IP-based service on GSM technology, we should use the GPRS service provided by the GSM operators. In this thesis, we evaluate Voice over IP service over GPRS in terms of feasibility and quality. Following this we ported a locally developed VoIP program to a Pocket PC (with GSM SIM-card support) which runs Microsoft’s Windows Mobile in order to provide suitable software as needed to offer the service from such a portable device.
VoIP tekniken har blivit en rådande teknik numera på grund av dess lägre kostnader och mervärdestjänster som erbjuds jämfört med traditional telefoni. Samtidigt som tendensen mot mer tillgänglig trådlöst internet har underlättat och därmed driver mera studier inom dessa områden. Den allt mer utbredda användningen av avancerade mobiltelefoner och handdatorer numera har lett till ökat behov av att använda VoIP tekniken för dessa mobila utrustningar är alltmer kännbar. För att möjliggöra användadet av VoIP tekniken så behöver vi först och främst utvärdera dagens existerande teknologier för att stödja iden och för det andra måste vi kunna implementera en mjukvara vilket kan erbjuda olika typer av tjänster för slutanvändaren. För att kunna använda en IP-baserad tjänst på GSM teknologin så måste vi använda oss utan GPRS tjänster som tillhandahålls av GSM opratörer. I detta examens arbete kommer vi att utvärdera VoIP tjänster på GPRS när det gäller kvalitet och möjligheter. Därefter kommer vi att Portning en VoIP mjukvara till en handdator (utrustad med GSM sim-kort) vilket har windows Mobile operativsystemet som erbjuder en rad olika tjänster.
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10

Grace, Thomas, and Dave Hodack. "VEHICLE NETWORK TECHNOLOGY DEMONSTRATION." International Foundation for Telemetering, 2007. http://hdl.handle.net/10150/604401.

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ITC/USA 2007 Conference Proceedings / The Forty-Third Annual International Telemetering Conference and Technical Exhibition / October 22-25, 2007 / Riviera Hotel & Convention Center, Las Vegas, Nevada
iNET is a project tasked to foster advances in networking and telemetry technology to meet emerging needs. This paper describes one objective of the project, which is standardization and interoperability. It begins to explore issues for achieving a level of interoperability among differing vendor’s hardware such as data acquisition units, data recorders, video systems, transceivers, and network encryption. Specifically, this paper addresses the expansion of the current demonstration system with the addition of multiple vendor data acquisition units. It will also attempt to address the level of standardization necessary for achieving interoperability while still enabling vendors to add their value added contributions into their products.
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11

Blain, Emmanuel. "Sports over IP : dynamics and perspectives." Thesis, Massachusetts Institute of Technology, 2010. http://hdl.handle.net/1721.1/59669.

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Thesis (S.M. in Technology and Policy)--Massachusetts Institute of Technology, Engineering Systems Division, Technology and Policy Program, 2010.
Cataloged from student submitted PDF version of thesis.
Includes bibliographical references (p. 144-149).
Technological advances are allowing the use of the Internet Protocol and the Internet infrastructure mainly built by cable and telecom operators for broadcasting purposes in the U.S. This disruption raises a business issue for the existing value chain, as new entrants are upsetting the existing value chain for video delivery. This thesis will focus on the implications of online video delivery for Sports content in the United States. The general question that is addressed in this thesis is: How would a shift toward Internetbased delivery systems influence the current value chain for Sports broadcasting? Case studies will first show that the broadcasting industry is prone to disruption, and that online video delivery has the potential to upset the existing value chain, and the business models associated with it. The question can be then divided into three sub-categories, which deal with the technical issues, regulatory needs, and business model changes that Sports over IP may cause -- which are explored in separate chapters in the second part of this thesis. In the last part of the thesis, a system dynamics model is used to study the technological and regulatory conditions under which the market will tip towards predominance of broadcasters by content owners in the post-disruption value chain. Building the model with the standard procedure (described in the thesis) will give insights on what the main dynamics are, and how interrelations and feedbacks among those main dynamics can influence the whole system.
by Emmanuel Blain.
S.M.in Technology and Policy
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Syed, Muhammad Yasir Jafri. "IP BASED PATIENT MONITORING SYSTEMS." Thesis, Högskolan i Borås, Institutionen Ingenjörshögskolan, 2010. http://urn.kb.se/resolve?urn=urn:nbn:se:hb:diva-19797.

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13

Aktalay, Banu. "Intellectual Property Management Strategy In New Technology-based Start-up Companies." Master's thesis, METU, 2004. http://etd.lib.metu.edu.tr/upload/12605598/index.pdf.

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To draw up an intellectual property management strategy is one of the most important tasks to do when starting up technology-based companies, which play a very important role in the knowledge-based economies. IP management due to protection must be regarded as a strategic asset aimed at improving the competitive advantages, increasing the revenue of a technology-based start up company and encouraging to continue to develop new technologies, by securing a technological platform for a future development, preventing competitors from gaining access to emerging markets, creating retaliatory power against competitors and preventing innovative products from being plagiarized. Through this study the aim is forming a guide including why and how an IP management strategy develop and implement to a new technology-based start up company. Beside these it is proved that from the properties that characterize the start-up configuration of a high-tech firm there can be derived three organizational archetypes of firms each of which requires different IP management strategies.
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Abad, Caballero Israel Manuel. "Secure Mobile Voice over IP." Thesis, KTH, Mikroelektronik och Informationsteknik, IMIT, 2003. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-93113.

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Voice over IP (VoIP) can be defined as the ability to make phone calls and to send faxes (i.e., to do everything we can do today with the Public Switched Telephone Network, PSTN) over IP−based data networks with a suitable quality of service and potentially a superior cost/benefit ratio. There is a desire to provide (VoIP) with the suitable security without effecting the performance of this technology. This becomes even more important when VoIP utilizes wireless technologies as the data networks (such as Wireless Local Area Networks, WLAN), given the bandwidth and other constraints of wireless environments, and the data processing costs of the security mechanisms. As for many other (secure) applications, we should consider the security in Mobile VoIP as a chain, where every link, from the secure establishment to the secure termination of a call, must be secure in order to maintain the security of the entire process. This document presents a solution to these issues, providing a secure model for Mobile VoIP that minimizes the processing costs and the bandwidth consumption. This is mainly achieved by making use of high− throughput, low packet expansion security protocols (such as the Secure Real−Time Protocol, SRTP); and high−speed encryption algorithms (such as the Advanced Encryption Standard, AES). In the thesis I describe in detail the problem and its alternative solutions. I also describe in detail the selected solution and the protocols and mechanisms this solution utilizes, such as the Transport Layer Security (TLS) for securing the Session Initiation Protocol (SIP), the Real−Time Protocol (RTP) profile Secure Real−Time Protocol (SRTP) for securing the media data transport , and the Multimedia Internet KEYing (MIKEY) as the key−management protocol. Moreover, an implementation of SRTP, called MINIsrtp, is also provided. The oral presentation will provide an overview of these topics, with an in depth examination of those parts which were the most significant or unexpectedly difficult. Regarding my implementation, evaluation, and testing of the model, this project in mainly focused on the security for the media stream (SRTP). However, thorough theoretical work has also been performed and will be presented, which includes other aspects, such as the establishment and termination of the call (using SIP) and the key−management protocol (MIKEY).
Voice over IP (VoIP) kan defineras som förmågan att göra ett telefonsamtal och att skicka fax (eller att göraallting som man idag kan göra över det publika telefonnätet) över ett IP−baserat nätverk med en passande kvalitet och till lägre kostnad, alternativt större nytta. VoIP måste tillhandahållas med nödvändiga säkerhetstjänster utan att teknikens prestanta påverkas. Detta blir allt viktigare när VoIP används över trådlösa länktekniker (såsom trådlösa lokala nätverk, WLAN), givet dessa länkars begränsade bandbredd och den bearbetningkraft som krävs för att exekvera säkerhetsmekanismerna. Vi måste tänka på VoIPs säkerhet likt en kedja där inte någon länk, från säker uppkoppling till säker nedkoppling, får fallera för att erhålla en säker process. I detta dokument presenteras en lösning på detta problem och innefattar en säker modell för Mobile VoIP som minimerar bearbetningskostnaderna och bandbreddsutnyttjandet. Detta erhålls huvudsakligen genom utnyttjande av säkerhetsprotokoll med hög genomströmning och låg paketexpansion, såsom "Secure Real− time Protocol" (SRTP), och av krypteringsprotokoll med hög hastighet, såsom "Advanced Encryption Standard" (AES). I detta dokument beskriver jag problemet och dess alternativa lösningar. Jag beskriver också den valda lösningen och dess protokoll och mekanismer mer detaljerat, till exempel "Transport Layer Security" (TLS) för att säkra "Session Initiation Protocol" (SIP), SRTP för att skydda transporten av data och "Multimedia Internet KEYing" (MIKEY) för nyckelhantering. En implementation av SRTP, kallad MINIsrtp, finns också beskriven. Beträffande praktiskt arbete och tester av lösningsmodellen har detta projekt fokuserats på skyddandet av datatransporten (SRTP), dess implementation och prestanda. Emellertid har en grundlig teoretisk undersökning genomförts, vilken innefattar andra aspekter såsom telefonsamtalets uppkoppling och nedkoppling (med hjälp av SIP) och valet av passande nyckelhanteringsprotokoll (MIKEY) för att stödja SRTP.
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Marco, Arranz Carlos Aurello. "IP Telephony : Peer-to-peer versus SIP." Thesis, KTH, Mikroelektronik och Informationsteknik, IMIT, 2005. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-92288.

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In recent years dramatic technology developments have exploited the development of better transmission media and allowed for broad internet penetration. This in turn has fostered the growth of IP telephony calls, i.e., Voice over IP (VoIP). New VoIP products are introduced almost daily, each seeking an opportunity in the market. Some of these products are free - thus putting pressure on other vendors. A good example of a commercial VoIP product is Skype. It is possibly the most important one as it has gained more than 3 millions users in approximately 2 years time. In contrast, Minisip is a non-commercial implementation of SIP developed by students at KTH. These programs are based on different architectures. While Skype is said to be based on a peer-to-peer protocol, Minisip utilized the Session Initiation Protocol (SIP) protocol. The aim of this thesis was to evaluate these two VoIP programs not only in terms of development, but also in terms of the quality of service and user perceived voice quality. The study of efficiency, usability, and installation of both are also in the scope of this thesis. The devices used for the evaluation included a HP iPAQ 5550, two PCs running in RedHat Linux 9, and a laptop running Microsoft Windows XP.
På senare år har den dramatiska teknikutvecklingen exploaterat utvecklingen av bättre överföringsmedia samt möjliggjort för en bred Internetpenetrering. Det i sin tur har gynnat ökningen av telefonsamtal med IP-telefoner, d.v.s. Voice over IP (VoIP). Nya VoIP produkter introduceras nästan dagligen och varje produkt söker sin möjlighet på marknaden. Vissa av dessa produkter är gratis och sätter alltså press på andra försäljarna. Ett bra exempel på en VoIP produkt är Skype. Det är möjligtvis den viktigaste produkten då den har fått tre miljoner användare på ungefär två års tid. I kontrast till detta finns Minisip som är en icke kommersiell implementation av SIP, utvecklad av studenter från KTH. Dessa program är baserade på olika arkitekturer. Medan Skype är baserat på ett peer-to-peer protokoll, utnyttjar Minisip protokollet Session Initiation Protocol (SIP) Protocol. Syftet med denna avhandling var att evaluera dessa två VoIP program, inte bara i termer av utveckling utan också i termer av ”quality of service” och hur användaren uppfattar röstkvaliteten. Studien innefattar också effektivitet, användarvänlighet och installation av de båda programmen. Enheterna som användes under denna evaluering var en HP iPAQ 5550, två pc:s med Linux Red Hat 9 samt en bärbar pc med Windows XP.
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16

Short, Kristin Ilene 1971. "Towards integrated Intranet services : modeling the costs of corporate IP technology." Thesis, Massachusetts Institute of Technology, 1997. http://hdl.handle.net/1721.1/43530.

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17

Aguirre, Zoe, and Gabe Beltran. "WSMR Telemetry Capabilities: Today's Technology in Telemetry." International Foundation for Telemetering, 2008. http://hdl.handle.net/10150/606237.

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ITC/USA 2008 Conference Proceedings / The Forty-Fourth Annual International Telemetering Conference and Technical Exhibition / October 27-30, 2008 / Town and Country Resort & Convention Center, San Diego, California
White Sands Missile Range is the largest overland test range in North America occupying over 3,200 square miles in Southern New Mexico and nearby territory. One of the most critical test elements at White Sand Missile Range is it's capabilities in the telemetry field. Much significant advancement in technology has given WSMR and the entire electronics world the ability to achieve new levels of data acquisition that were not achievable a decade ago. And as attention to our nation's defense is of high priority, White Sands Missile Range provides to highest levels of telemetry competence in the Western Hemisphere.
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18

Ozdem, Mehmet. "Video Distribution Over Ip Networks." Master's thesis, METU, 2007. http://etd.lib.metu.edu.tr/upload/12608187/index.pdf.

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As applications like IPTV and VoD (Video on demand) are gaining popularity, it is becoming more important to study the behavior of video signals in the Internet access infrastructures such as ADSL and cable networks. Average delay, average jitter and packet loss in these networks affect the quality of service, hence transmission and access speeds need to be determined such that these parameters are minimized. In this study the behavior of the above mentioned IP networks under variable bit rate (VBR) video traffic is investigated. ns-2 simulator is used for this purpose and actual as well as artificially generated signals are applied to the networks under test. Variable bit rate (VBR) traffic is generated synthetically using ON/OFF sources with ON/OFF times taken from exponential or Pareto distributions. As VBR video shows long range dependence with a Hurst parameter between 0.5 and 1, this parameter was used as a metric to measure the accuracy of the synthetic sources. Two different topologies were simulated in this study: one similar to ADSL access networks and the other behaving like cable distribution network. The performance of the networks (delay, jitter and packet loss) under VBR video traffic and different access speeds were measured. According to the obtained results, minimum access speeds in order achieve acceptable quality video delivery to the customers were suggested.
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19

Bilien, Johan. "Key Agreement for Secure Voice over IP." Thesis, KTH, Mikroelektronik och Informationsteknik, IMIT, 2003. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-93069.

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This thesis reviews the usual properties and requirements for key agreement protocols. It then focuses on MIKEY, a work-in-progress protocol designed to conduct key agreements for secure multimedia exchanges. The protocol was implemented and incorporated in a SIP user agent - minisip. This implementation was used to measure the additional delay required for key exchange during call establishment. Finally, some schemes are proposed regarding the use of MIKEY in advanced VoIP scenarios, such as conferences and terminal mobility.
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20

Parnes, Peter. "The mStar environment : scalable distributed teamwork using IP multicast." Licentiate thesis, Luleå tekniska universitet, Datavetenskap, 1997. http://urn.kb.se/resolve?urn=urn:nbn:se:ltu:diva-18409.

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This thesis addresses the question of how a scalable, distributed teamwork environment should be designed and realized. Central design criteria includes that the system should be scalable and robust, allow for easy access and be symmetric. The system should allow for project team members to collaborate even though they are not located at the same physical location. The resulting system presented in this thesis, called the mStar environment have been created to address exactly these questions. mStar is scalable and robust through the usage of standard networks and IP-multicast, it allows for easy access as it is desktop based and finally it is symmetric allowing for easy peer-communication. mStar includes support for desktop conferencing, including mAudio for audio, reuse of the MBone Vic tool for video, mWB for whiteboard, mChat for text based chat and mVote for voting. It also supports distributed synchronized presentations using the WWW and the mWeb application. As all traffic is network and IP-multicast based it allows for easy recording and playback of teamwork sessions using the mMOD application. To allow for easy access to users behind non-multicast capable network segments (primarily modem and ISDN), mTunnel was created. It allows for tunneling and transformation of the traffic. Another member of the mStar environment is Director for remote control of video equipment. mStar also includes support for easy creation of new teamwork tools and applications using the Tunable Multicast Platform - /TMP and the Generic Agent Architecture. The mStar environment can be used and is being used on a daily basis for electronic meetings, distance education and lectures, and daily teamwork. The usage mStar creates group awareness between project members and helps users from not becoming isolated from their department and project team. mStar allows for usage 24 hours a day and have resulted in, among other things, a new usage patterns, which resembles electronic corridors more than specific meetings, where users can and do meet spontaneously to talk about anything they want, but also overhear other interesting and important conversations just as in a physical office corridor.
Godkänd; 1997; 20061004 (ysko)
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21

Rash, James, Keith Hogie, and Ralph Casasanta. "INTERNET TECHNOLOGY FOR FUTURE SPACE MISSIONS." International Foundation for Telemetering, 2002. http://hdl.handle.net/10150/606363.

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International Telemetering Conference Proceedings / October 21, 2002 / Town & Country Hotel and Conference Center, San Diego, California
Ongoing work at National Aeronautics and Space Administration Goddard Space Flight Center (NASA/GSFC), seeks to apply standard Internet applications and protocols to meet the technology challenge of future satellite missions. Internet protocols and technologies are under study as a future means to provide seamless dynamic communication among heterogeneous instruments, spacecraft, ground stations, constellations of spacecraft, and science investigators. The primary objective is to design and demonstrate in the laboratory the automated end-to-end transport of files in a simulated dynamic space environment using off-the-shelf, low-cost, commodity-level standard applications and protocols. The demonstrated functions and capabilities will become increasingly significant in the years to come as both earth and space science missions fly more sensors and the present labor-intensive, mission-specific techniques for processing and routing data become prohibitively. This paper describes how an IP-based communication architecture can support all existing operations concepts and how it will enable some new and complex communication and science concepts. The authors identify specific end-to-end data flows from the instruments to the control centers and scientists, and then describe how each data flow can be supported using standard Internet protocols and applications. The scenarios include normal data downlink and command uplink as well as recovery scenarios for both onboard and ground failures. The scenarios are based on an Earth orbiting spacecraft with downlink data rates from 300 Kbps to 4 Mbps. Included examples are based on designs currently being investigated for potential use by the Global Precipitation Measurement (GPM) mission.
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22

Papazafeiropoulos, Christos. "Facilitating the adoption and use of the IP Multimedia System." Thesis, KTH, Kommunikationssystem, CoS, 2009. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-91530.

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The IP Multimedia Subsystem (IMS) is still under development and not widely adopted in the market. Some companies are reluctant to deploy IMS and some telecommunications vendors believe that IMS will not achieve a desirable market share. The purpose of this thesis work is to give a boost to this technology (i.e., to accelerate its market growth) by providing the community (both developers and operators who might adopt this technology) with an evaluation of the Ericsson Java Application Programming Interfaces (APIs) called Mobile Java Communication Framework (MJCF) APIs. Developers with or even without knowledge of the IMS architecture and signaling should be able to use these interfaces in order to develop applications on top of IMS. A client-server application is designed and implemented to facilitate this evaluation and to serve as an example for others. The motivation behind this application is the every day needs of the people who search for discounts while they are shopping. Users set up their profile by specifying their preference concerning discounts for specific products; while shop owners publish discounts. When a user is near a store which offers interesting discounts (i.e., discounts that match their profile) new notifications will be sent to his/her mobile device. This application exploits the MJCF APIs and uses several of its basic functions; specifically subscriptions, messages, notifications, and publications are some of the messages that can be utilized through these interfaces.</p> Throughout the application development, bugs were found in the APIs and corrections were suggested for the documentation. Measurements were made in order to evaluate the memory utilization and delay associated with these APIs. It was observed that the delays added by the APIs are somewhat high and may negatively affect the experience of users. However memory utilization seamed to be low for client applications and quite high for the server side given the resources of today's services and cellular phones.
Systemet IP Multimedia Subsystem (IMS) är under utveckling och är inte vida etablerat än. Nogra företag tvekar inför etablering av IMS och nogra telekomföretag anser att IMS inte kommer uppnå önskad marknadsandel. Syftet med detta examensarbete är att ge denna teknologi en skjuts framåt (d.v.s. att öka marknadstillväxten) genom att tillhandahålla den gemenskap av både utvecklare och operatörer som kan tänka sig ta in denna teknologi, med en utvärdering of Ericsson’s Java Applications Programming Interfaces (APIs) kallade MJCF API. Utvecklare med eller t.o.m. utan kunskap om arkitekturen och signalleringen hos IMS ska kunna använda dessa gränssnitt till att utveckla tjänster på IMS. En klient-server applikation är designade och implementerad för att möjliggöra denna utvärdering och för att agera exempel för andra. Motiveringen bakom denna applikation är det vardagliga behovet hos människor som söker efter rabatter/erbjudanden när de handlar. Användare sätter upp sin profil genom att specificera sina önskemål angående erbjudanden för specifika produkter medan butiksinnehavare publicerar sina erbjudanaden. När en användare är nära en butik som erbjuder någonting interessant (d.v.s. produkter som matchar använadarens profil), så kommer nya notifikationer att anlända till hans/hennes mobil. Denna applikation uttnyttjar MJCF APIet och använder ett flertal av dess basala funktioner; speciellt gällande prenumerationer, meddelanden, notifieringar och publiceringar är några av de meddelanden som möjliggörs genom dessa gränssnitt.</p> Genom applikationsutvecklingen så blev flera buggar i APIerna upptäckta och förbättringar till dokumentationen föreslogs. Mätningar gjordes för att utvärdera minnesåtgången och fördröjningar associerade med dessa APIer. Det observerades att API fördröjningar är något höga och kan påverka användarupplevelsen negativt. Däremot verkade minnesåtgången vara låg på klientsidan och hög på serversidan, givet de resurser dagens tjänster och mobila telefoner förfogar över.
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23

Masiello, Elizabeth. "Service identification in TCP/IP : well-known versus random port numbers." Thesis, Massachusetts Institute of Technology, 2005. http://hdl.handle.net/1721.1/34537.

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Thesis (S.M.)--Massachusetts Institute of Technology, Engineering Systems Division, Technology and Policy Program, 2005.
Includes bibliographical references (p. 49-52).
The sixteen-bit well-known port number is often overlooked as a network identifier in Internet communications. Its purpose at the most fundamental level is only to demultiplex flows of traffic. Several unintended uses of the port number evolved from associating services with a list of well-known port numbers. This thesis documents those unintended consequences in an effort to describe the port number's influence on Internet players from ISPs to application developers and individual users. Proposals and examples of moving away from well-known port numbers to randomly assigned ones are then presented, with analysis of impacts on the political and economic systems on which Internet communication is dependent.
by Elizabeth Masiello.
S.M.
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24

Жураківський, І. В., and І. О. Товкач. "IP-АТС в середовищі інформаційно-комунікаційної системи «ПОЛІДАР»." Thesis, Cумський державний університет, 2016. http://essuir.sumdu.edu.ua/handle/123456789/46392.

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У даний час активно розвивається віртуальна IP телефонія, яка за- безпечує потужний та економічний засіб зв‘язку. З метою розширення функціонального складу інформаційно- комунікаційної системи «ПОЛІДАР» в її середовище за допомогою розробленого модуля взаємодії було інтегровано кросплатформенну програмну VoIP-систему Asterisk.
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25

Peterström, Dan. "IP Multimedia for Municipalities : The supporting architecture." Thesis, KTH, Kommunikationssystem, CoS, 2009. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-91496.

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Fiber deployment is becoming popular and is seen as a way to increase a community’s attractiveness to new inhabitants and companies. A new Open Access network is emerging, leading to a more horizontal network architecture. Combining this architecture with IMS enables developers to easily develop new and attractive services. To facilitate the development of new IMS services there needs to be an easy to use development environment and a reliable hardware/software platform upon which to deploy them. This thesis project will explore the design, development, and evaluation of new IMS applications targeted at municipal networks as well as the service platforms they are deployed on. The thesis will also examined what role IMS plays in Municipalities and why they may need a tailored IMS solution. During the thesis a reference network for Municipalities was put together and tested. Different service platforms were tested and evaluated.
Fiberutbyggnad har blivit mycket populärt och ses som ett sätt att öka en stads attraktivitet för nya invånare och företag. Ett nytt öppen nät håller på att växa fram, detta leder till en mer horisontell nätverksarkitektur. Kombinera denna arkitektur med IMS så kan utvecklare lättare utveckla nya och attraktiva tjänster. För att underlätta utvecklingen av nya IMS-tjänster måste det finnas ett lättanvänd utvecklingsmiljö och en pålitlig hårdvaru/mjukvaru plattform att installera dem på. Detta examensarbete kommer att undersöka design, utveckling och utvärdering av nya IMSlösningar riktade till standsnät samt de tjänsteplattformar de används på. Rapporten kommer även granska vilken roll IMS spelar i stadsnät och varför de kan behöva en skräddarsydd IMS-lösning. Under examensarbetet byggdes och testades ett referensnätverk. Olika tjänsteplatformar testades och utvärderades.
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Arvidsson, Daniel, Joakim Bergqvist, and Robin Bünger. "IP-telefoni : en teknisk undersökning." Thesis, Högskolan Kristianstad, Sektionen för Hälsa och Samhälle, 2011. http://urn.kb.se/resolve?urn=urn:nbn:se:hkr:diva-7977.

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Projektet har till syfte att undersöka den revolutionerande tekniken bakom IP-telefoni. Eftersom tekniken är relativt ung så finns det fortfarande mycket att lära om hur det fungerar och hur den används. Projektet behandlar fördelar och nackdelar samt vilka hot som finns och hur man kan känna sig säker när man använder IP-telefoni till vardags. Det primära objektivet med projektet är att få en helhetsbild och förstå hur tekniken verkligen fungerar. Vidare behandlas telefonens historia då det är viktigt att se var tekniken har sina rötter och hur den utvecklats sedan dess. Som en röd tråd från telefonens teknologi och det publika telefonnätets rötter görs en jämförelse mellan de två teknologierna för att se skillanderna och förmodade fördelar och nackdelar med de båda systemen. Olika sorters protokoll som har ett nära förhållande till IP-telefoni så som SIP, RTP, UDP och IP behandlas. För att få en sanningsenlig bild över hur vardagliga användare upplever IP-telefoni så har en enkätundersökning angående vanor och möjliga problem gjorts hos de anställda på Region Skåne.
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Hasselström, Emil, and Therese Sjögren. "Evaluation of Load Balancing Algorithms in IP Networks : A case study at TeliaSonera." Thesis, Linköping University, Department of Science and Technology, 2005. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-2842.

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The principle of load balancing is to distribute the data load more evenly over the network in order to increase the network performance and efficiency. With dynamic load balancing the routing is undated at certain intervals. This thesis was developed to evaluate load balancing methods in the IP-network of TeliaSonera.Load balancing using short path routing, bottleneck load balancing and load balancing using MPLS have been evaluated. Short path routing is a flow sharing technique that allows routing on paths other than the shortest one.

Load balancing using short path routing is achieved by dynamic updates of the link weights. Bottleneck is in its nature a dynamic load balancing algorithm. Unlike load balancing using short path routing it updates the flow sharing, not the metrics. The algorithm uses information about current flow sharing and link loads to detect bottlenecks within the network. The information is used to calculate new flow sharing parameters. When using MPLS, one or more complete routing paths (LSPs) are defined at each edge LSR before sending any traffic. MPLS brings the ability to perform flow sharing by defining the paths to be used and how the outgoing data load is to be shared among these.

The model has been built from data about the network supplied by TeliaSonera. The model consists of a topology part, a traffic part, a routing part and cost part. The traffic model consists of a OD demand matrix. The OD demand matrix has been estimated from collected link loads. This was done with estimation models; the gravity model and an optimisation model.

The algorithms have been analysed at several scenarios; normal network, core node failure, core link failure and DWDM system failure. A cost function, where the cost increases as the link load increases has been used to evaluate the algorithms. The signalling requirements for implementation of the load balancing algorithm have also been investigated.

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Israel, Dave, Ron Parise, Keith Hogie, and Ed Criscuolo. "SPACE COMMUNICATION DEMONSTRATION USING INTERNET TECHNOLOGY." International Foundation for Telemetering, 2002. http://hdl.handle.net/10150/607486.

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International Telemetering Conference Proceedings / October 21, 2002 / Town & Country Hotel and Conference Center, San Diego, California
This paper presents work being done at NASA/GSFC by the Operating Missions as Nodes on the Internet (OMNI) project to demonstrate the application of Internet communication technologies to space communication. The goal is to provide global addressability and standard network protocols and applications for future space missions. It describes the communication architecture and operations concepts that will be deployed and tested on a Space Shuttle flight in July 2002. This is a NASA Hitchhiker mission called Communication and Navigation Demonstration On Shuttle (CANDOS). The mission will be using a small programmable transceiver mounted in the Shuttle bay that can communicate through NASA’s ground tracking stations as well as NASA’s space relay satellite system. The transceiver includes a processor running the Linux operating system and a standard synchronous serial interface that supports the High-level Data Link Control (HDLC) framing protocol. One of the main goals will be to test the operation of the Mobile IP protocol (RFC 2002) for automatic routing of data as the Shuttle passes from one contact to another. Other protocols to be utilized onboard CANDOS include secure login (SSH), UDP-based reliable file transfer (MDP), and blind commanding using UDP. The paper describes how each of these standard protocols available in the Linux operating system can be used to support communication with a space vehicle. It will discuss how each protocol is suited to support the range of special communication needs of space missions.
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Weir, Malcolm. "TECHNOLOGY CONVERGENCE: OBSERVATIONS ON TRANSITIONAL APPROACHES FOR DATA ACQUISITION IN A TCP/IP ENVIRONMENT." International Foundation for Telemetering, 2007. http://hdl.handle.net/10150/604539.

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ITC/USA 2007 Conference Proceedings / The Forty-Third Annual International Telemetering Conference and Technical Exhibition / October 22-25, 2007 / Riviera Hotel & Convention Center, Las Vegas, Nevada
This paper discusses how IRIG 106 Chapter 10 recording techniques could be employed in a network-centric environment, while maintaining as many of the strengths of the traditional approach. In the course of that discussion, aspects of the published standard which would have to be disregarded or reinterpreted for a network-centric approach to be adopted are illustrated.
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30

Nesh-Nash, Ali. "Voice over IP in a resource constrained environment." Thesis, KTH, Kommunikationssystem, CoS, 2006. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-92257.

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Today, the telecommunication world is focused on mobility. This is popular because since the 1990s most people have integrated their mobile phones into their life. A new factor is the rise of the voice over IP(VoIP) technology, with VoIP over Wireless LANs (WLANs) as the clear next growth area for mobile communications. The purpose of this thesis was to understand how to save power based upon changing when some operations are performed in a VoIP client. In order to do this, we decided to port minisip to an HP iPAQ 5500 Personal Digital Assistant (PDA), in order to explore some of the issues of running such a client on a PDA - due to its constraints with regard to storage, processing power, and battery power. Minisip is a SIP open source user agent running on Linux and Windows. This thesis builds upon earlier theses which showed that minisip can offer a secure communications platform with the latest functions which are desired in a mobile personal VoIP system. However, most of these earlier theses utilized desktop, laptop, or server based system, i.e., with few resources constrains. The focus of this thesis was to examine the case of a highly constrained user platform such as an iPAQ.
Dagens telekommunikationssystem fokuserar på mobilitet. Detta har blivit populärt under 90-talet då mobilitet blev naturligt integrerad i människans vardagliga liv i form av exempelvis mobiltelefoner. Voice over IP (VoIP) har blivit en stor del av dagen teknik där trådlösa system Wireless LANs (WLANs) har blivit en större del av mobilkommunikation. Målet med denna rapport är att förstå hur strömförbrukningen kan minimeras genom att utföra vissa operationer med hjälp av en VoIP-klient. För att åstadkomma detta porterade vi minisip, en SIP agent som är baserad på öppen källkod och körs på Linux och Windows, till en HP iPAQ 5500, en så kallad Personal Digital Assistant (PDA). Vi valde PDAn för att kunna utforska de begränsningar den medför i form av lagringsutrymme, processorkapacitet, och batteri. Denna rapport bygger vidare på tidigare rapporter som visar att minisip kan erbjuda en säker kommunikationsplattform med de senaste funktionerna som önskas i mobila VoIPsystem. De flesta av dessa tidigare rapporter baseras på system med få begränsningar rörande resurser såsom stationära- eller bärbara datorer samt serverbaserade system. Denna rapports fokus är att utforska detta fall i en miljö med större begränsningar på resurser som till exempel en iPAQ.
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Samiuddin, Asim. "IPAM : a web-based IP/DNS management system /." free to MU campus, to others for purchase, 2004. http://wwwlib.umi.com/cr/mo/fullcit?p1422961.

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32

Beard, LaShandra M. "Imput and tracking of continued education units and qualification data for the Information Professional (IP) community /." Monterey, Calif. : Springfield, Va. : Naval Postgraduate School ; Available from National Technical Information Service, 2004. http://library.nps.navy.mil/uhtbin/hyperion/04Sept%5FBeard.pdf.

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33

Jin, Shanlun. "Implementation and Analyses of the Mobile-IP Protocol : Under Windows." Thesis, KTH, Kommunikationssystem, CoS, 2008. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-91690.

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This report is the result of a masters degree project conducted at the Department of Teleinformatics at the Royal Institute of Technology starting from the autumn 1996. The area investigated is the Mobile Internet Protocol, especially its implementation under Windows NT environment. Network driver writing under Windows NT was practised. Recent development in improving Mobile IP protocol to support micro-mobility have also been investigated.
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34

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.

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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.
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35

Lusilao-Zodi, Guy-Alain. "A Bandwidth Market in an IP Network." Thesis, Stellenbosch : University of Stellenbosch, 2008. http://hdl.handle.net/10019.1/1933.

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Thesis (MSc (Mathematical Sciences. Computer Science))--University of Stellenbosch, 2008.
Consider a path-oriented telecommunications network where calls arrive to each route in a Poisson process. Each call brings on average a fixed number of packets that are offered to route. The packet inter-arrival times and the packet lengths are exponentially distributed. Each route can queue a finite number of packets while one packet is being transmitted. Each accepted packet/call generates an amount of revenue for the route manager. At specified time instants a route manager can acquire additional capacity (“interface capacity”) in order to carry more calls and/or the manager can acquire additional buffer space in order to carry more packets, in which cases the manager earns more revenue; alternatively a route manager can earn additional revenue by selling surplus interface capacity and/or by selling surplus buffer space to other route managers that (possibly temporarily) value it more highly. We present a method for efficiently computing the buying and the selling prices of buffer space. Moreover, we propose a bandwidth reallocation scheme capable of improving the network overall rate of earning revenue at both the call level and the packet level. Our reallocation scheme combines the Erlang price [4] and our proposed buffer space price (M/M/1/K prices) to reallocate interface capacity and buffer space among routes. The proposed scheme uses local rules and decides whether or not to adjust the interface capacity and/or the buffer space. Simulation results show that the reallocation scheme achieves good performance when applied to a fictitious network of 30-nodes and 46-links based on the geography of Europe.
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Danielsson, Fredrick, Gabriel Elias, and Dan Jacobsen. "IP-telefoni : Motiv för införande, och skapande av acceptans." Thesis, Jönköping University, JIBS, Business Informatics, 2005. http://urn.kb.se/resolve?urn=urn:nbn:se:hj:diva-126.

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På några år har användningen av IP-telefoni inom svenska företag tredubblats och vi börjar nu se ett attitydskifte gentemot en teknik som länge dragits med dåligt rykte på grund av kvalitetsbrister.

IP-telefoni är ett resultat av att data- och telefonikommunikation vävs samman och att kapaciteten i nätverken ökar. Detta medför att man genom att införa IP-telefoni kan genomföra kostnadsbesparingar och effektivisering. Men tekniken erbjuder även ökad funktionalitet och därmed även mervärde för organisationer som använder IP-telefoni. Inser organisationer detta och utnyttjar man fördelarna med tekniken för att skapa acceptans inom organisationen?

Syftet är att undersöka vilka faktorer som motiverar en övergång från kretskopplad telefoni till IP-telefoni, och hur man inom en organisation skapar acceptans för den nya tekniken.

Syftet har uppfyllts genom en kvalitativ studie som byggt på semi-standardiserade intervjuer. Studien genomfördes hos fyra svenska organisationer som implementerat en ”ren” IP-telefonilösning med IP-växel och hårdvarutelefoner.

Resultatet av studien visar att det är vanligt att utnyttja en större händelse som startskott för implementationen. Detta kan röra sig om en flytt eller en renovering av organisationens lokaler. De främsta orsakerna till att man väljer IP-telefoni har visat sig vara kostnadsbesparingar, effektivisering av nätverksinfrastrukturen samt att man ser tekniken som en investering för framtiden. Det har visat sig att de företag som genomfört en grundlig studie av IP-telefoni som teknik, inledningsvis har identifierat fler fördelar med tekniken.

Vidare har studien även visat att organisationer upplever ett initialt motstånd till förändringen från anställda och att detta motstånd ofta uppkommer i anknytning till tekniska problem som uppstår i samband med implementationen.

Det är mycket ovanligt att organisationer har en speciell fas för att skapa acceptans för förändringen, vilket troligtvis beror på att man ser användandet som liknande med traditionell PSTN-telefoni och att resurser istället läggs på att lösa tekniska problem.


Over the last few years there has been a threefold increase in the use of IP telephony within Swedish companies. We are now beginning to see a change in attitude towards a technology that thanks to poor Quality of Service has been struggling with a bad reputation. IP telephony is a result of computer and telephone networks being con-verged, and by doing this, costs can be saved and efficiency boosted. But the technology does also offer increased functionality and thereby added value to the organisations using it. But do organisations realize this, and do they exploit the benefits of the technology in order to create acceptance for it within the organisation?

The purpose is to examine what the motivating factors are for a transition from PSTN telephony to IP telephony, and also to study how acceptance for the new technology is created within the organisation.

The purpose has been fulfilled by conducting a qualitative study based on semi-standardized interviews. The study has been conducted at four Swedish organisations that have implemented a "pure" IP telephony solution including IP gateways and IP telephones.

The result of the study shows that it is common to use a "big event" as a starting block for the implementation. This could be a relocation or the restoration of an office building. The main reason for choosing IP telephony is the fact that convergence cuts costs, boosts efficiency, and that organisations see IP telephony as an investment for the future. We have noticed that organisations that conduct a thorough study of IP telephony have identified more functional advantages with the technology.

Furthermore, the study has shown that organisations experience initial resistance from users, and that this resistance often arises in connection to technical problems that transpire during the implementation.

It is rare for organisations to have special phase for creation of acceptance, which most likely is due to the fact that organisations see the use of IP telephony as similar to the use of traditional PSTN telephony, and that resources instead are being placed on solving technical problems

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37

Parnes, Peter. "An IP-multicast based framework for designing synchronous distributed multi-user applications on the Internet." Doctoral thesis, Luleå tekniska universitet, Datavetenskap, 1999. http://urn.kb.se/resolve?urn=urn:nbn:se:ltu:diva-25971.

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This thesis examines the problems of how to design scalable, robust, and distributed real-time media applications for the Internet. An agent based framework for solving this problem is presented. The framework promotes several important design requirements such as modularity and reusability of software modules, remote management of applications, adaptiveness, robustness, accessibility, symmetry and the most important scalability. The issue of scalability is examined on several levels including data-distribution, different end-user platforms, scalability of real-time media flows, bandwidth adaption and scalability of sessions. The design of the framework is based on the important concept of IP-multicast, which allows for scalable distribution of data on the Internet. How this decision has influenced the design of the framework is examined in the thesis. The design of the framework have been verified by building a reference implementation in Java and building a number of high-level end-user applications for real-time e-meetings, e-learning (net-based learning) and social awareness via the e-corridor.
Godkänd; 1999; 20061004 (ysko)
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Chan, Siu Ping. "End-to-end streaming protocols with QoS control for secure IP multimedia communications /." View abstract or full-text, 2004. http://library.ust.hk/cgi/db/thesis.pl?ELEC%202004%20CHAN.

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Thesis (Ph. D.)--Hong Kong University of Science and Technology, 2004.
Includes bibliographical references (leaves 167-175). Also available in electronic version. Access restricted to campus users.
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Beard, LaShandra M. "Input and tracking of continued education units and qualification data for the Information Professional (IP) community." Thesis, Monterey, California. Naval Postgraduate School, 2004. http://hdl.handle.net/10945/1459.

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Approved for public release; distribution is unlimited
The objective of this research is to provide recommendations and guidelines for building and maintaining a comprehensive Continuing Education Units (CEU) and qualification tracking system specifically for the Continuing Education Units (CEU) and Qualification program. The guidance includes topics ranging from designing, managing, and implementing a tracking program, through post-implementation of the program. This research includes the training needs of all personnel within the IP community, from users to supervisors to executive-level mangers extending to include designated sponsors/mentors and external subject matter experts. The key research focus of this thesis is to examine the risk and cost benefits in automating the training record for the Information Professional community and further discuss interface design issues and considerations to maximize the flexibility and functionality provided by automation.
Lieutenant, United States Navy Reserve
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40

Harder, Benjamin. "Voice over IP 2.0: an analysis of limits and potential of IP2IP telecommunication." reponame:Repositório Institucional do FGV, 2012. http://hdl.handle.net/10438/9712.

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Internet Telephony (VoIP) is changing the telecommunication industry. Oftentimes free, VoIP is becoming more and more popular amongst users. Large software companies have entered the market and heavily invest into it. In 2011, for instance, Microsoft bought Skype for 8.5bn USD. This trend increasingly impacts the incumbent telecommunication operators. They see their main source of revenue – classic telephony – under siege and disappear. The thesis at hand develops a most-likely scenario in order to determine how VoIP is evolving further and it predicts, based on a ten-year forecast, the impact it will have on the players in the telecommunication industry.The paper presents a model combining Rogers’ diffusion and Christensen’s innovation research. The model has the goal of explaining the past evolution of VoIP and to isolate the factors that determine the further diffusion of the innovation. Interviews with industry experts serve to assess how the identified factors are evolving.Two propositions are offered. First, VoIP operators are becoming more important in international, corporate, and mobile telephony. End-to-end VoIP (IP2IP) will exhibit strong growth rates and increasingly cannibalize the telephony revenues of the classic operators. Second, fix-net telephony in SMEs and at home will continue to be dominated by the incumbents. Yet, as prices for telephony fall towards zero also they will implement IP2IP in order to save costs. By 2022, up to 90% of the calls will be IP2IP. The author recommends the incumbents and VoIP operators to proactively face the change, to rethink their business strategies, and to even be open for cooperation.
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41

Andrén, Magnus. "MPEG-4-Compatible Set-Top Box for IP-networks Based on Open Standards : A Systems Study." Thesis, Linköping University, Department of Electrical Engineering, 2003. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-1665.

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The purpose of this thesis is to examine the possibilities of creating a MPEG-4-compatible set-top box for IP-networks based on open standards.

Existing alternatives for transporting MPEG-4 over IP are evaluated and ISMA is found to be an important actor within the area. ISMA is a non-profit corporation formed to provide a forum for the creation of specifications that define an interoperable implementation for streaming rich media over IP-networks.

Two different designs based on ISMA's recommendation are constructed and evaluated. The designs have different levels of complexity and the more complex design is found to be better due to its extended functionality.

During the design process a number of problems related to this kind of set-top box are discovered. It is believed, however, that many of these problems will be solved within the near future.

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42

Tevemark, Jonas. "Intrusion Detection and Prevention in IP Based Mobile Networks." Thesis, Linköping University, Department of Electrical Engineering, 2008. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-12015.

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Ericsson’s Packet Radio Access Network (PRAN) is a network solution for packet transport in mobile networks, which utilizes the Internet Protocol (IP). The IP protocol offers benefits in responsiveness and performance adaptation to data bursts when compared to Asynchronous Transfer Mode (ATM), which is still often used. There are many manufacturers / operators providing IP services, which reduce costs. The IP’s use on the Internet brings greater end-user knowledge, wider user community and more programs designed for use in IP environments. Because of this, the spectrum of possible attacks against PRAN broadens. This thesis provides information on what protection an Intrusion Prevention System (IPS) can add to the current PRAN solution.

A risk analysis is performed to identify assets in and threats against PRAN, and to discover attacks that can be mitigated by the use of an IPS. Information regarding placement of an IPS in the PRAN network is given and tests of a candidate system are performed. IPS features in hardware currently used by Ericsson as well as missing features are pinpointed . Finally, requirements for an IPS intended for use in PRAN are concluded.

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43

Guitérrez, Barquín Jesús Miguel. "Network Independent Quality of Service : The role of Authentication, Authorization, and Accouting in a roaming environment." Thesis, KTH, Kommunikationssystem, CoS, 2006. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-92253.

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With the increasing shift to the Internet Protocol [3] for all networks and the desire of telecommunications service providers to offer new value to their customers, the need exists to coordinate the delivery of end-to-end quality of service so that providers may offer new services to support their customer’s applications. The key objective of the EuQoS project is to research, integrate, test, validate, and demonstrate end-to-end QoS technologies to support advanced QoS-aware applications over multiple, heterogeneous research, scientific, and industrial network domains. End-to-end quality of service support for multiple applications is a great added value and could become the next major growth spurt in the telecommunications industry. A crucial requirement for the QoS model is that it must not add significant complexity to the existing mechanisms and must be compatible with legacy applications and equipment. Proxy signaling handlers will be used to satisfy the latter constraint. This thesis analyzes possible roaming scenarios and how Authentication, Authorization, and Accounting should be tackled. I seek to provide reasonable solutions and to consider the current environment, always trying to re-use, when possible, the existing architecture and components.
De huvudsakliga målen med EuQoS projektet är att integrera, testa, utvärdera och demonstrera QoS från en ende av kommunikationen till en annan för att stödja avancerade QoS tillämpningar inom multipla nätverksdomäner industrioch forskningsverksamhet. Denna nya teknik är av stort värde och kan komma att bli nästa stora steg i utvecklingen av telekommunikation. Användandet av IPnätverk ökar och i samma takt ökar även efterfrågan av nya och bättre kommunikationstjänster. Därför finns det ett stort behov av QoS hos Internetleverantörerna som ständigt måste tillgodose kundernas önskemål. Det är viktigt att QoS modellen inte komplicerar den redan existerande tekniken. Därför måste den vara kompatibel med befintliga tekniker och utrustning. För att uppfylla dessa krav kommer Proxy signalbehandlare att användas. Den här rapporten behandlar möjliga roaming scenarier och hur AAA bör hanteras. Jag har som mål att presentera rimliga lösningar som tar hänsyn till miljön. För detta används i största möjliga mån redan existerande infrastruktur.
Con la continua evolución de las aplicaciones basadas en el protocolo IP en todas las redes, y el deseo de los proveedores de servicios de telecomunicaciones de ofrecer un valor añadido a sus clientes, cohabita la necesidad de coordinar la entrega de calidad de servicio (QoS) extremo a extremo. De esta forma los proveedores pueden aumentar la oferta de servicios mediante nuevas aplicaciones. El principal objetivo del proyecto EuQoS es investigar, desarrollar, integrar y probar una tecnología independiente de la red de acceso que garantice QoS extremo a extremo. El sistema esta pensado inicialmente para dar soporte a las aplicaciones: VoIP, VoD, video conferencia, y a una aplicación médica llamada MEDIGRAF, sobre múltiples y heterogéneas redes de acceso. Los parámetros que EuQoS tiene en cuenta para la reserva de la calidad de servicio son el ancho de banda, el retardo, la variación del retardo (jitter), y las pérdidas permitidas. Un requisito fundamental para el modelo de QoS es que debe añadir la mínima complejidad posible al existe funcionamiento del sistema y debe ser compatible con el legado de aplicaciones y equipo. Esto se solucionará mediante el uso de señalización a nivel de Proxy. Este proyecto analiza los posibles escenarios de roaming y cómo se debería afrontar la Autenticación, Autorización, y Accounting (AAA) en estas condiciones de itinerancia. En los capítulos iniciales hacen una descripción general del sistema EuQoS, para tener una visión global del proceso de reserva de recursos. Es necesario conocer la estructura completa para lograr una integración mayor y con el menor coste posible. El proyecto EuQoS propone y desarrolla un nuevo mecanismo de QoS que se construye sobre un estado del arte que incorpora los siguientes mecanismos: Monitorización y Medición, Control de Admisión, Gestión de Fallos, Señalización y Negociación de Servicio, Seguridad y AAA, Charging, Ingeniería de Tráfico y Optimización de Recursos. Con el fin de conocer lo que anteriormente otros habían hecho en este campo, antes de escribir una sola línea de este proyecto, llevé a cabo una extensa búsqueda de documentación. Parte de la información utilizada en este documento ha sido extraída de las entregas públicas del proyecto EuQoS hechas a la Comisión Europea. Además de la bibliografía mostrada en las referencias, Ericsson tiene sus propios informes técnicos e implementaciones de protocolos como el protocolo de iniciación de sesión (SIP) y DIAMETER, que se han consultado en varias ocasiones y han contribuido a lo largo de la investigación. Existe una enorme similitud entre la arquitectura del sistema y la de IMS. De esta manera, algunos de los conceptos aplicados a la hora de desarrollar una solución para el caso de roaming para EuQoS se basan en los flujos de señalización utilizados en IMS y en los anteproyectos de nuevos RFCs.
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44

Бічук, І. С. "Сервіси голосових дзвінків для потреб міжособистісного спілкування та бізнесу." Master's thesis, Сумський державний університет, 2018. http://essuir.sumdu.edu.ua/handle/123456789/72163.

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Проведено аналіз існуючих програмних рішень (мобільних додатків) для здійснення голосових дзвінків. Основою виконання став аналіз даних створення програмного забезпечення для додатку для виконання голосових дзвінків у мережі інтернет. Розроблений алгоритм реалізовано у формі програмного забезпечення, створеного за допомогою інструментального програмного середовища Android Studio. Створено модель застосування VoIP в сфері бізнесу в декількох офісах географічно розподіленої компанії за допомогою симулятора Cisco Packet Tracer.
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45

Graham, Derek. "The impact of soft errors in logic and its commercialisation in ARM IP." Thesis, University of Glasgow, 2011. http://theses.gla.ac.uk/2663/.

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The significance of soft errors in logic has grown because of reduced memory vulnerability and the shrinking dimensions of semiconductor technology coupled with the increasing amount of logic integrated into a chip. Consequently, some of ARM’s customers are concerned about how soft errors on the bus interconnect will affect the dependability of their systems, since the interconnect is a critical hub of communication in a SoC and represents a substantial and growing amount of logic. With the rising complexity of their systems, the interconnect will become larger and more complex in the future, adding to their concern. In this work the impact of soft errors on the bus interconnect logic was investigated and a product was developed to ameliorate the effects of such errors on ARM’s customers’ products. Methods to measure the SER of ARM IP were investigated by focusing on logical masking, which is a component in the calculation of the SER. The effect that the topology of a combinatorial logic circuit has on its logical masking rate was considered by performing gate-level statistical fault injection on different implementations of adder circuits. Significant variation in logical masking was found ranging from a factor of 3.1 at a synthesis frequency of 100 MHz to a factor of 2.1 at 900 MHz. This difference is explained in an original way by correlating logical masking with the circuit’s path length and fan-out. These properties could be used to create a static method of measuring the logical masking rather than the current time-consuming method of dynamic simulation. Additionally, nearly 30% of faults injected cause more than one error, which means that the combinational SER will be underestimated if research does not take gate fan-out into consideration. Using this methodology a circuit designer can now base his choice or development of a circuit on its reliability as well as its performance, power, and area. Studying the variation in the factors that affect the SER is important to ensure accuracy in addressing customer requirements. Although it is important to consider the rate of soft error occurrence, in this work the impact of errors is demonstrated to be critical. Using protocol-level fault injection it is shown that faults on the ARM AXI bus interconnect can have a serious effect on the reliability of the entire SoC such as deadlock, memory corruption, or undefined behaviour. Using a fault-path traversal algorithm, it is demonstrated that traditional error detection codes are not sufficient at preventing these failures when faults occur on certain AXI bus signals. This led to the development of novel fault tolerant methods that provide protection for these identified signals. Based on these developments, a product was proposed for an add-on to the AXI bus interconnect that can detect, correct, and report logic soft errors without changing the AMBA standard or the customer’s connecting IP.
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Vuza, Xolisa. "Social and technical issues of IP-based multi-modal semi-synchronous communication: rural telehealth communication in South Africa." Thesis, University of the Western Cape, 2005. http://etd.uwc.ac.za/index.php?module=etd&amp.

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Most rural areas of developing countries are faced with problems like shortage of doctors in hospitals, illiteracy and poor power supply. Because of these issues, Information and Communication Technology (ICT) is often sees as a useful solution for these areas. Unfortunately, the social environment is often ignored. This leads to inappropriate systems being developed for these areas. The aims of this thesis were firstly, to learn how a communication system can be built for a rural telehealth environment in a developing country, secondly to learn how users can be supported to use such a system.
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Соколов, Сергій Вікторович, Сергей Викторович Соколов, Serhii Viktorovych Sokolov, and В. І. Тітарев. "Дослідження і модернізація корпоративних систем ip-телефонії на базі телекомунікаційної мережі." Thesis, Сумський державний університет, 2014. http://essuir.sumdu.edu.ua/handle/123456789/39249.

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Телефонний зв'язок був і залишається критично важливою частиною інфраструктури будь-якого підприємства. Такі можливості як організація конференцій, відеотелефонія, уніфікований обмін повідомленнями, інтерактивні голосові меню, інформаційні сервіси на телефонах і багато інші перетворюють мережу зв'язку в потужний інструмент підвищення продуктивності праці на підприємстві.
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48

Kern, Andreas [Verfasser], and Jürgen [Akademischer Betreuer] Teich. "Ethernet and IP for Automotive E/E-Architectures - Technology Analysis, Migration Concepts and Infrastructure / Andreas Kern. Gutachter: Jürgen Teich." Erlangen : Friedrich-Alexander-Universität Erlangen-Nürnberg (FAU), 2013. http://d-nb.info/1065379137/34.

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49

Dzaferagic, Samir. "Secure Session Mobility for VoIP." Thesis, KTH, Kommunikationssystem, CoS, 2008. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-91676.

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Abstract:
High data rate wireless packet data networks have made real-time IP based services available through mobile devices. At the same time, differences in the characteristics of radio technologies (802.11/WiFi and 3G networks) make seamless handoff across heterogeneous wireless networks difficult. Despite this, many believe that the ultimate goal of next generation networks (often referred to as the fourth generation) is to allow convergence of such dissimilar heterogeneous networks. Supporting voice over Internet Protocol in next-generation wireless systems is thought by some to require support for mobility and quality of service features. Currently a mobile node can experience interruptions or even sporadic disconnections of an on going real-time session due to handovers between both networks of different types and networks of the same type. Many tests have already been done in this area and one may wonder why it is worth spending even more time investigating it? This thesis focuses on the important problem of providing session security despite handovers between networks (be they operated by the same operator or different operators and be they the same link technologies or different). One of the goals in this thesis is to investigate how an ongoing speech session can continue despite a change in transmission media1. Additionally, a number of security threats that could occur due to the handover will be identified and presented. Finally, the most suitable solution to address these threats will be tested in a real environment. Eventual shortcomings and weaknesses will be identified and presented; along with suggestions for future work.  1 When utilizing IP over carriers such as wired Ethernet, WLAN, and 3G.
Trådlösa hög-hastighets datanät har möjliggjort appliceringen av realtids tjänster på mobil utrustning över IP. Samtidigt har skillnaderna i de olika radioteknologierna (802.11/WiFi och 3G näten) introducerat nya problem med att upprätthålla trådlösa kommunikationen tvärs den heterogena trådlösa accessen. Många tror att slutmålet för nästa generations nätverk (ofta refererade som fjärde generationens nätverk) är att tillåta konvergensen av dessa olika heterogena nätverk. Stödet för Voice over Internet Protokollet (VoIP) i nästa generations trådlösa nät tror somliga kräver ett inslag av kombination mellan mobilitet samt upprätthållandet av kvaliteten. För närvarande kan den mobila noden (MN) råka ut för störningar och även sporadiska avbrott av en pågående realtidssessionen på grund av övergångar mellan samma eller olika typer av medier. Många tester har redan gjorts inom det här området och man kan fråga sig varför det är värt att lägga ner ännu mer tid på att undersöka det här? Det här examensarbetet fokuserar på det viktiga problemet som handlar om att kunna erbjuda sessions säkerhet trots övergångar mellan näten (oavsett om dessa drivs av samma eller olika operatörer samt oavsett om de är av samma eller olika nätverks typ). Ett av målen för det här examensarbetet är att undersöka hur en pågående talsession behålls vid byte av transmissionsmedia2. Vidare kommer olika säkerhetsaspekter och hot som kan tänkas uppstå vid bytet att identifieras och presenteras. Slutligen kommer den mest lämpade lösningen till problemet att testas i verklig miljö. Eventuella brister och svagheter kommer att identifieras och redovisas i slutet av rapporten tillsammans med förslag på framtida arbete. 2 Då man nyttjar IP bärare som trådbundet Ethernet, WLAN och 3G.
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50

Бураківська, А. О., and А. В. Омельченко. "Аналіз методів забезпечення якості послуг мультисервісних мереж." Thesis, ХНУРЕ, 2021. https://openarchive.nure.ua/handle/document/16487.

Full text
Abstract:
The concept of Next Generation Network NGN (Next Generation Network), which uses TCP / IP technology as a basis for building multiservice networks, gives the operator great opportunities to organize a virtually unlimited number of services. But at the same time it sets new challenges in terms of creating and implementing new methods of traffic service. The struggle for resources affects the quality of service for all types of traffic, including IPTV traffic. The modern model of differentiated service provides for the division of traffic into classes with giving them different priorities in service. The problem of establishing priorities for different types of traffic was solved by the method of reducing individual quality indicators to a generalized value function of the additive type using weights.
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