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Novel type aquaporin SIPs are mainly localized to the ER membrane andIshikawa, Fumiyoshi, Suga, Shinobu, Uemura, Tomohiro, Sato, Masa, H., Maeshima, Masayoshi, 前島, 正義 January 2005 (has links)
No description available.
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Auswahl, Test und Anpassung eines SIP-ClientDonner, Sandra, January 2004 (has links)
Chemnitz, Techn. Univ., Studienarb., 2004.
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Implementeringen av samordnad individuell plan : Socialsekreterares upplevelser av hinder och möjligheter / The implementation of coordinated individual plan : Child-welfare workers experiences of obstacles and possibilitiesJakobsson, Elin, von Post, Patrik January 2015 (has links)
Studiens syfte är att undersöka vilka möjligheter och utmaningar som implementeringen av samverkansverktyget samordnad individuell plan (SIP) kan medföra för socialsekreterare inom individ- och familjeomsorgen barn och unga. Detta har gjorts genom att sju socialsekreterare från tre olika kommuner har intervjuats. Genom att använda en kvalitativ innehållsanalys på datamaterialet har fem olika kategorier skapats där det framgår att socialsekreterarna anser att de har tillräcklig kunskap om samordnad individuell plan trots att de fått en mycket begränsad utbildning. Dock uppger flera socialsekreterare att de skulle kunnat göra fler samordnade individuella planer än de har gjort, vilket förklaras med stress samt i vissa fall osäkerhet. Samtliga socialsekreterare är positivt inställda till samordnad individuell plan men de upplever ändå vissa hinder för att samverkan ska fungera optimalt. Bland annat upplever många att de som representanter får ta ett större ansvar i arbetet med samordnad individuell plan än andra samverkansaktörer. Det framkommer också att socialsekreterarna ser samordnad individuell plan som en möjlighet att dela och inhämta information samt att det förenklar planeringen i ett ärende. Även om arbetet med samordnad individuell plan läggs till socialsekreterarnas övriga arbetsuppgifter kunde samtliga se hur det faktiskt kunde effektivisera deras arbete. För att tydligare förklara resultaten från undersökningen kopplas dessa ihop med tidigare forskning om samverkan och implementering.
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Open Source SIP Application Servers For IMS Applications: A SurveyKumlin, Veronica January 2007 (has links)
I ett IMS-nät (IP Multimedia Subsystem) finns Applikations Servrar (AS) som tillhandahåller tjänster, dessa AS är ofta SIP-servrar (Session Initation Protocol). Målet för det här examensarbetet var att finna den SIP-server som är mest lämpad att använda när man ska lägga till en tjänst i ett IMS-nät. Först gjordes en undersökning av tillgängliga SIP AS och ett tiotal hittades. Några av dem valdes ut för att jämföras och utredas mer noggrant. Inledningsvis inspekterades deras dokumentation, sedan analyserades proceduren att installera AS och att lägga till en tjänst på dem. Till sist lades tjänsten till i ett IMS-nät. Eftersom varken deras fördelar eller nackdelar utmärkte sig på ett sådant sett att någon av dem sammantaget var bättre än den andra, kunde tyvärr inte någon av dem pekas ut som den mest lämpade. / In an IMS (IP Multimedia Subsystem) network there are Application Servers (ASs) which host and execute services. These are often SIP (Session Initiation Protocol) servers. The goal for this thesis was to evaluate a selection of open source SIP ASs to see which one was most suited for IMS service development. Ten different SIP ASs was found in an initial search. They were briefly reviewed and some of them were chosen as candidates for further investigation. The documentation of the chosen servers were thoroughly explored and evaluated; as was the installation process and how to deploy a service to them. Finally the servers were installed and a service was implemented and deployed to an IMS network. Unfortunately, I could not point out one of the ASs to be more suited than another for deploying an IMS service on. This was because they had different pros and cons and none of them were distinguishing in a way that made them superior to the other.
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Towards Secure SIP Signalling Service for VoIP applications : Performance-related Attacks and PreventionsZhang, Ge January 2009 (has links)
Current Voice over IP (VoIP) services are regarded less secure than the traditional public switched telephone network (PSTN). This is due to the fact that VoIP services are frequently deployed in an relatively open environment so that VoIP infrastructures can be easily accessed by potential attackers. Furthermore, current VoIP services heavily rely on other public Internet infrastructures shared with other applications. Thus, the vulnerabilities of these Internet infrastructures can affect VoIP applications as well. Nevertheless, deployed in a closed environment with independent protocols, PSTN has never faced similar risks. The main goal of this licentiate thesis is the discussion of security issues of the Session Initiation Protocol (SIP), which serves as a signalling protocol for VoIP services. This work especially concentrates on the security risks of SIP related to performance. These risks can be exploited by attackers in two ways: either actively or passively. The throughput of a SIP proxy can be actively manipulated by attackers to reduce the availability of services. It is defined as Denial of Service (DoS) attacks. On the other hand, attackers can also profile confidential information of services (e.g., calling history) by passively observing the performance of a SIP proxy. It is defined as a timing attack. In this thesis, we carefully studied four concrete vulnerabilities existing in current SIP services, among which, three of them can lead to DoS attacks and one can be exploited for timing attacks. The results of our experiments demonstrate that these attacks can be launched easily in the real applications. Moreover, this thesis discusses different countermeasure solutions for the attacks respectively. The defending solutions have all in common that they are influencing the performance, by either enhancing the performance of the victim during a DoS attack, or abating the performance to obscure the time characteristic for a timing attack. Finally, we carefully evaluated these solutions with theoretical analyses and concrete experiments.
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Surface-mountable LTCC-SiP module approach for reliable RF and millimetre-wave packagingKangasvieri, T. (Tero) 11 November 2008 (has links)
Abstract
The rapid growth in the wireless communications markets together with the emerging
need for high-bit-rate (≫ 100 Mb/s) multimedia/data services are pushing the usage of radio spectrum resources below 10 GHz to the uttermost limit. The lack of bandwidth has led to an extensive development of mobile/fixed BWA systems for the higher microwave and millimetre-wave regions up to the V-band frequencies (50–75 GHz). In order for these systems to have mass deployment and to meet cost-sensitive consumer markets’ requirements, their cost and size must be reduced from current levels. One of the most viable packaging approaches to satisfy these demands is the low-temperature co-fired ceramic (LTCC) based system-in-package (SiP) module technology combined with fully automated surface-mount assembly techniques. However, one of the main challenges of this approach has been previously associated with the broadband radio frequency (RF) and reliability performance constraints of the board-level solder joints in LTCC/PCB assemblies. In this thesis the primary focus is to tackle these limitations and significantly extend the feasibility of the LTCC module technology to various wireless/mixed-signal packaging applications.
The thesis is divided into three main parts. In the first part, design, modelling and implementation of vertical package transitions (BGA, flip-chip, substrate via) over a very wide frequency range are presented. In the second part, the emphasis is on the improvement of the thermal fatigue endurance of the board-level solder joints in the LTCC/PCB assemblies. In the last part, the results are merged to realize a high-performance LTCC module platform for use in a wide variety of SiP products in the telecommunication sector.
The flip-chip, substrate-via and BGA transition structures exhibited excellent signal transmission properties up to the V-band frequencies. The developed equivalent circuit models of the transitions matched well with the measurements. Cascading the transitions together allows the building of different combinations of vertical interconnection paths in SiP modules. To demonstrate this, a surface-mountable LTCC filter package for K-band radio link frequencies was implemented.
The developed composite BGA solder joint structure with plastic-core solder balls significantly enhanced the thermal fatigue life in LTCC/PCB assemblies in different thermal cycling conditions, indicating adequate board-level reliability for many practical LTCC-BGA packaging applications. Moreover, electromagnetic analysis showed that the use of the plastic-core solder ball has no detrimental impact on the RF performance of the solder joint.
Finally, based on the obtained results a reliable lead-free LTCC-BGA module platform was developed for broadband packaging applications. The BGA module platform with a size of 15 mm × 15 mm included 38 low-frequency and two wideband RF input/output connections up to the K-band frequencies. The module structure also allowed plenty of space to mount discrete SMD/bare-die components on the surface and/or to embed passive components in the 1.2 mm thick substrate. Preliminary thermal cycling results of the soldered LTCC/PCB assemblies demonstrated sufficient reliability for telecommunication applications. Therefore, the presented module platform can serve as a physical building block for various wireless/mixed-signal SiP products, and hence significantly reduce their development time and associated costs.
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Správa a konfigurace VoIP ústředny Asterisk / Management and configuration of Asterisk VoIP exchangeBinder, Tomáš January 2008 (has links)
This diploma dissertation is dealing with the VoIP software exchange Asterisk. In the dissertation there are described its abilities and possible ways of its configuration. Special attention is given to the signalling protocol SIP, which is described in one of the chapters. Within this dissertation a dial plan, which demonstrates the technique of dial plan creating, was created. Within the boundaries of the dialplan following services could be found: a voicemail, conference, Interactive Voice Response and call queues. Configuration files, with the help of which the exchange is configurated, are described in my dissertation as well. Finally, three laboratory assignments for purposes of the subject Multimedia Services are mentioned. Their main aim is to familiarise students with the creation of SIP accounts in the exchange, their mutual connections, defining the Interactive Voice Response and forming a new call centre.
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IP Multimedia Subsystem a jeho využití pro zajištění kvality služeb / IP Multimedia Subsystem and its usage for quality-of-services assuranceBetlach, Michal January 2008 (has links)
The aim of this work is to give information about the structure and elements of the IP Multimedia Subsystem, using signaling protocol and transferring messages between the terminal and elements of IMS. Create model, which implements messages transfer between terminal and IMS elements, by environment tools OPNET Modeler. This work describes system function IMS, its architecture, basic elements, architecture of using signaling protocol SIP and its elements, SIP and SDP messages format and the way of ensuring Quality of Services. There is a project of model terminal registration and reservation data flow by terminal in the IMS. These models were realized in a simulation environment called OPNET Modeler by using tools: project editor, node editor and process editor. In the last part of this work, the method of verification correct function of realized models was created by simulation tools called Debbuger. This tool is one of the environment tools OPNET Modeler.
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Vývoj SIP klienta v JavaME / SIP client development in JavaMEJaneček, Aleš January 2008 (has links)
This thesis analyzes various alternatives of SIP protocol implementation on Java Micro Edition devices. This analysis is especially concentrated on SIP protocol, SIP API (JSR-180) and the development in contemporary mobile data networks. I design a midlet for a mobile phone SIP client on the basis of this analysis. The design is especially concentrated on communication, registration and text mesage transfer.
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Princip hlasové komunikace v IP sítích a její bezpečnost / Voice-over-IP principle and security problemsBořuta, Petr January 2008 (has links)
This master’s thesis deals with security properties of protocols used for VoIP systems. In the first part, there is a description of most commonly used protocols and structure of VoIP systems. This part also discuss signaling and transport protocols. The second part of this paper describes techniques of ensuring quality of services. The next part presents SIP messages and communication. Last part of this paper overviews security risks of VoIP protocol. Practical part of this thesis describes creation of a testing VoIP network, on which several attacks has been made, fallowed by securing of mentioned VoIP network. Result of this thesis is evaluation of security risks connected to VoIP communication.
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