To see the other types of publications on this topic, follow the link: Protocollo SPI.
Dissertations / Theses on the topic 'Protocollo SPI'
Create a spot-on reference in APA, MLA, Chicago, Harvard, and other styles
Consult the top 50 dissertations / theses for your research on the topic 'Protocollo SPI.'
Next to every source in the list of references, there is an 'Add to bibliography' button. Press on it, and we will generate automatically the bibliographic reference to the chosen work in the citation style you need: APA, MLA, Harvard, Chicago, Vancouver, etc.
You can also download the full text of the academic publication as pdf and read online its abstract whenever available in the metadata.
Browse dissertations / theses on a wide variety of disciplines and organise your bibliography correctly.
1
Pastore, Cesare. "Progetto software/firmware di un’interfaccia per acquisizione dati da un nodo sensore basato su microcontrollore." Bachelor's thesis, Alma Mater Studiorum - Università di Bologna, 2017. http://amslaurea.unibo.it/12957/.
Full textAbstract:
L'elaborato descrive le modalità di progettazione e programmazione di un firmware per il microcontrollore PIC16F1823 che implementa lo scambio di dati tra un nodo sensore e un personal computer, tramite i protocolli UART e SPI.
2
Costa, Daniel Gouveia. "Uma arquitetura baseada em SCTP e SIP para suporte a aplica??es VoIP m?veis e a especifica??o formal do seu m?dulo de controle." Universidade Federal do Rio Grande do Norte, 2006. http://repositorio.ufrn.br:8080/jspui/handle/123456789/15461.
Full textAbstract:
Made available in DSpace on 2014-12-17T14:56:09Z (GMT). No. of bitstreams: 1
DanielGC.pdf: 538651 bytes, checksum: 34bfc134a2af9166b846b044a2968b16 (MD5)
Previous issue date: 2006-05-25New versions of SCTP protocol allow the implementation of handover procedures in the transport layer, as well as the supply of a partially reliable communication service. A communication architecture is proposed herein, integrating SCTP with the session initiation protocol, SIP, besides additional protocols. This architecture is intended to handle voice applications over IP networks with mobility requirements. User localization procedures are specified in the application layer as well, using SIP, as an alternative mean to the mechanisms used by traditional protocols, that support mobility in the network layer. The SDL formal specification language is used to specify the operation of a control module, which coordinates the operation of the system component protocols. This formal specification is intended to prevent ambiguities and inconsistencies in the definition of this module, assisting in the correct implementation of the elements of this architecture
Novas vers?es do protocolo SCTP permitem sua utiliza??o para implementa??o de mecanismos de handover em n?vel de transporte, bem como o fornecimento de um servi?o de transmiss?o de dados parcialmente confi?vel. Integrando o SCTP com o protocolo de inicia??o de sess?es, SIP, al?m de utilizar adicionalmente servi?os de outros protocolos auxiliares, uma arquitetura de comunica??o p?de ser proposta, a fim de atender ?s aplica??es de voz sobre IP com requisitos de mobilidade. S?o especificados ainda os procedimentos de localiza??o de usu?rio em n?vel de aplica??o, utilizando o protocolo SIP, como alternativa aos mecanismos empregados por protocolos tradicionais que suportam mobilidade na camada de rede. A linguagem de especifica??o formal SDL ? utilizada para especificar o funcionamento de um M?dulo de Controle, relacionado ? opera??o coordenada dos protocolos que comp?e a arquitetura. Pretende-se assim evitar ambig?idades e inconsist?ncias na defini??o desse m?dulo, o que pode auxiliar em implementa??es corretas de elementos dessa arquitetura
3
El, Sawda Samer. "Contribution à l'amélioration de la sécurité pour le protocole SIP (Session Initiation Protocol)." Paris 6, 2011. http://www.theses.fr/2011PA066019.
Full textAbstract:
SIP est un protocole de signalisation de la couche application pour créer, modifier et terminer des sessions, pour les appels téléphoniques sur Internet et la diffusion et les conférences multimédia. Nous proposons dans cette thèse l’extension « SIP SIGN » permettant de signer quelques champs d’entête pour fournir un service de non-répudiation et une authentification de bout-en-bout tout en utilisant le protocole TLS ou DTLS. Nous proposons aussi «SIP SecLite » comme une solution complète, simple, efficace et rapide pour sécuriser les communications SIP en fournissant les propriétés de sécurité exigées par SIP. Notre solution permet de réaliser la sécurité de bout en bout d’un réseau SIP tout en assurant la confidentialité et l'intégrité des messages SIP, la non-répudiation de la communication, l’anonymat par rapport aux serveurs intermédiaires et la protection d’identité et des données privées des utilisateurs finaux.
4
Raappana, M. (Markku). "Multipurpose synthesizable SystemVerilog Spi-Bus protocol verification system." Master's thesis, University of Oulu, 2017. http://urn.fi/URN:NBN:fi:oulu-201702231190.
Full textAbstract:
The complexity of System-on-a-Chip (SoC) is continuing to increase due to the shrinking die size, increase in the number of sub-modules, power efficiency, performance, higher functionality and used protocols. This has an impact on the verification process related to the overall design process. For the verification process, there are commercial products that can be applied in order to verify and test certain Intellectual Properties (IP) but also platforms that lack these tools. This thesis focuses on the issue where a system has to be constructed that helps the verification and testing process of a data bus protocol used by the Device Under Testing (DUT). The study of the Serial Peripheral Interface (SPI) gives the examples of some issues that can be faced during applying this data bus protocol to any given system. Different kinds of testing and verifying methods are addressed in order to show what the tools can be when applying new DUT to a system or examining the data bus protocol it uses. The flow of a design process is studied by showing the iterations of a particular system that was to be created to meet the need that were introduced while examining the issues relating this subject. This flow can be said to start from ground level and end to the final iteration where the system could be created from. The functionality and structure of a Multipurpose Verification System that was created during this thesis are explained. The proofing process of this system is showed by examining the simulation and synthesis reports. The outcomes and future development ideas are discussed as well. This thesis showed that the study in hand has benefits to Nokia as the applying company and the system could be added to the company tool library after modifying it to be used as a stand-alone IPJärjestelmäpiirien (SOC) kompleksisuus on jatkuvassa kasvussa johtuen johdinpiirien pienenemisestä, alijärjestelmien määrän kasvusta, vaatimuksista tehonkulutuksessa, suorituskyvyn kasvusta, toiminnallisuuden kasvusta ja käytettävistä protokollista. Näillä attribuuteilla on vaikutusta kokonaissuunnitteluprosessin verifiointiprosessiin. Verifiointiprosessiin on tällä hetkellä saatavilla kaupallisia sovelluksia, joita voidaan hyödyntää kolmannen osapuolen suunnitteleman systeemin testaukseen ja verifioitiin. Samalla on myös olemassa testausalustoja, joista nämä sovellukset puuttuvat. Tämä diplomityö keskittyy tilanteeseen, jossa täytyy rakentaa systeemi joka helpottaa testattavassa laitteessa (DUT) käytettävän tietoväyläprotokollan verifiointi- ja testausprosessia. Serial Peripheral Interface (SPI)-väyläprotokollan tutkimus tuo esiin esimerkkejä, joihin voidaan törmätä, kun kyseistä väyläprotokollaa käytetään missä tahansa sitä hyödyntävässä systeemissä. Diplomityössä tutkitaan erilaisia testaus- ja verifiointimetodeja, jotta voidaan osoittaa mitä erilaisia työkaluja voidaan hyödyntää, kun uusi testattava laite lisätään olemassa olevaan systeemiin tai tutkitaan tämän käyttämää väyläprotokollaa. Kokonaissuunnitteluprosessia on tutkittu esittelemällä tietyn järjestelmän iteraatiovaiheita, joka kehitettiin ratkaisemaan aiemmin tarkasteltuja, tähän aiheeseen liittyviä ongelmia. Suunnitteluprosessin voidaan katsoa alkaneen tilanteesta, jossa mitään konkreettista ei ollut vielä valmiina ja päättyvän tilanteeseen jossa järjestelmän viimeinen iteraatio voitiin alkaa konkretisoimaan. Monikäyttöisen syntetisoituvan verifiointijärjestelmän funktionaalisuus ja rakenne esitellään. Tutkimalla simulointi- ja synteesiraportteja näytetään tämän järjestelmän varmennusprosessi. Diplomityön toteutumista ja tulevaisuuden jatkokehitysideoista keskustellaan. Tämä diplomityö osoittaa, että Nokia soveltajayrityksenä pystyy hyödyntämään tämän tutkielman lopputulemia. Lisäksi työn tulokset voidaan lisätä, modifioinnin jälkeen, yrityksen komponenttikirjastoon toimimaan itsenäisenä instanssina
5
Lakay, Elthea Trevolee. "SIP-based content development for wireless mobile devices with delay constraints." Thesis, University of the Western Cape, 2006. http://etd.uwc.ac.za/index.php?module=etd&action=viewtitle&id=gen8Srv25Nme4_9048_1182233050.
Full textAbstract:
SIP is receiving much attention these days and it seems to be the most promising candidate as a signaling protocol for the current and future IP telephony services. Realizing this, there is the obvious need to provide a certain level of quality comparable to the traditional telephone service signalling system. Thus, we identified the major costs of SIP, which were found to be delay and security. This thesis discusses the costs of SIP, the solutions for the major costs, and the development of a low cost SIP application. The literature review of the components used to develop such a service is discussed, the networks in which the SIP is used are outlined, and some SIP applications and services previously designed are discussed. A simulation environment is then designed and implemented for the instant messaging service for wireless devices. This environment simulates the average delay in LAN and WLAN in different scenarios, to analyze in which scenario the system has the lowest costs and delay constraints.
6
Sharma, Neena. "SERIAL PROTOCOL BRIDGE." University of Cincinnati / OhioLINK, 2012. http://rave.ohiolink.edu/etdc/view?acc_num=ucin1352403332.
Full text
7
Camilletti, Luca. "Comunicazione sicura mediante il protocollo SIP; uno studio di fattibilità." Bachelor's thesis, Alma Mater Studiorum - Università di Bologna, 2012. http://amslaurea.unibo.it/3141/.
Full text
8
Mosavat, Vahid. "SIP Extensions for the eXtensible Service Protocol." Thesis, KTH, Mikroelektronik och Informationsteknik, IMIT, 2003. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-93107.
Full textAbstract:
The switched telephony network was designed for voice calls. Expansion of datacommunication has lead to a wide range of experimentation to create new services. Theses services take place outside the network. When adding new services we currently encounter problems due to limitations of the simple devices at end points. Theo Kanter has proposed a new model to remove these limitations; this model is called “Adaptive Personal Mobile Communication”. The model consists of several components in the application layer of ISO standard. This model is based on peer to peer connections and the purpose of this model is to move services from within the networks to end point devices and avoid using central servers within the network. The Session Initiation Protocol (SIP) for establishing multimedia sessions allows us to move the point of integration for multimedia service integration out to the end-points. This project concerns the implementing of a prototype of this model as an SIP extension along with it evaluation. SIP offer addressing, naming, and localization of resources in this project. This report presents different design alternatives for XSP as an SIP extension, and the chosen model presents as a result of comparing of these design alternatives.
9
Evloguieva, Evelina. "Light-weight SIP protocol for internet telephony services." Thesis, McGill University, 1999. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=30117.
Full textAbstract:
The technology that governs the Telecommunications today is based on the Intelligent Networks (IN). But we can see that the Internet Telephony is emerging rapidly and that it has good chances to become the basis for the next generation telecommunication networks. There are two major competing standards for Internet Telephony - H.323 protocol stack and SIP. This thesis focuses on SIP and the Value Added Services in the SIP based Internet Telephony. It describes and analyzes two existing SIP based approaches to the VAS implementation and presents new hybrid SIP-IN approach based on the concept of reusing the existing IN nodes. The major part of the study is devoted to the design of a lightweight protocol, built as SIP extension, providing for VAS in the hybrid SIP-IN environment. To illustrate the hybrid SIP-IN approach to VAS implementation and the SIPext protocol operation the execution of the Freephone and the Call Distribution services is described. Finally the functional modules supporting the SIPext communication in a hybrid SIP-IN architecture are outlined.
10
Evloguieva, Evelina. "Light-weight SIP protocol for Internet telephony services." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 2000. http://www.collectionscanada.ca/obj/s4/f2/dsk1/tape3/PQDD_0020/MQ55052.pdf.
Full text
11
Wu, YanHao. "SIP-based location service provision." Thesis, University of the Western Cape, 2005. http://etd.uwc.ac.za/index.php?module=etd&.
Full textAbstract:
Location-based service (LBS) is a geographical location-related service that provides highly personalized services for users. It is a platform for network operators to provide new and innovative ways of increasing profits from new services. With the rapidly growing trend toward LBS, there is a need for standard LBS protocols. This thesis started with introducing the Internet Engineering Task Force GEOPRIV working group, which endeavors to provide standard LBS protocols capable of transferring geographic location information for diverse location-aware applications. Through careful observation, it was found that Session Initiation Protocol (SIP) is well suited to the GEOPRIV requirements. The aim of this research was therefore to explore the possibility of the integration of LBS and the SIP protocol and, to some extent fulfill the GEOPRIV requirements.
12
Al-Canaan, Amer. "Création de services de télécommunications en utilisant le protocole SIP et l'API JAIN-SIP." [S.l. : s.n.], 2005.
Find full text
13
Al-Canaan, Amer. "Création de services de télécommunications en utilisant le protocole SIP et l'API JAIN-SIP." Mémoire, Université de Sherbrooke, 2005. http://savoirs.usherbrooke.ca/handle/11143/1301.
Full textAbstract:
L'évolution du domaine des télécommunications et des réseaux d'Internet a abouti à l'intégration et la collaboration de ces deux domaines, ce qui a favorisé des résolutions de plusieurs problèmes de la téléphonie traditionnelle. Afin d'élargir les zones de services au-delà des frontières et des continents, les grandes compagnies de télécommunications et d'Internet ont investi dans le domaine des protocoles de signalisation dont H.323, MGCP (Media Gateway Control Protocol) et SIP. Depuis que l'Internet est devenu répandu, la téléphonie sur l'Internet (ou la téléphonie IP) est devenue populaire et fiable, ce qui a augmenté les demandes de services. SIP a été conçu pour remplacer le protocole H.323, étant plus simple et plus efficace. SIP est fondé sur le protocole HTTP. L'utilisation de SIP permet de concevoir des services de téléphonie IP qui s'intègrent facilement dans les réseaux informatiques. Comparativement à H.323 et MGCP, SIP est plus léger et plus simple. De plus, SIP est horizontal puisqu'il utilise des protocoles IP, ce qui le rend intéressant au niveau de l'implantation et de la création de services traditionnels et même des nouveaux services qui nécessitent de l'interaction avec les applications et les protocoles sur l'Internet. La motivation de ce travail vient de la croissance de la demande des services IP et le besoin de nouvelles méthodes plus simples et fiables s'intégrant aux réseaux de l'Internet afin de répondre à la résolution de la problématique de la création de services de télécommunications. Le premier objectif de ce travail est d'étudier et de proposer des solutions de conception de services dans l'environnement SIP. Le deuxième objectif est de proposer des solutions de réalisation et de mise en oeuvre de ces services.
14
Vincenzi, Selene. "Supporto al multihoming nei protocolli SIP e RTP: uno strumento simulativo." Bachelor's thesis, Alma Mater Studiorum - Università di Bologna, 2010. http://amslaurea.unibo.it/1626/.
Full text
15
Piagnani, Paolo. "Studio ed analisi della sicurezza del protocollo SIP nelle reti Voice over IP." Master's thesis, Alma Mater Studiorum - Università di Bologna, 2013. http://amslaurea.unibo.it/5178/.
Full textAbstract:
Nell'era di Internet e della digitalizzazione, anche la telefonia ha avuto la possibilità di evolversi, e grazie alle tecnologie Voice-over-IP è stato possibile realizzare servizi di comunicazione avanzata su reti di dati. Anche se la comunicazione vocale è l'aspetto chiave di questi sistemi, le reti VoIP supportano altri tipi di servizi, tra cui video, messaggistica istantanea, condivisione di file,
ecc. Il successo di questa nuova tipologia di rete è dovuto ad una migliore flessibilità rispetto ai vecchi sistemi analogici, grazie ad architetture aperte e implementazioni a livello software, e soprattutto ad un minor costo legato alle apparecchiature ed ai collegamenti utilizzati, ed ai nuovi modelli di business e di consumo sempre più orientati allo sfruttamento della connettività a banda larga.
Tuttavia, l'implementazione dei sistemi VoIP rappresenta anche un grado di complessità maggiore in termini di architetture di rete, di protocolli, e di implementazione, e con questo ne segue un incremento delle possibili vulnerabilità. Una falla nella sicurezza in questi sistemi può portare a disservizi e violazione della privacy per gli utenti con conseguenti ripercussioni economiche per i relativi gestori.
La tesi analizza la sicurezza delle reti VoIP concentrandosi sul protocollo che sta alla base dei servizi multimediali, il protocollo SIP. SIP è un protocollo di livello applicativo realizzato per creare, modificare e terminare delle sessioni multimediali tra due o più utenti. Dopo un'introduzione alle generalità del protocollo, vengono esaminate le classi di vulnerabilità delle reti VoIP e gli
attacchi a SIP, e vengono presentate alcune contromisure attuabili. Viene mostrato un esempio di come vengano attuati alcuni dei principali attacchi a SIP tramite l'utilizzo di appositi strumenti. L'eborato conclude con alcune considerazioni sulle minacce al protocollo e sugli obiettivi futuri che la comunità scientifica dovrebbe perseguire.
16
Dzieweczynski, Marcin. "Implementation of Caller Preferences in Session Initiation Protocol (SIP)." Thesis, Linköping University, Department of Electrical Engineering, 2004. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-2238.
Full textAbstract:
Session Initiation Protocol (SIP) arises as a new standard of establishing and releasing connections for vast variety of multimedia applications. The protocol may be used for voice calls, video calls, video conferencing, gaming and many more.
The 3GPP (3rd Generation Partnership Project) suggests SIP as the signalling solution for 3rd generation telephony. Thereby, this purely IP-centric protocol appears as a promising alternative to older signalling systems such as H.323, SS7 or analog signals in PSTN. In contrast to them, SIP does not focus on communication with PSTN network. It is more similar to HTTP than to any of the mentioned protocols.
The main standardisation body behind Session Initiation Protocol is The Internet Engineering Task Force (IETF). The most recent paper published on SIP is RFC 3261 [5]. Moreover, there are working groups within IETF that publish suggestions and extensions to the main standard. One of those extensions is “Caller Preferences for the Session Initiation Protocol (SIP)” [1].
This document describes a set of new rules that allow a caller to express preferences about request handling in servers. They give ability to select which Uniform Resource Identifiers (URI) a request gets routed to, and to specify certain request handling directives in proxies and redirect servers. It does so by defining three new request header fields, Accept-Contact, Reject-Contact, and Request-Disposition, which specify the caller preferences. [1].
The aim of this project is to extend the existing software with caller preferences and evaluate the new functionality.
17
Saporetti, Lorenzo. "Multiplexing di protocolli SIP e RTP su canali virtuali multipercorso: oscuramento pacchetti." Bachelor's thesis, Alma Mater Studiorum - Università di Bologna, 2011. http://amslaurea.unibo.it/2712/.
Full text
18
Rajaram, Vijay Sundar. "Session initiation protocol for wireless channels." Texas A&M University, 2006. http://hdl.handle.net/1969.1/4917.
Full textAbstract:
The Session Initiation Protocol (SIP) was designed for wire line networks. It was
developed to initiate, modify and terminate sessions between two hosts on a network.
When the Internet expanded to include wireless hosts, SIP did not scale well for these
wireless hosts because of the nature of the wireless channel. Also, there were issues
with mobility and real time communication. This thesis proposes improvements to
some of the extensions to SIP, for better performance over wireless channels. We
investigate the call setup time for various transport mechanisms viz. TCP and UDP,
and study the performance of a dynamic Session Timers compared to the current
standard of a periodic refresh mechanism, where the frequency of UPDATEs vary with
the condition of the wireless channel. We also propose a handoff algorithm that
reduces the handover time with decreased packet losses.
19
Angeles, Piña Carlos. "Distribution of Context Information using the Session Initiation Protocol (SIP)." Thesis, KTH, Kommunikationssystem, CoS, 2008. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-91854.
Full textAbstract:
Context-aware applications are applications that exploit knowledge of the situation of the user (i.e. the user’s context) to adapt their behavior, thus helping the user achieve his or her daily tasks. Today, the transfer of context information needs to take place over unreliable and dynamically changing networks. Moreover context information may be produced in different devices connected to different networks. These difficulties have limited the development of context-aware applications. This thesis presents a context distribution method exploiting the event notification mechanisms of the Session Initiation Protocol (SIP), aiming to provide access to context information regardless of where it is produced. The context distribution component presented in this thesis uses SIP for Instant Messaging and Presence Leveraging Extensions (SIMPLE) to enable context sharing by using a SIP presence server, specifically the SIP Express Router (SER) and its presence module. This context distribution component allows distribution of context information in both synchronous and asynchronous mode. The distribution mode depends on the application requirements for context distribution, as well as the nature and characteristics of the contextinformation. In this thesis, based on system scalability, the user’s mobility, and latency - recommendations are given about in which situations each mode is more suitable for distributing context information. The system was evaluated using a load generator. The evaluation revealed that the server is highly scalable. The response time for synchronous retrieval of context information is nearly constant, while in asynchronous mode the time to process a subscription increases with the amount of information in the database regarding previous subscriptions. Notifications are sent at a regular rate (≈2800 notifications per second); however there is a purposely random delay (0 to 1 second), between an update of context information (i.e. receipt of a publish message) and the start of notifications to subscribed users. The requirements of the context-aware applications using the distribution component, such as response time, have to be taken into account when deciding upon the mode of context distribution for each application. This thesis provides some empirical data to help an application developer make this selection.Kontext-medvetna (eng. Context-aware) applikationer är applikationer som utnyttjar information om användarens situation (d.v.s. användarens kontext) och förändrar applikationens beteende i syfte att hjälpa användaren i dennes vardagliga arbetsuppgiften. Idag överförs kontextuell-information (eng. context information) i nätverk som är opålitliga och dynamiskt föränderliga. Därtill tillkommer komplexiteten att kontextuell-information är ibland producerad i olika noder anslutna till olika nätverk.Utvecklingen av kontext-medvetna applikationer har hittills begränsats av ovannämnda svårigheter. Denna avhandling presenterar en metod för att distribuera kontextuell-information genom användning av mekanismer för händelsemeddelande (eng. event notification mechanisms) inbyggda i Session Initiation Protocol (SIP). Målet är att undersöka hur metoden kan användas för att möjliggöra tillgång till kontextuell-information oavsett vart den är producerad. Komponenten för distribution av kontextuell data, som presenteras i denna uppsats, använder SIP för direktmeddelanden (eng. Instant Messaging) och tekniken “Presence Leveraging Extensions (SIMPLE)” för datadelning av kontextuell data (eng. Context sharing). För detta ändamål används SIP närvaroserver (eng. SIP presence server), mer specifikt modulen för närvaroinformation tillhörande SIP Expressroutrar (SER). Komponenten för distribution av kontextuell information möjliggör både synkront och asynkront distribution. Valet mellan de två beror delvist på applikationens kravspecifikation för distribution av kontextuell information, delvist på typen av den kontextuella informationen. Baserat på systemet skalbarhet (eng. Scalability), användarens rörlighet och latens (eng. latency) kan man ge rekommendationer vilken av de två distributionssätten, synkront eller asynkront, som är lämpligast för distributionen av kontextuell information. Systemet utvärderades med hjälp av ett program som genererar belastning (eng. load generator). Resultaten visar att systemet är mycket skalbart. Responstiden för synkront åtkomst av kontextuell information är nästan konstant, medan responstiden för asynkront åtkomst ökar med informationsmängden i databasen, i respekt till den föregående prenumerationen av kontextändringar. Händelsemeddelande skickas regelbundet ( 2800 meddelande per sekund). Vi har dock medvetet valt att skapa en slumpmässigt dröjsmål (0 till 1 sekund) mellan varje uppdatering av kontextuell information (t.ex. en kvitto på en Publish-meddelande) och den tidpunkten då händelsemeddelande skickas till de användare som prenumererar på ändringarna. För utvecklingen av varje kontext-medveten applikation, som distribuerar kontextuell information måste man ta hänsyn till responstid vid beslut huruvida man ska välja synkront eller asynkront sätt för distribution. Denna uppsats ger empirisk data som hjälper applikationsutvecklare i detta val.
20
Romero, Eduardo Luis. "Mise en oeuvre des protocoles SIP et RTP sur système embarqué." Mémoire, Université de Sherbrooke, 2009. http://savoirs.usherbrooke.ca/handle/11143/1504.
Full textAbstract:
L'avènement de la VoIP (Voice over IP) a déclenché une période de profonds changements dans le marché des télécommunications. En particulier, dans le secteur de la téléphonie résidentielle, cette technologie s'est consolidée, rapidement et pour de nombreuses raisons, comme l'évolution de la téléphonie traditionnelle. Dès les tous débuts, et afin d'établir une base de compatibilité permettant l'interconnexion de plusieurs réseaux téléphoniques et la convergence entre les systèmes traditionnels analogiques et leur évolution numérique, l'industrie a demandé l'établissement de cadres normatifs. En réponse à ces besoins, plusieurs standards et protocoles, avec de successives modifications et corrections, ont été publiés dans une période relativement brève. Parmi les plus populaires, SIP (Session Initiation Protocol), un protocole de signalisation, et RTP (Real-Time Transport Protocol), un protocole de transport de flots temps réel, se démarquent et ils sont au coeur de la majorité des applications conçues actuellement. Bien que, aujourd'hui, SIP et RTP sont liés fortement à la téléphonie sur IP, leur portée et leurs possibilités sont beaucoup plus vastes, ce qui déclenche un grand intérêt et justifie l'effort mis dans la conception des implémentations plus performantes et orientées plus spécifiquement à divers serveurs mandataires UA (User Agent). Dans ce contexte, le but du présent projet de maîtrise est de concevoir des piles de protocoles SIP et RTP orientées vers des applications de téléphonie sur IP, dans un environnement embarqué. Des conditions additionnelles sont que les piles doivent être codées en langage C et s'appuyer sur le système d'exploitation en temps réel MicroC/OS-II. Afin de faciliter la portabilité, il doit se prévoir des couches d'abstraction du matériel et du système d'exploitation. Même si les applications ciblées pour le projet sont, principalement, celles de VoIP, la pile SIP doit viser d'autres domaines, notamment des applications de domotique et de contrôle à distance. Cette dernière condition impose, de façon indirecte, d'autres conditions sur la taille du code et la puissance de calcul demandée, car le matériel pour ces types d'applications est d'habitude plus simple et moins puissant que les ordinateurs qui sont souvent utilisés dans les applications professionnelles de communication. Ce mémoire, qui décrit le travail effectué, est organisé en deux parties. La première fait une introduction théorique à la téléphonie sur IP, et sert de fondement à la deuxième partie, où la mise en oeuvre des protocoles SIP et RTP est décrite en détail.L'accent a été mis sur les justifications des décisions prises pendant toute la conception afin d'aider à mieux comprendre la logique appliquée et de permettre sa reconsidération et analyse dans de futures itérations. Comme résultat des contraintes et limitations imposées dans le cadre de ce projet, les piles de protocoles conçues se sont révélées très compactes et performantes, ce que justifie pleinement la continuité du travail dans l'avenir.
21
El, Saghir Bassam. "A new approach for context-aware management of SIP communications." Evry, Institut national des télécommunications, 2009. http://www.theses.fr/2009TELE0009.
Full textAbstract:
Le secteur mondial de télécommunications a connu des bouleversements sans précédent pendant ces dernières années avec l’avènement de nouveaux services et de nouvelles technologies de communication. Les opérateurs de télécommunications subissent en effet une décroissance continue de leur revenu par utilisateur en raison d’une compétition toujours accrue et, dans une certaine mesure, de la saturation du marché pour les services les plus traditionnels. Afin d’attirer de nouveaux usagers, de retenir ceux qui existent déjà et d’augmenter le revenu par usager, les services de communication proposés par les opérateurs doivent prendre en compte le contexte de l’utilisateur. L’interfonctionnement entre les solutions proposées pour la prise en compte du contexte et les réseaux de communication actuels et de nouvelle génération représentent un grand défi tant pour les opérateurs que les fournisseurs de services de communications. Ce travail de thèse traite des questions relatives au développement des systèmes de communications adaptés au contexte en proposant un agent coté réseau nommé INCA (Intelligent Network-based Communication Assistant). L’INCA permet une gestion avancée des communications SIP (Session Initiation Protocol) adaptée aux informations de contexte qui sont recueillies à travers un cadre dédié à la publication et la notification de contexte. Son architecture multicouche est basée sur un modèle de couche générique et elle implémente une approche orientée plan pour la gestion des sessions SIP. Elle repose aussi sur un nouveau modèle de communication avec prise en compte du contexte pour permettre une adaptation des communications basée sur les préférences utilisateurIn recent years, the world telecommunications sector has undergone unprecedented changes driven mainly by the deployment of new communication technologies and services. Telecom operators are suffering from a steady decline in their revenues per user due to fierce competition and market saturation for traditional services. In order to attract new customers and retain existing ones, communication services proposed by these operators need to be aware of the user’s context, which includes information related to the user himself as well as his environment (e. G. His location, current activities and available devices). Unfortunately, interworking proposed context-aware solutions with current and next-generation networks still represents a big challenge for communication service providers as well as operators. This thesis addresses issues related to the development of context-aware communication systems by proposing a network-based agent called INCA (Intelligent Network-based Communication Assistant). INCA provides advanced management of SIP communications based on context information that is retrieved through a dedicated framework for context publication and notification. Its multilayered architecture is based on a generic layer model and implements a plan-centric approach for SIP session management. It also relies on a new context-aware communication model for providing communication adaptation based on user preferences
22
Trioschi, Luca. "Multiplexing di protocolli SIP e RTP su canali virtuali multipercorso: integrità e sicurezza." Bachelor's thesis, Alma Mater Studiorum - Università di Bologna, 2011. http://amslaurea.unibo.it/2706/.
Full text
23
Chtatou, Abdessamad. "Extension du protocole SIP pour le contrôle de la domotique distante." Mémoire, Université de Sherbrooke, 2005. http://hdl.handle.net/11143/8611.
Full textAbstract:
Il existe de nombreux standards destinés à contrôler les dispositifs servant à automatiser un habitat. En revanche, la plupart d'entre eux ne supportent pas le contrôle à distance de ces dispositifs, ni l'interconnexion de différentes technologies réseautiques. Un tel support devrait produire une radicale révolution dans le domaine de l'habitat communicant, puisqu'il serait possible de faire cohabiter, à la fois et dans le même domaine, non seulement les terminaux informatiques comme les PCs portables et les PDAs, mais également les dispositifs qui ont des tâches spéciales (une machine à café par exemple). Ainsi, l'usager final profiterait d'un menu riche contenant une multitude de services qui étaient inimaginables il n'y a pas longtemps. Le protocole SIP (Session Initiation Protocole) qui est utilisé originellement pour la signalisation en téléphonie IP, peut fournir l'habilité à se connecter via Internet aux dispositifs domestiques. Ce projet propose une extension de SIP dans le cadre de la domotique distante : contrôle et gestion d'une résidence à distance. Une méthode "DO" pour contrôler les appareils domestiques ainsi que le support du protocole de description "DMP" (Device Message Protocol) ont été ajoutés au protocole SIP pour permettre l'envoi de commandes à des dispositifs dans la résidence (Network Appliances). L'ensemble de commandes a été testé avec une mise en œuvre sur la plateforme OSG et en utilisant le protocole UPnP sur le réseau résidentiel.
24
Deng, Xianglin. "Security of VoIP : Analysis, Testing and Mitigation of SIP-based DDoS attacks on VoIP Networks." Thesis, University of Canterbury. Computer Science and Software Engineering, 2008. http://hdl.handle.net/10092/2227.
Full textAbstract:
Voice over IP (VoIP) is gaining more popularity in today‟s communications. The Session Initiation Protocol (SIP) is becoming one of the dominant VoIP signalling protocol[1, 2], however it is vulnerable to many kinds of attacks. Among these attacks, flood-based denial of service attacks have been identified as the major threat to SIP. Even though a great deal of research has been carried out to mitigate denial of service attacks, only a small proportion has been specific to SIP. This project examines the way denial of service attacks affect the performance of a SIP-based system and two evolutionary solutions to this problem that build on each other are proposed with experimental results to demonstrate the effectiveness of each solution. In stage one, this project proposes the Security-Enhanced SIP System (SESS), which contains a security-enhanced firewall, which evolved from the work of stage one and a security-enhanced SIP proxy server. This approach helps to improve the Quality-of-Service (QoS) of legitimate users during the SIP flooding attack, while maintaining a 100 percent success rate in blocking attack traffic. However, this system only mitigates SIP INVITE and REGISTER floods. In stage two, this project further advances SESS, and proposes an Improved Security-Enhanced SIP System (ISESS). ISESS advances the solution by blocking other SIP request floods, for example CANCEL, OK and BYE flood. JAIN Service Logic Execution Environment (JAIN SLEE) is a java-based application server specifically designed for event-driven applications. JAIN SLEE is used to implement enhancements of the SIP proxy server, as it is becoming a popular choice in implementing communication applications. The experimental results show that during a SIP flood, ISESS cannot only drop all attack packets but also the call setup delay of legitimate users can be improved substantially compared to and unsecured VoIP system.
25
Rufino, Leonardo Maccari. "Integração do protocolo SIP à norma IEEE 1451 para redes de sensores sem fio." Florianópolis, SC, 2012. http://repositorio.ufsc.br/xmlui/handle/123456789/96189.
Full textAbstract:
Dissertação (mestrado) - Universidade Federal de Santa Catarina, Centro Tecnológico. Programa de Pós-Graduação em Ciência da Computação.Made available in DSpace on 2012-10-26T09:38:57Z (GMT). No. of bitstreams: 1 302647.pdf: 3317702 bytes, checksum: 0b78bafe8cfda4c10b5581993bff11db (MD5)
Redes de sensores sem fio (RSSF) são compostas por dispositivos chamados nós sensores, os quais são capazes de monitorar alguns fenômenos do meio ambiente que os rodeia, tais como informações escalares (temperatura, aceleração) ou multimídia (áudio, vídeo), transformando-os em sinais digitais e comunicando-se com outros nós da rede. A fim de padronizar o acesso e o comportamento das diversas plataformas existentes, a família de padrões IEEE 1451 foi desenvolvida. Esta padronização introduz conceitos interessantes, como a divisão do sistema em duas partes principais, NCAP (Network Capable Application Processor) e TIM (Transducer Interface Module), e a definição dos TEDS (Transducer Electronic Data Sheet). Porém, o padrão não trata eficientemente os requisitos das RSSF atuais, tal como a necessidade dos sensores executarem de forma eficiente e energeticamente consciente, permitindo economizar sua energia, fator crítico em grande parte destes dispositivos. Assim, este trabalho apresenta um novo modo de execução chamado TIM-IM (TIM Initiated Message), o qual permite que TIMs reportem seus dados sempre que houver novas leituras sensoriadas, ao invés de aguardar por polling originado pelo NCAP, evitando permanecer com o módulo de comunicação ligado grande parte do tempo. Adicionalmente, o padrão IEEE 1451 limita-se às redes de sensores que captam informações escalares. Assim, a presente dissertação visa, também, a integração de sensores multimídia à norma, apresentando algumas modificações tanto nos TEDS quanto nas mensagens trafegadas entre NCAP e TIM. A fim de permitir o acesso aos sensores através da rede do usuário, foi utilizado o protocolo SIP (Session Initiation Protocol). SIP vem sendo bastante utilizado atualmente junto à tecnologia VoIP (Voice over Internet Protocol), sendo responsável por estabelecer, modificar e finalizar uma sessão. Devido ao seu tamanho, torna-se inviável seu uso em muitos sistemas embarcados com restrição de recursos. Logo, este trabalho apresenta uma miniaturização do mesmo, alcançada através da eliminação de algumas requisições e campos de cabeçalho (do inglês header fields). Por fim, é apresentada a integração do protocolo SIP ao IEEE 1451. Para isto, foi utilizado o estabelecimento de sessões, assim como o esquema de notificação de presença presente no SIP e a extensão relativa à transferência de mensagens instantâneas. Assim, com a união de ambas as normas, permite-se que sensores sejam acessados por usuários remotos utilizando SIP phones, através da Internet, independentemente de sua localização física.
Wireless sensor networks (WSN) are formed by devices called sensor nodes capable of monitoring some phenomena around them, such as scalar information (temperature, acceleration) or multimedia (audio, video), transforming them into digital signals and communicating with other nodes. In order to standardize the access and behavior of the various platforms available, the IEEE 1451 standards family was developed. This standardization introduces interesting concepts, such as splitting the system into two major parts, NCAP (Network Capable Application Processor) and TIM (Transducer Interface Module), and the definition of TEDS (Transducer Electronic Data Sheet). However, the standard does not address efficiently the requirements of current WSN, such as the need for sensors perform efficiently and energyconscious, saving its energy, which is critical for most of these devices. This work presents a new execution mode called TIM-IM (TIM Initiated Message), which allows TIMs to report its data whenever there are new sensed readings, rather than wait for polling originated by NCAP, avoiding remain with the communication module connected all the time. Additionally, IEEE 1451 is limited to sensor networks that collect scalar information. Thus, this thesis also aims at the integration of multimedia sensors to the standard, presenting some modifications in TEDS and in the messages sent between NCAP and TIM. In order to allow the access to sensors via user#s network, it was used the SIP (Session Initiation Protocol) protocol. SIP has been widely used today by the VoIP (Voice over Internet Protocol) technology and it is responsible to establish, modify and terminate a session. Due to its size, its use is not feasible in many resource-constrained embedded systems. Thus, this work presented a miniaturization of the protocol, achieved through the elimination of some requests and header fields. Finally, it was presented the integration of SIP to IEEE 1451. For this, it was used the session establishment, as well as the presence notification scheme of the SIP protocol and the extension for the transfer of instant messages. Thus, with the union of both standards, sensors can be accessed by remote users using SIP phones through the Internet, regardless of their physical location.
26
Moreno, Carlos. "Design of a network architecture for multimedia services with SIP multicast." Evry, Institut national des télécommunications, 2009. http://www.theses.fr/2009TELE0006.
Full textAbstract:
L'architecture proposée permet à un serveur de conférence d'agir sur les autres domaines avec des domaines de multicast de recouvrement et de multicast d'IP. C'est le résultat de l'amélioration des audioconférences conventionnelles avec 3 modules complémentaires : une unité d'extension de SIP, un agent de multicast à l'intérieur du serveur de conférence appelé MGA (agent de passage de multicast) et un directeur de multicast (responsable de la topologie). Le premier laisse le trafic unicast passer normalement entre un SIP UA (agent d'utilisateur) et le serveur de conférence bi-directionnel. Si le directeur de multicast donne n'importe quelles instructions, des messages de SIP sont convertis en multicast SIP et vice-versa. Le trafic téléphonique est également conduit vers le serveur de conférence, ou vers MGA pour l'expédition correcte. MGA est responsable du trafic de multicast de cheminement, après interprétation des messages prolongés par multicast de SIP envoyés de l'unité d'extension de SIP et renvoie des réponses selon une table applicable de multicast. Il peut également expédier le trafic téléphonique aux groupes de multicast d'IP en utilisant un ensemble de réflecteurs. Le serveur de conférence peut également inclure cette fonction elle-même. Le directeur de multicast envoie des instructions au prolongateur dans des groupes SIP multicast selon des ordres d'opérateurs. Les résultats prouvent que les temps pour rejoindre ou quitter le SIP multicast pour des membres de groupe sont plus courts que ceux du cas de multicast d'IP et la fiabilité augmente également, principalement dans les scénarios de congestion forteThe proposed architecture permits a conference server to interact with multicast domains of Overlay and IP Multicast. It is the result of the improvement of conventional audio conferences with 3 complementary modules: a SIP extender, a multicast agent inside the conference server called MGA (Multicast Gateway Agent) and a multicast manager (Virtualization of connections). SIP extender lets unicast traffic go through a SIP UA (user agent) and the conference server bidirectionally. If the multicast manager gives any instruction, SIP messages are converted into multicast SIP extended ones and vice versa. Voice traffic is also routed towards the conference server, or towards MGA for the correct dispatching. MGA is in charge of routing multicast traffic, after interpreting SIP multicast extended messages sent from the SIP extender and sending back responses according to a multicast applicative table. It can also forward voice traffic to IP multicast groups by using a set of reflectors. The conference server could also include this function itself. The multicast manager sends instructions to the SIP extender just to let it know exactly which user agents should be included into SIP multicast groups according to operator orders. The results regarding QoS show that joining and leaving time in SIP multicast for group members are shorter in comparison with the IP multicast approach and reliability also increases, mainly in high congestion scenarios
27
Benisse, Mohamed Taib. "Transmission média sur les réseaux IP en utilisant les protocoles SIP et IAX." Mémoire, École de technologie supérieure, 2009. http://espace.etsmtl.ca/27/1/BENISSE_Mohamed_Ta%C3%AFb.pdf.
Full textAbstract:
Les progrès technologiques du réseau Internet ont permis le développement de nouvelles
applications multimédia; la voix, la vidéo et la vidéoconférence sont devenues des domaines
importants de recherche et de développement pour l’industrie des télécommunications. Ces
dernières années ont été remarquables par la mise en oeuvre de connexion haute débit, et de
terminaux mobile et fixe performants. Plusieurs standards ont été conçus spécifiquement
pour permettre la transmission média sur les réseaux IP avec une meilleure qualité de service.
Ce travail a pour but d’étudier les protocoles de transmission média sur les réseaux IP, en
commençant par l’état de l’art de technologies principales pour accéder au réseau, les
techniques utilisées pour encoder l’audio et la vidéo, et en finissant par les protocoles de
transport combinés avec d’autres protocoles temps réels. L’objectif principal du mémoire est
d’analyser, et intégrer les protocoles de transmission (SIP, RTP et IAX) sur les réseaux IP.
Le projet se compose de deux parties : expérimentale et applicative. La première partie a
pour objectif de mettre en place une plateforme IPPBX capable de fournir une solution assez
complète de transmission média sur le réseau IP en utilisant les protocoles SIP et IAX.
Ensuite, nous allons calculer le temps requis de signalisation SIP/IAX et la qualité de service
d’une communication IAX en utilisant les codecs G.711 et GSM. La deuxième partie se
compose de la conception et l’implémentation du protocole RTP dans les téléphones mobiles
en utilisant la technologie J2ME pour permettre un environnement mobile de
vidéoconférence. Nous allons effectuer un rapport technique assez complet décrivant la
technologie mobile J2ME. Nous allons également tester les émulateurs et outils capables
d’offrir un environnement de vidéoconférence mobile et les difficultés associées aux codecs
supportés
Les résultats des expériences ont montré que le temps requis de signalisation SIP et IAX est
sous un seuil acceptable dans un réseau local. Selon les valeurs obtenues du délai et de la
gigue, la qualité de service de la communication IAX avec les codecs G.711 et GSM est
adéquate. Le résultat obtenu de la partie applicative nous a permis de prouver que le client
mobile de vidéoconférence est capable de s’enregistrer auprès d’un Proxy/Registrar pour
joindre une session multimédia et de signaliser avec d’autres clients de la session via le
protocole SIP. La conception du protocole RTP dans la technologie mobile adopte le RFC
3250 sur le plan théorique. L’architecture du système utilisé et les composantes logicielles
ont été bien mises en place. La transmission des paquets RTP a été bien réalisée. La
manipulation des paquets RTP en mode binaire a été bien effectuée pour rediriger les flux
audio et vidéo au lecteur JMStudio.
28
Kullenwall, Jonas. "Study of security aspects for Session Initiation Protocol." Thesis, Linköping University, Department of Electrical Engineering, 2002. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-1164.
Full textAbstract:
The objective with this thesis is to describe security mechanisms that are inte-grated or are proposed to be integrated with the Session Initiation Protocol (SIP). SIP is used for establishing, modifying, and terminating multimedia ses-sions over the IP network. This thesis is divided into two main parts, where the first part describes the implemented security mechanisms in SIP and the second part describes a number of proposed security mechanisms that may be implemented in SIP. At the end of the report there is a section that presents the scripts and results from different security tests that were performed on two implementations of SIP. Apart from describing different security mechanisms in the first part of this thesis, this section also contains an analysis on how possible security threats against SIP may be used to launch different attacks. The analysis also describes how these attacks may be prevented, if possible, by using the secu-rity mechanisms provided by SIP. The second part also contains an analysis section, which is focusing on finding the advantages and disadvantages of using a specific security mechanism compared to a similar security mechanism that is currently used or has been used in SIP. In the last section of this thesis I present my conclusions and a summary of the results.
29
Muswera, Walter Tawanda. "Developing a cross platform IMS client using the JAIN SIP applet phone." Thesis, Rhodes University, 2015. http://hdl.handle.net/10962/d1017934.
Full textAbstract:
Since the introduction of the IP Multimedia Subsystem (IMS) by the Third Generation Partnership Project (3GPP) in 2002, a lot of research has been conducted aimed at designing and implementing IMS capable clients and network elements. Though considerable work has been done in the development of IMS clients, there is no single, free and open source IMS client that provides researchers with all the required functionality needed to test the applications they are developing. For example, several open and closed source SIP/IMS clients are used within the Rhodes University Conver- gence Research Group (RUCRG) to test applications under development, as a result of the fact that the various SIP/IMS clients support different subsets of SIP/IMS features. The lack of a single client and the subsequent use of various clients comes with several problems. Researchers have to know how to deploy, configure, use and at times adapt the various clients to suit their needs. This can be very time consuming and, in fact, contradicts the IMS philosophy (the IMS was proposed to support rapid service creation). This thesis outlines the development of a Java-based, IMS compliant client called RUCRG IMS client, that uses the JAIN SIP Applet Phone (JSAP) as its foundation. JSAP, which originally offered only basic voice calling and instant messaging (IM) capabilities, was modified to be IMS compliant and support video calls, IM and presence using XML Configuration Access Protocol (XCAP).
30
Milanez, Mateus Godoi. "Avaliação dos protocolos VoIP SIP e IAX utilizando simulação e parâmetros de qualidade de voz." Universidade de São Paulo, 2009. http://www.teses.usp.br/teses/disponiveis/55/55134/tde-17062009-155138/.
Full textAbstract:
Recentemente, as tecnologias de telecomunicações esão convergindo para a concepção da Next Generation Network, onde propõe-se que todas as informações trocadas sejam classificadas por prioridade e segurança. Porém, como as redes atuais ainda não promovem tais práticas, protocolos VoIP, em conjunto a outras soluçõoes, buscam a melhoria da qualidade das ligações. Como o protocolo VoIP IAX vem ganhando credibilidade na comunidade open source nos úlltimos anos, torna-se relevante compará-lo ao protocolo SIP, o qual é bastante investigado pela literatura. Desta forma, o objetivo deste trabalho é o estudo e avaliação dos protocolos SIP e IAX, através de verificações de qualidade do áudio em ligações VoIP. Para a realização dos experimentos foi desenvolvida uma estrutura que representasse chamadas VoIP no simulador Network Simulator e, para tais ligações, empregou-se método de avaliação de qualidade PESQ. Assim, foi possível a verficação das semelhanças compreendidas entre os protocolos SIP e IAX diante dos problemas de perda de pacotes, atraso, limitação da taxa de dados e jitterTelecommunications technologies are recently converging to the Next Generation Network conception, where it is proposed that all exchanged information should be classied by security and priority. As the currently available networks do not provide such practices, VoIP protocols, among other solutions, aim for the improvement of the calls quality. As the IAX VoIP protocol had been receiving credibility in the open source community in the last years, it is relevant to compare it to the SIP protocol, which is widely investigated in the literature. In this way, the objective of this work is the study and evaluation of the SIP and IAX protocols through verications of audio quality in VoIP calls. To implement the experiments, a structure that represents VoIP calls was developed in the \"Network Simulator\" software. For these calls, the PESQ method was used to evaluate the calls quality. Using this approach, it was possible to verify similarities between the SIP and IAX protocols regarding the problems of packet loss, delay, limitation in the data rate and jitter
31
SCHEINER, LEONARDO NAHMIAS. "PERFOMANCE ANALISYS OF SIP PROTOCOL ON THE SIGNALING OF VOICE OVER IP CALLS." PONTIFÍCIA UNIVERSIDADE CATÓLICA DO RIO DE JANEIRO, 2005. http://www.maxwell.vrac.puc-rio.br/Busca_etds.php?strSecao=resultado&nrSeq=7065@1.
Full textAbstract:
COORDENAÇÃO DE APERFEIÇOAMENTO DO PESSOAL DE ENSINO SUPERIORImpulsionada pelo grande crescimento da Internet, a telefonia IP conquistou a atenção do mercado e dos grandes fabricantes com promessas de redução de custo na operacão, gerência, provisionamento, manutenção e tarifação. Diversos protocolos foram desenvolvidos de modo a prover VoIP como o H.323, MGCP, Megaco e SIP. O SIP tem se destacado por ser um protocolo baseado em texto, estensível, independente do protocolo de transporte, e portanto mais flexível e simples que seu concorrente direto, o H.323. O SIP (Session Initiation Protocol) é um protocolo de sinalização utilizado para iniciar, modificar e terminar sessões, podendo ser usado para chamadas de voz sobre IP (VoIP) ou para troca de mensagens instantâneas, entre outras aplicações. Ele foi desenvolvido originalmente em 1996 e foi padronizado pela IETF em 1999. Neste trabalho, o desempenho do protocolo SIP para estabelecimento de chamadas VoIP será avaliado, já que há uma grande quantidade de trabalhos focando a qualidade da voz e poucos têm avaliado a sinalização [3]. Serão montados ambientes experimentais a fim de variar parâmetros como retardo, perda de pacotes, jitter, largura de banda e protocolo de transporte, permitindo verificar como esses parâmetros afetam isoladamente os tempos de post-dial delay, post-pickup delay e call release delay.
Pushed by the growth of the Internet, the IP Telephony conquered a great attention of the market and big suppliers, with promises of cost reductions on operation, management, provisioning, maintenance and billing. Different protocols were developed for providing VoIP such as H.323, MGCP, Megaco and SIP. SIP has been highlighted for being a text based protocol, extensible, independent of the transport protocol, therefore more flexible and simpler than your competitor, the H.323. SIP (Session Initiation Protocol) is a signaling protocol used for establish, modify and terminate sessions. It can be used for voice calls over IP (VoIP) or to exchange instant messaging, among other applications. It has been developed originally in 1996 and has been standardized by IETF in 1999. In this work, the performance of SIP protocol for establishing VoIP calls will be estimated, since there are a lot of papers focalizing in the voice quality and few treated the signaling [3]. Experimental environments will be used for varying parameters like delay, packet loss, jitter, bandwidth and transport protocol, allowing to verify how there parameters affect separately the post-dial delay, post-pickup delay and call release delay.
32
Richert, Adam. "Developing a Portable System for Medicine Dosage." Thesis, KTH, Skolan för elektroteknik och datavetenskap (EECS), 2018. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-235738.
Full textAbstract:
The project presented in this report is set out to develop a portable electronic system to be used as a medicine pill container. With the functionality to configure up to twelve daily repeated alarms, the purpose of the medicine dosage system is first and foremost to remind the user when they should take their medicine. Secondly, LED lights and user-recorded voice notifications are to be implemented to further aid the user in taking the right medicine each time. The device is also to have a memory log, recording up to one hundred missed dosages, enabling an authorized medicine professional to verify the medicine adherence of the user.Prior to the start of the project, an outline for the functionality and physical appearance of the device was set by the project owner, Victrix AB. This project covers the hardware and software development, as well as the design choices within. The aim is to follow the proposed functionality specification as close as possible, while making justified hardware and software choices considering simplicity, efficiency, power consumption, and availability. By following the specification, the goal is ultimately to increase the medicine adherence for users of the device developed with this thesis.Using qualitative research methods, a valid background study was created, preceding the development of the medicine dosage system. Hardware for a first prototype of the device was then chosen based on the gathered information about existing technologies and related work. With thorough testing and recurrent information exchange with the client, a prototype of the medicine dosage system, based on an Arduino microcontroller, was constructed. The prototype was evaluated to fulfill 92% of the requirements considered as high priority by Victrix.Projektet som presenteras i denna rapport är tänkt att utveckla ett portabelt elektroniskt system för användning som en medicinsk pillerbehållare. Med funktionaliteten att konfigurera upp till tolv dagligen upprepande alarm är syftet med medicindoseringssystemet först och främst att påminna användaren när de ska ta sin medicin. Lysdioder och användarens egna inspelade röst som notifikationer ska implementeras för att vidare hjälpa användaren att ta rätt medicin vid varje tillfälle. Enheten ska också ha en minneslogg som sparar upp till etthundra missade doseringar, vilket gör det möjligt för auktoriserad sjukvårdspersonal att verifiera användarens följsamhet till medicineringen.En översiktlig beskrivning av funktionaliteten samt det fysiska utseendet av enheten skrevs av projektägaren Victrix AB innan projektet startades. Det som detta projekt täcker är hårdvaruoch mjukvaruutvecklingen, så väl som där tillhörande designval. Projektet siktar på att följa den föreslagna funktionalitetsspecifikationen så nära som möjligt, och samtidigt göra välgrundade val för hårdoch mjukvara med enkelhet, effektivitet, energiförbrukning och tillgänglighet i åtanke. Genom att följa specifikationen är det slutliga målet att frambringa ökad medicinföljsamhet för användare av den med det här projektet utvecklade enheten.Utvecklingen av medicindoseringssystemet föregicks av en befogad bakgrundsstudie utformad genom användningen av kvalitativa forskningsmetoder. Hårdvara att användas för en första prototyp av enheten valdes sedan baserat på den insamlade informationen om existerande teknologier och relaterat arbete. Genom grundliga tester och regelbundet informationsutbyte med kunden konstruerades en prototyp av medicindoseringssystemet baserat på en Arduinomikrokontroller. Prototypen utvärderades att uppfylla 92% av kraven som Victrix ansåg vara av hög prioritet.
33
Elleuch, Wajdi. "Mobilité des sessions dans les communications multimédias en mode-conférence basées sur le protocole SIP." Thèse, Université de Sherbrooke, 2011. http://hdl.handle.net/11143/5799.
Full textAbstract:
Ce mémoire traite la problématique de la mobilité des sessions pour le transfert des communications multimédias basées sur le protocole SIP. Plusieurs aspects sont étudiés et des mécanismes proposés afin de permettre la mobilité des sessions avant, durant et après leur établissement. En plus d'une communication impliquant deux intervenants, Il a été possible d'étendre l'utilisation de la mobilité des sessions pour l'appliquer aux scénarios de communications en mode conférence regroupant plusieurs intervenants. Les mécanismes de mobilité de session développés au cours de cette thèse sont par la suite déployés pour (1) permettre des transformations entre différentes topologies de conférences et (2) construire un modèle de conférence adapté pour l'échange de la voix au sein des groupes de communication à large échelle.
34
Rondini, Rogério Augusto. "Uma arquitetura baseada em espaço de tuplas para redes IMS." Universidade de São Paulo, 2012. http://www.teses.usp.br/teses/disponiveis/3/3141/tde-22052014-235246/.
Full textAbstract:
A arquitetura IP Multimedia Subsystem, proposta pelo consórcio 3rd Generation Partnership Project como base para o suporte à convergência entre telefonia móvel e a Internet, define uma série de elementos arquiteturais, entre os quais, o componente Call Session Control Function e o protocolo Session Initiation Protocol. Session Initiation Protocol é um protocolo da camada de aplicação utilizado para estabelecer, modificar e terminar sessões multimídia entre dispositivos. Em redes baseadas na arquitetura IP Multimedia Subsystem, o Session Initiation Protocol é o responsável pela comunicação entre dispositivos e a rede, e entre os componentes responsáveis pelo gerenciamento de sessão. Nos últimos anos, estudos detectaram degradação de desempenho em redes baseadas na arquitetura IP Multimedia Subsystem em função das características centralizadas do Session Initiation Protocol e dos componentes de gerenciamento de sessão. Este trabalho apresenta uma arquitetura distribuída para redes baseadas em IP Multimedia Subsystem, tendo como fundamento o paradigma de computação paralela baseado em espaço de tuplas onde os servidores são organizados em uma rede P2P, com objetivo de prover uma infraestrutura escalável e tolerante a falhas. A validação da arquitetura em termos de desempenho e escalabilidade se deu através de modelagem formal e simulação com Redes de Petri Coloridas.The IP Multimedia Subsystem architecture, proposed by the 3rd Generation Partnership Project consortium as basis to support the convergence between mobile networks and the Internet, defines a set of architectural elements, among them, the Call Session Control Function and the Session Initiation Protocol. The Session Initiation Protocol is an application layer protocol used to establish, modify and terminate sessions between devices. On the IP multimedia subsystem based network, the Session Initiation Protocol play a key role on the communication between devices and the network, and between session management components. In the last years, studies have detected a performance bottleneck on IP multimedia subsystem networks due to centralized characteristic of the Session Initiation Protocol and in Session Control components. This work shows a distributed architecture for IP Multimedia Subsystem networks based on the tuple space paradigm, and the servers structured in a P2P network, aiming to achieve a scalable and fault-tolerant infrastructure. The validation of the architecture on the performance and scalability took place through the Coloured Petri Net formal modeling and simulation.
35
Jacobs, Ashley. "Investigating call control using MGCP in conjuction with SIP and H.323." Thesis, Rhodes University, 2005. http://hdl.handle.net/10962/d1006516.
Full textAbstract:
Telephony used to mean using a telephone to call another telephone on the Public Switched Telephone Network (PSTN), and data networks were used purely to allow computers to communicate. However, with the advent of the Internet, telephony services have been extended to run on data networks. Telephone calls within the IP network are known as Voice over IP. These calls are carried by a number of protocols, with the most popular ones currently being Session Initiation Protocol (SIP) and H.323. Calls can be made from the IP network to the PSTN and vice versa through the use of a gateway. The gateway translates the packets from the IP network to circuits on the PSTN and vice versa to facilitate calls between the two networks. Gateways have evolved and are now split into two entities using the master/slave architecture. The master is an intelligent Media Gateway Controller (MGC) that handles the call control and signalling. The slave is a "dumb" Media Gateway (MG) that handles the translation of the media. The current gateway control protocols in use are Megaco/H.248, MGCP and Skinny. These protocols have proved themselves on the edge of the network. Furthermore, since they communicate with the call signalling VoIP protocols as well as the PSTN, they have to be the lingua franca between the two networks. Within the VoIP network, the numbers of call signalling protocols make it difficult to communicate with each other and to create services. This research investigates the use of Gateway Control Protocols as the lowest common denominator between the call signalling protocols SIP and H.323. More specifically, it uses MGCP to investigate service creation. It also considers the use of MGCP as a protocol translator between SIP and H.323. A service was created using MGCP to allow H.323 endpoints to send Short Message Service (SMS) messages. This service was then extended with minimal effort to SIP endpoints. This service investigated MGCP’s ability to handle call control from the H.323 and SIP endpoints. An MGC was then successfully used to perform as a protocol translator between SIP and H.323.
36
Doring, Mathieu. "Développement d'une méthode SPH pour les applications à surface libre en hydrodynamique." Nantes, 2005. http://www.theses.fr/2005NANT2116.
Full textAbstract:
Le développement de méthodes numériques plus performantes associé à l’augmentation de la puissance de calcul disponible ont permis la simulation d’écoulement toujours plus complexes. Cependant, les problèmes impliquant un interface restent difficiles à résoudre, notamment dans les cas où interviennent du déferlement, des reconnections d’interface ou des phénomènes d’impacts. Ces difficultés sont liés principalement à la gestion de l’interface dans des méthodes numériques principalement Euleriennes (Volume Of Fluid, Level Set). La méthode SPH, étant Lagrangienne permet une prise en compte simplifiée de l’interface. Dans ce travail de thèse, une méthode numérique basée sur la méthode SPH a été développée, permettant de simuler des écoulements complexes faisant intervenir des phénomènes d’impacts et de déferlements très importants. Une attention particulière a été portée sur l’amélioration de la précision de la méthode ; ainsi différents types de schéma (Moving Least Square, Renormalisation) et différents types de conditions aux limites (Particules Gelées, Particules Fantômes) ont été implémentés et testés. Les résultats obtenus portent sur différents cas tests: ffondrement d’une colonne d’eau avec ou sans obstacle, écoulement dans une cuve en cavalement avec impact sur le plafond, simulation d’un bassin de houle avec batteur piston. Ces résultats ont été comparés avec des résultats issus d’autres méthodes numériques actuellement en développement au sein du laboratoire de l’école Centrale de Nantes (une méthode VOF-Volume Fini, et une méthode potentielle spectrale), mais aussi avec les résultats expérimentaux disponibles. Le développement d’une méthode originale permettant d’obtenir les efforts sur les parois solides a permis la simulation de cas d’impact d’un dièdre en mouvement libre, et la comparaison d’efforts d’impacts dans le cas d’un effondrement de colonne d’eau en présence d’un obstacle avec des efforts expérimentaux. Par ailleurs, la parallélisation du code a été entamée, par deux méthodes : la première était basée sur un décomposition des données (OpenMP) mais a été abandonnée du fait de son manque d’efficacité. La méthode actuellement employée est basée sur une décomposition du domaine physique, chaque morceau du domaine étant alloué à un processeur. Des tests menés sur un PC ont permis de montrer une bonne efficacité de la méthode de décomposition de domaine, notamment grâce au recouvrement des temps de communications entre procès par d’autres opérations. On s’est également intéressé à l’optimisation algorithmique du code, et notamment des procédures de recherche de voisins, ou un algorithme de Verlet est utilisé. Le développement du code a été effectué en gardant toujours la possibilité de réaliser des simulations tridimensionnelles en faisant un minimum d’adaptation. C’est pourquoi l’outil développé permet d’envisager dans un futur proche des simulations précises d’envahissement tridimensionnelRecent development in numerical methods together with the increase of computational power available have allowed simulations of more and more complex flows. However interfacial flows remains a difficult task, especially when breaking, interface reconnection or impacts occurs. Theses difficulties arise mainly from the deformations of the computational domain during the simulation which are badly handled by mesh based Eulerian numerical methods. Smoothed Particle Hydrodynamics, being meshless and Lagrangian allows a simplified management of the interface. In this PhD a SPH based numerical methods has been developed in order to simulate complex free surface flows with impacts and huge breaking. A particular care was taken concerning increasment of the precision; thus different discretization schemes have been tested (Moving Least Square, Renormalisation) as well as boundary conditions (frozen particles, ghost particles) were implemented and tested. Comparison of obtained results with both experimental results and numerical simulations from different numerical methods (Volume of Fluid_Finite volume solvers, spectral potential solver) in development in the Fluid Mechanics Laboratory in a variety of test cases as dam breaking, sloshing in a tank, virtual bassin, impact of solid through free surface shows good agreement, confirming the potential of the SPH method in naval hydrodynamics. The development of an original and new method allowing the obtention of loads on solid boundary made us able to compute loads exerted on an obstacle in a dam breaking test case and to simulate impacts of a wedge in free motion with favourable comparison against experimental dynamic conditions in both configurations. Moreover, a work on parallelization of the code has been carried out, firstly in a data decomposition approach (OpenMP) which we gave up due to its poor efficiency. Then a domain decomposition method was implemented using MPI library and showed good results concerning speed-up in various configurations (PC cluster, SuperComputer) thanks to the overlapping of communication time by standard SPH serial operations. Finally the use of a Verlet like algorithm for neighbor search allowed the optimisation of computationnal efficiency. Thanks to code organisation, three dimensionnal simulations are possible with minimum adaptation
37
Seifert, David. "Brána pro překlad signalizačních zpráv pro videokonference." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2013. http://www.nusl.cz/ntk/nusl-220209.
Full textAbstract:
Generally, the term gateway (gateway) in computer networks holds a node that connects two networks with different protocols. This thesis deals with the translation between two protocols, SIP protocol and WebSocket. This translation is the work first described theoretically and then work deals with the way this translation solutions using tools available on the Internet.
38
Hsieh, Ming Chih. "Service provisioning in two open-source SIP implementation, cinema and vocal." Thesis, Rhodes University, 2013. http://hdl.handle.net/10962/d1008195.
Full textAbstract:
The distribution of real-time multimedia streams is seen nowadays as the next step forward for the Internet. One of the most obvious uses of such streams is to support telephony over the Internet, replacing and improving traditional telephony. This thesis investigates the development and deployment of services in two Internet telephony environments, namely CINEMA (Columbia InterNet Extensible Multimedia Architecture) and VOCAL (Vovida Open Communication Application Library), both based on the Session Initiation Protocol (SIP) and open-sourced. A classification of services is proposed, which divides services into two large groups: basic and advanced services. Basic services are services such as making point-to-point calls, registering with the server and making calls via the server. Any other service is considered an advanced service. Advanced services are defined by four categories: Call Related, Interactive, Internetworking and Hybrid. New services were implemented for the Call Related, Interactive and Internetworking categories. First, features involving call blocking, call screening and missed calls were implemented in the two environments in order to investigate Call-related services. Next, a notification feature was implemented in both environments in order to investigate Interactive services. Finally, a translator between MGCP and SIP was developed to investigate an Internetworking service in the VOCAL environment. The practical implementation of the new features just described was used to answer questions about the location of the services, as well as the level of required expertise and the ease or difficulty experienced in creating services in each of the two environments.KMBT_363
Adobe Acrobat 9.54 Paper Capture Plug-in
39
Yeftsifeyeu, Aliaksandr. "Hlasová služba v integrovaných sítích." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2014. http://www.nusl.cz/ntk/nusl-220652.
Full textAbstract:
The master’s thesis presents routing protocols, SIP handshake, describes modern VoIP networks and its features. Thesis is primary focused on networks with integrated MPLS. The most known advantages and disadvantages were described of modern network based on such technology. Further, networks and a router with installed Cisco Call Manager Express 7.1 were configured, which is provided by CISCO. For recreating of fully working telephone network IP PBX10 has been used from SMC Networks. For connection between different phones within these integrated networks, SIP trunk was configured on SMC PBX10 as well as on router, which is connecting three configured Cisco IP phones. Each phone has its own number according to the dial plan. For calls from one network to another the special pattern has been established, so the number of each caller can be easily identified by added digit. On one of the CISCO routers was shown a configuration of MPLS and OSPF protocols. With an analyzer VePAL TX300e, the lab network has been measured to analyze QoS parameters of the network according to standard RFC 2544. Thesis also gives references to lab devices, which have been used accordingly to the work.
40
Crespi, Noël. "Evolutions des architectures de services pour maîtriser l'hétérogénéité de l'IMS." Paris 6, 2006. http://www.theses.fr/2006PA066513.
Full textAbstract:
SIP (Session Initiation Protocol) est un protocole de signalisation conçu pour le contrôle de session multimédia dans un réseau IP. Le protocole SIP est déjà largement adopté par l’industrie pour les services convergents de prochaine génération qui incluent notamment voix, vidéo, jeux, information de Présence, Chat, conférences multimédia. L’IP Multimedia Subsystem (IMS) constitue l’évolution majeure du Cœur de Réseau UMTS permettant l'utilisation du protocole SIP en adéquation avec les contraintes des réseaux mobiles. L’objet de cette thèse est d’analyser les obstacles existants au déploiement de services multimédia basés sur le protocole SIP dans les réseaux fixes et mobiles et de proposer des solutions aux verrous correspondants. Les études menées se décomposent en deux principaux volets : d’une part on analyse l’architecture multimédia SIP définie par l’IP Multimédia Subsystem afin de le mutualiser à partir d’accès hétérogènes (xDSL, câble, 3G). D’autre part on analyse les aspects relatifs aux architectures de service, en particulier : mécanismes de déclenchement, gestion des interactions entre services ou entre primitives, prise en compte des caractéristiques des terminaux dans la logique de service et commande de média. Pour chacun de ces domaines nous proposons les modifications protocolaires et d’architecture ainsi que les mécanismes correspondants. Le but de ces évolutions est de permettre - ou de faciliter- la mise en œuvre de services convergents et innovants dans les réseaux fixes et mobiles.
41
Švec, Michal. "Dohledový systém pro Internet Protocol Multimedia Subsystem." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2013. http://www.nusl.cz/ntk/nusl-220193.
Full textAbstract:
The master’s thesis describes IMS (IP Multimedia Subsystem) in terms of IMS core ele- ments (functional description, different implementation, signaling etc.) Communication protocols SIP and DIAMETER, together with SNMP protocol, which is used for collecting data are briefly described in this thesis.Thesis is also describing various IMS projects to- gether with Open IMS project, for whom was this surveillance system designed. Next part deals with architecture design of surveillance system along with management options implemented in surveillance system for users and administrators. The main part of master’s thesis deals with the description of the surveillance system for the experimental school Open IMS network and describes the remote configuration of core elements and monitoring of network traffic, together with the monitoring servers performance. The most of the data in the designed surveillance system are processed into graphs, which are regularly updated. The final part of master’s thesis describes the configuration and implementation of monitoring tools MRTG and NfSen that were used in created web based surveillance system.
42
Hayrapetyan, Anush. "Formalized, validated and executable CPN models of SIP-based presence and dynamic discovery protocols for mobile applications." Click here for download, 2007. http://proquest.umi.com/pqdweb?did=1288671611&sid=2&Fmt=2&clientId=3260&RQT=309&VName=PQD.
Full text
43
Che, Xiaoping. "Cross-fertilizing formal approaches for protocol conformance and performance testing." Thesis, Evry, Institut national des télécommunications, 2014. http://www.theses.fr/2014TELE0012/document.
Full textAbstract:
Les technologies de communication et les services web sont devenus disponibles dans notre vie numérique, les réseaux informatiques continuent de croître et de nouveaux protocoles de communication sont constamment définis et développés. Par la suite, la standardisation et la normalisation des protocoles sont dispensables pour permettre aux différents systèmes de dialoguer. Bien que ces normes peuvent être formellement vérifiés, les développeurs peuvent produire des erreurs conduisant à des implémentations défectueuses. C'est la raison pour laquelle leur mise en œuvre doit être strictement examinée. Cependant, la plupart des approches de tests actuels exigent une stimulation de l’exécution dans le cadre des tests (IUT). Si le système ne peut être consulté ou interrompu, l'IUT ne sera pas en mesure d'être testé. En outre, la plupart des travaux existants sont basées sur des modèles formels et très peu de travaux s'intéressent à la formalisation des exigences de performance. Pour résoudre ces problèmes, nous avons proposé une approche de test basé sur la logique "Horn" afin de tester passivement la conformité et la performance des protocoles. Dans notre approche, les exigences peuvent être formalisées avec précision. Ces exigences formelles sont également testées par des millions de messages collectés à partir des communicants réels. Les résultats satisfaisants des expériences effectuées ont prouvé le bon fonctionnement et l'efficacité de notre approche. Aussi pour satisfaire les besoins croissants de tests distribués en temps réel, nous avons également proposé un cadre de tests distribués et un cadre de tests en ligne et nous avons mis en œuvre notre plateforme dans un environnement réel à petite échelle avec succèsWhile today’s communications are essential and a huge set of services is available online, computer networks continue to grow and novel communication protocols are continuously being defined and developed. De facto, protocol standards are required to allow different systems to interwork. Though these standards can be formally verified, the developers may produce some errors leading to faulty implementations. That is the reason why their implementations must be strictly tested. However, most current testing approaches require a stimulation of the implementation under tests (IUT). If the system cannot be accessed or interrupted, the IUT will not be able to be tested. Besides, most of the existing works are based on formal models and quite few works study formalizing performance requirements. To solve these issues, we proposed a novel logic-based testing approach to test the protocol conformance and performance passively. In our approach, conformance and performance requirements can be accurately formalized using the Horn-Logic based syntax and semantics. These formalized requirements are also tested through millions of messages collected from real communicating environments. The satisfying results returned from the experiments proved the functionality and efficiency of our approach. Also for satisfying the increasing needs in real-time distributed testing, we also proposed a distributed testing framework and an online testing framework, and performed the frameworks in a real small scale environment. The preliminary results are obtained with success. And also, applying our approach under billions of messages and optimizing the algorithm will be our future works
44
Mehta, Anil. "MAC AND APPLICATION LAYER PROTOCOLS FOR HIGH PERFORMANCE NETWORKING." OpenSIUC, 2011. https://opensiuc.lib.siu.edu/dissertations/396.
Full textAbstract:
High-performance networking (HPN) is of significance today in order to enable next-generation applications using wired and wireless networks. Some of the examples of HPN include low-latency industrial sensing, monitoring and automation using Wireless Sensor Networks (WSNs). HPN however requires protocol optimization at many layers of the open system interface (OSI) network model in order to meet the stringent performance constraints of the given applications. Furthermore, these protocols need to be impervious to denial of service (DoS) and distributed DoS (DDoS) attacks. Some of the key performance aspects of HPN are low point-to-point and end-to-end latency, high reliability of transmitted frames and performance predictability under various network load situations. This work focuses on two discrete issues in designing protocols for HPN applications. The first research issue looks at the Medium Access Control (MAC) layer of the OSI network model for designing of MAC protocols that provide low-latency and high reliability for point-to-point communication under a WSN. Existing standards in this area are governed by IEEE 802.15.4 specification which defines protocols for MAC and PHY layers for short-range, low bit-rate, and low-cost wireless networks. However, the IEEE 802.15.4 specification is inefficient in terms of latency and reliability performance and, as a result, is unable to meet the stringent operational requirements as defined by counterpart wired sensor networks. Work presented under current research issue describes new MAC protocols that are able to show low-latency transmission performance under strict timing constants for power limited WSNs. This enhancement of the MAC protocols is named extended GTS (XGTS) contained under extended CFP (ECFP) and is published under the IEEE's 802.15.4e standard. The second research issue focuses on the application layer of the OSI network model to design protocols that enhance the robustness of the text based protocols to various traffic inputs. The purpose of this is to increase the reliability of the given text based application layer protocol under a varied load. Session Initiation Protocol (SIP) is used as a case study and the work aims to build algorithms that ensure that SIP can continue to function under specific traffic conditions, which would otherwise deem the protocol useless due to DoS and DDoS attacks. Proposed algorithms investigate techniques that enhance the robustness of the SIP against parsing attacks without performing a deep parse of the protocol data unit (PDU). The desired effect of this is to reduce the time spent in parsing the SIP messages at a SIP router and as a result increase the number of SIP messages processed per unit time at a SIP router.
45
Deusajute, Alexandre Machado. "Proposta de um mecanismo de segurança alternativo para o SIP utilizando o protocolo Massey-Omura aperfeiçoado com o uso de emparelhamentos bilineares." Universidade de São Paulo, 2010. http://www.teses.usp.br/teses/disponiveis/3/3141/tde-20122010-155116/.
Full textAbstract:
Voz sobre IP (ou VoIP) vem sendo adotada progressivamente não apenas por um grande número de empresas mas também por um número expressivo de pessoas, no Brasil e em outros países. Entretanto, essa crescente adoção de VoIP no mundo traz consigo algumas preocupações tais como ameaças e riscos de segurança, sobretudo no que diz respeito à autenticidade, privacidade e integridade da comunicação. Para proteger a sessão de mídia existem protocolos muito eficientes, como o Secure Real-time Transport Protocol (SRTP). Mas ele depende de uma chave secreta para tornar a comunicação segura de fato. Assim, uma boa estratégia é aproveitar o processo de sinalização que estabelece a sessão de mídia e negociar uma chave secreta de sessão que seja comum às partes comunicantes. Esse processo de sinalização é realizado por tipos específicos de protocolo tais como o Session Initiation Protocol (SIP), um protocolo de sinalização muito importante e que vem sendo usado cada vez mais por softphones para comunicação na Internet. Todavia, os riscos e ameaças mencionados já existem no próprio processo de sinalização e, dentre eles, o ataque do tipo man-in-the-middle é o mais perigoso, devido aos prejuízos que ele pode causar. Depois de fazer uma revisão bibliográfica dos riscos e ameaças inerentes ao SIP, bem como de seus mecanismos de segurança (analisando os pontos fortes e de atenção deles), foi possível originar um novo mecanismo de segurança, o qual é apresentado neste trabalho. O mecanismo proposto usa um protocolo para troca segura de informações o protocolo Massey-Omura o qual, quando combinado com emparelhamentos bilineares, provê ao SIP um melhor nível de segurança em todos os aspectos (autenticidade, privacidade e integridade). Além disso, o novo mecanismo é avaliado através de uma prova de conceito, na qual utilizou-se um softphone SIP funcional. A análise de segurança realizada e os resultados obtidos da prova de conceito fazem do mecanismo de segurança proposto uma alternativa viável para o SIP.Voice over IP (or VoIP) has been progressively adopted not only by a great number of companies but also by an expressive number of people, in Brazil and in other countries. However, this increasing adoption of VoIP in the world brings some concerns such as security risks and threats, mainly on the authenticity, privacy and integrity of the communication. In order to protect the media session, efficient protocols like the Secure Real-time Transport Protocol (SRTP) have been used. However, it depends on a secret key to make the communication secure. Thus, a good strategy is to take advantage of the signaling process to establish the media session, and agree on a common secret session key between the communicating parties. This signaling process is performed by specific types of protocols such as the Session Initiation Protocol (SIP), a very important signaling protocol, which has been used more and more by softphones in the Internet communication. Nevertheless, those risks and threats already exist in the own signaling process and, among them, the man-in-the-middle attack is the worst of all due to its high danger degree. After doing a bibliographical revision of the SIP security risks and threats, as well as its security mechanisms (analyzing their advantages and drawbacks), it was possible to generate a new security mechanism, which is presented in this work. The proposed mechanism uses a protocol for secure information exchange the Massey-Omura protocol which, when combined with bilinear pairings, provides a better security level for SIP in all its aspects (authenticity, privacy and integrity). Besides this, the new mechanism is evaluated by a proof of concept, in the which a functional SIP softphone was used. The security analysis and the results obtained from the proof of concept, make the proposed security mechanism a viable alternative for SIP.
46
Hussain, Intesab. "Solving flooding and SPIT based denial of service problems in voice over IP communications." Thesis, Paris 5, 2013. http://www.theses.fr/2013PA05S007.
Full textAbstract:
Pas de résumé en françaisSession Initiation Protocol (SIP) is the widely used signaling protocol for voiceand video communication as well as other multimedia applications. Despiteof its flexibility and a common standard that can be leveraged to efficientlycombine a wide array of communication systems and technologies, it is exposedto a number of problems, including the vulnerability to several types of attacksdue to its open nature, in particular, and lack of a clear defense line. Likewise,flooding attack is one of the most destructive attacks targeting both User AgentServer (UAS) and User Agent Client (UAC), leading to a Denial of Service (DoS)in VoIP applications. In particular, INVITE message is considered as one of themajor root causes of flooding attacks in SIP. This is due to the fact that an attackermay send numerous INVITE requests without waiting for responses from theUAS or proxy in order to exhaust their resources. Moreover, SPIT problem inSIP is also a challenging issue which needs proper attention and appropriatesolutions.Most of the solutions proposed to overcome the flooding attacks are eitherdifficult to deploy in practice or require significant changes in the SIP servers.Additionally, the diverse nature of flooding attacks offers a huge challenge toenvisage appropriate prevention mechanisms. In this survey, we present acomprehensive study on flooding attacks against SIP by addressing its differentvariants and analyzing its consequences. We also classify the existing solutionscorresponding to different flooding behaviors, types and targets, and then weperform an extensive investigation of their main weaknesses and strengths.Additionally, we also take into account the underlying assumptions of eachsolution for a better understanding of its limitations. Specifically, we havethoroughly analyzed SPIT problems and few of the existing solutions proposedfor their prevention.The theoretical framework derived from our extensive literature survey led us topropose a solution for handling specific number of SIP requests in a particulartime window. Our proposed "Light Weight Scheme" is implemented in a SERSIP server. The evaluation results presented in this thesis depict the satisfactoryperformance of this approach. In order to cope with SIP flooding attacks, wepropose another solution based on "Strategy Based Proxy". This solution isdesigned for a SIP proxy that calculates the probability of a call being maliciouson the basis of its current experience. The obtained experience is also utilized tocalculate the probabilities of a successful call setup. This approach is useful forboth state-ful and state-less proxy servers.For dealing with SPIT, we have designed a 2-step solution. In first step, weextract the useful information from the VoIP traffic. In second step, we apply aNaive Bayes classifier on the date extracted from first step to determine whetherthe nature of an incoming SIP call is malicious or it is a harmless routine call.With this mechanism, we can detect the SPIT calls from a group of incomingSIP calls. Finally, we presents a detailed discussion and conclusions derivedfrom our case study carried out in this thesis along with future directions andpotential research areas related to VoIP security threats
47
Silva, Davison Gonzaga da. "Implementação de um sistema SIP para o sistema operacional Linux." [s.n.], 2003. http://repositorio.unicamp.br/jspui/handle/REPOSIP/259317.
Full textAbstract:
Orientador: Leonardo de Souza MendesDissertação (mestrado) - Universidade Estadual de Campinas, Faculdade de Engenharia Eletrica e de Computação
Made available in DSpace on 2018-08-03T18:53:15Z (GMT). No. of bitstreams: 1 Silva_DavisonGonzagada_M.pdf: 885396 bytes, checksum: 09530eec1ff6b03c4cb01ae4e9f06123 (MD5) Previous issue date: 2003
Mestrado
48
Tanriverdi, Eda. "Simulation Based Investigation Of An Improvement For Faster Sip Re-registration." Master's thesis, METU, 2004. http://etd.lib.metu.edu.tr/upload/12605210/index.pdf.
Full textAbstract:
ABSTRACT
SIMULATION BASED INVESTIGATION OF AN IMPROVEMENT
FOR FASTER SIP RE-REGISTRATION
TANRIVERDi, Eda
M.Sc., Department of Electrical and Electronics Engineering
Supervisor: Prof. Dr. Semih BiLGEN
July 2004, 78 pages
In this thesis, the Session Initiation Protocol (SIP) is studied and an improvement
for faster re-registration is proposed. This proposal, namely the &ldquoregistration &ndash
activation&rdquo
, is investigated with a simulation prepared using OPNET. The literature about wireless mobile networks and SIP mobility is reviewed. Conditions for an effective mobile SIP network simulation are designed using message sequence charts. The testbed in [1] formed by Dutta et. al. that has been used to observe SIP handover performance is simulated and validated. The mobile nodes, SIP Proxy v servers, DHCP servers and network topology are simulated on &ldquo
OPNET Modeler Radio&rdquo
. Once the simulation is proven to be valid, the &ldquo
registration &ndash
activation&rdquo
is implemented. Different simulation scenarios are set up and run, with different mobile node speeds and different numbers of mobile nodes. The results show that the re-registration delay is improved by applying the &ldquo
registration &ndash
activation&rdquo
but the percentage of improvement depends on the improvement in the database access delay in the SIP Proxy server.
49
Bédard, Normand. "Sécurité d'une application de communication multimédia sous protocole IP dans un contexte médical." Mémoire, Université de Sherbrooke, 2010. http://savoirs.usherbrooke.ca/handle/11143/1530.
Full textAbstract:
L'application MédiclP est un logiciel de télémédecine permettant à des spécialistes de la santé d'entrer en communication lors de situations d'urgence. Ce prototype, développé depuis 2005, permet des communications audio ainsi que le transfert d'électrocardiogrammes en temps réel. Le scénario typique visé par ce projet était de permettre à une équipe ambulancière qui récupère un blessé sur la route, ou quelqu'un victime d'un malaise cardiaque, d'entrer en contact avec les hôpitaux les plus près afin de déterminer lequel est le plus apte à recevoir adéquatement ce patient. Cette approche permettrait d'améliorer la préparation, la qualité et la rapidité des opérations médicales à l'hôpital lorsque le patient se présente. Le présent projet, SécureMédic, se veut un moyen d'aborder le problème de la sécurité entourant ce prototype, étant donné son utilisation dans un contexte médical. Une analyse de MédicIP a permis d'identifier quatre failles de sécurité critiques reliées aux authentifications usagers, aux établissements des conférences, aux transferts des données audio ainsi qu'aux transferts des électrocardiogrammes. La contribution majeure de ce projet a été la création d'une infrastructure dédiée au processus d'authentification des usagers. Le système développé permet deux types d'authentification, fournissant ainsi un excellent niveau de robustesse. De plus, le serveur principal développé dans cette infrastructure intègre des mesures de protection permettant de minimiser les impacts de certains types d'attaque. Le projet SécureMédic a permis de démontrer la faisabilité de la sécurisation d'une application de télémédecine en utilisant les techniques de protection et les standards actuels de l'industrie. Les résultats de tests comparatifs ont cependant permis de constater que des impacts reliés à la performance ont été engendrés par l'ajout des mesures de sécurité, dus principalement aux ressources requises pour le chiffrement et le déchiffrement des données dans un environnement multimédia temps réel. Bien qu'il soit encore trop tôt pour envisager le déploiement du système actuel dans le milieu de la santé, les projets MédicIP et SécureMédic sont un pas dans la bonne direction. Le prototype actuel répond aux exigences techniques voulues, répond à un besoin bien réel et est parmi les premières solutions concrètes à démontrer la faisabilité d'un tel système. D'ici quelques années, l'apparition de solutions similaires est assurée.
50
Gianoto, Antonio Carlos. "O processo de migração de sistemas corporativos de comunicação TDM para plataformas convergentes IP com preservação de ativos." Universidade Presbiteriana Mackenzie, 2006. http://tede.mackenzie.br/jspui/handle/tede/2749.
Full textAbstract:
Made available in DSpace on 2016-04-18T21:39:45Z (GMT). No. of bitstreams: 4
Antonio Carlos GIANOTO1.pdf: 620239 bytes, checksum: aa779772dff23531ab717870d36823d3 (MD5)
Antonio Carlos GIANOTO2.pdf: 1522259 bytes, checksum: 928d1a566cc9977b69a13b358fe13a50 (MD5)
Antonio Carlos GIANOTO3.pdf: 2036577 bytes, checksum: 4ef7b886a0d20792d494deb085435041 (MD5)
Antonio Carlos GIANOTO4.pdf: 1519186 bytes, checksum: cce13633bbc1cf7d5714a8dd727dd008 (MD5)
Previous issue date: 2006-06-29Fundo Mackenzie de Pesquisa
The aim of this elaboration is to present a study of the migration process involved transforming the digital PBX (Private Branch Exchange), TDM (Time Division Multiplex), SPC (Stored Program Control) based platforms of corporate communications on technology to converged IP (Internet Protocol) systems supported by the TCP/IP (Transmission Control Protocol/Internet Protocol) protocol. This proposal analyzes the necessary interventions in order to preserve the investments made in these platforms, integrating them to existent data networks. Beside other benefits presented in this work, one key advantage is the possibility to transport voice over an existing data infrastructure, optimizing usage of carrier connections.
O objetivo desta dissertação é o de apresentar um estudo do processo de migração de plataformas de voz PABX (Private Automatic Branch Exchange) TDM (Time Division Multiplex) de comunicações corporativas baseadas na tecnologia CPA-T (Controle por Programa Armazenado estágio de comutação temporal digital), para sistemas convergentes suportados pelo protocolo TCP/IP (Transmission Control Protocol/Internet Protocol). São analisadas as intervenções necessárias para esta migração, preservando ao máximo os investimentos efetuados nestas plataformas, integrando-as as redes de dados existentes. Dentre outras vantagens apresentadas no texto, destaca-se a otimização dos acessos fornecidos pelas operadoras de telecomunicações pelo compartilhamento da infra-estrutura da rede de dados para o tráfego de sinais de voz.