131 resultados para VoIP
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The use of wireless local area networks, called WLANs, as well as the proliferation of the use of multimedia applications have grown rapidly in recent years. Some factors affect the quality of service (QoS) received by the user and interference is one of them. This work presents strategies for planning and performance evaluation through an empirical study of the QoS parameters of a voice over Internet Protocol (VoIP) application in an interference network, as well as the relevance in the design of wireless networks to determine the coverage area of an access point, taking into account several parameters such as power, jitter, packet loss, delay, and PMOS. Another strategy is based on a hybrid approach that considers measuring and Bayesian inference applied to wireless networks, taking into consideration QoS parameters. The models take into account a cross layer vision of networks, correlating aspects of the physical environment, on the signal propagation (power or distance) with aspects of VoIP applications (e.g., jitter and packet loss). Case studies were carried out for two indoor environments and two outdoor environments, one of them displaying main characteristics of the Amazon region (e.g., densely arboreous environments). This last test bed was carried out in a real system because the Government of the State of Pará has a digital inclusion program called NAVEGAPARÁ.
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Coordenação de Aperfeiçoamento de Pessoal de Nível Superior (CAPES)
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Pós-graduação em Ciência da Informação - FFC
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Teletandem is a virtual autonomous VoIP2 technology-based context (webcam images, voice and text). Within this context, two students help each other learn their native (or other) language through intercultural and linguistic collaboration. Performative Theory can shed light on the constitution of these students' national identities, as they engage in linguistic performances of marking and discussing differences between their countries during teletandem. Based on critical approaches to discourse and intercultural communication, my analysis shows that this online intercultural contact opens innovative possibilities for foreign language teachers to promote intercultural contact with the different. However, without teacher mediation, teletandem interactions may fall into shallow performances of sedimented and pre-given representations of self and other. Subsequently, this article concludes with a discussion of relevant pedagogical points for foreign language teachers.
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The main objective is to analyze the performance of some codecs supported by Asterisk with and without encryption using RTP and SRTP, respectively, providing important data for decision-making in the implementation of a VoIP system with Asterisk. Thus, it is possible to realize both codecs as the protocol can be chosen depending on the application, or the system's main feature is the speed packet switching, security level or lower tolerance for unsuccessful calls. For this, tests were made with the codec with and without the use of cryptography to obtain some findings on the use of the same, giving more attention to the response time for the start of a call.
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Pós-graduação em Estudos Linguísticos - IBILCE
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Pós-graduação em Estudos Linguísticos - IBILCE
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Skype is one of the well-known applications that has guided the evolution of real-time video streaming and has become one of the most used software in everyday life. It provides VoIP audio/video calls as well as messaging chat and file transfer. Many versions are available covering all the principal operating systems like Windows, Macintosh and Linux but also mobile systems. Voice quality decreed Skype success since its birth in 2003 and peer-to-peer architecture has allowed worldwide diffusion. After video call introduction in 2006 Skype became a complete solution to communicate between two or more people. As a primarily video conferencing application, Skype assumes certain characteristics of the delivered video to optimize its perceived quality. However in the last years, and with the recent release of SkypeKit1, many new Skype video-enabled devices came out especially in the mobile world. This forced a change to the traditional recording, streaming and receiving settings allowing for a wide range of network and content dynamics. Video calls are not anymore based on static ‘chatting’ but mobile devices have opened new possibilities and can be used in several scenarios. For instance, lecture streaming or one-to-one mobile video conferences exhibit more dynamics as both caller and callee might be on move. Most of these cases are different from “head&shoulder” only content. Therefore, Skype needs to optimize its video streaming engine to cover more video types. Heterogeneous connections require different behaviors and solutions and Skype must face with this variety to maintain a certain quality independently from connection used. Part of the present work will be focused on analyzing Skype behavior depending on video content. Since Skype protocol is proprietary most of the studies so far have tried to characterize its traffic and to reverse engineer its protocol. However, questions related to the behavior of Skype, especially on quality as perceived by users, remain unanswered. We will study Skype video codecs capabilities and video quality assessment. Another motivation of our work is the design of a mechanism that estimates the perceived cost of network conditions on Skype video delivery. To this extent we will try to assess in an objective way the impact of network impairments on the perceived quality of a Skype video call. Traditional video streaming schemes lack the necessary flexibility and adaptivity that Skype tries to achieve at the edge of a network. Our contribution will lye on a testbed and consequent objective video quality analysis that we will carry out on input videos. We will stream raw video files with Skype via an impaired channel and then we will record it at the receiver side to analyze with objective quality of experience metrics.
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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.