385 research outputs found

    An Analysis of the Skype Peer-to-Peer Internet Telephony Protocol

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    Skype is a peer-to-peer VoIP client developed by KaZaa in 2003. Skype claims that it can work almost seamlessly across NATs and firewalls and has better voice quality than the MSN and Yahoo IM applications. It encrypts calls end-to-end, and stores user information in a decentralized fashion. Skype also supports instant messaging and conferencing. This report analyzes key Skype functions such as login, NAT and firewall traversal, call establishment, media transfer, codecs, and conferencing under three different network setups. Analysis is performed by careful study of Skype network traffic

    A Survey on Handover Management in Mobility Architectures

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    This work presents a comprehensive and structured taxonomy of available techniques for managing the handover process in mobility architectures. Representative works from the existing literature have been divided into appropriate categories, based on their ability to support horizontal handovers, vertical handovers and multihoming. We describe approaches designed to work on the current Internet (i.e. IPv4-based networks), as well as those that have been devised for the "future" Internet (e.g. IPv6-based networks and extensions). Quantitative measures and qualitative indicators are also presented and used to evaluate and compare the examined approaches. This critical review provides some valuable guidelines and suggestions for designing and developing mobility architectures, including some practical expedients (e.g. those required in the current Internet environment), aimed to cope with the presence of NAT/firewalls and to provide support to legacy systems and several communication protocols working at the application layer

    Security aspects in voice over IP systems

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    Security has become a major concern with the rapid growth of interest in the internet. This project deals with the security aspects of VoIP systems. Various supporting protocols and technologies are considered to provide solutions to the security problems. This project stresses on the underlying VoIP protocols like Session Initiation Protocol (SIP), Secure Real-time Transport Procotol (SRTP), H.323 and Media Gateway Control Protocol (MGCP). The project further discusses the Network Address Translation (NAT) devices and firewalls that perform NAT. A firewall provides a point of defense between two networks. This project considers issues regarding the firewalls and the problems faced in using firewalls for VoIP; it further discusses the solutions about how firewalls can be used in a more secured way and how they provide security

    Detecting and Mitigating Denial-of-Service Attacks on Voice over IP Networks

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    Voice over IP (VoIP) is more susceptible to Denial of Service attacks than traditional data traffic, due to the former's low tolerance to delay and jitter. We describe the design of our VoIP Vulnerability Assessment Tool (VVAT) with which we demonstrate vulnerabilities to DoS attacks inherent in many of the popular VoIP applications available today. In our threat model we assume an adversary who is not a network administrator, nor has direct control of the channel and key VoIP elements. His aim is to degrade his victim's QoS without giving away his presence by making his attack look like a normal network degradation. Even black-boxed, applications like Skype that use proprietary protocols show poor performance under specially crafted DoS attacks to its media stream. Finally we show how securing Skype relays not only preserves many of its useful features such as seamless traversal of firewalls but also protects its users from DoS attacks such as recording of conversations and disruption of voice quality. We also present our experiences using virtualization to protect VoIP applications from 'insider attacks'. Our contribution is two fold we: 1) Outline a threat model for VoIP, incorporating our attack models in an open-source network simulator/emulator allowing VoIP vendors to check their software for vulnerabilities in a controlled environment before releasing it. 2) We present two promising approaches for protecting the confidentiality, availability and authentication of VoIP Services

    Past, present and future of IP telephony

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    “Copyright © [2008] IEEE. Reprinted from International Conference on Communication Theory, Reliability, and Quality of Service, 2008. CTRQ '08. ISBN:978-0-7695-3190-8. This material is posted here with permission of the IEEE. Internal or personal use of this material is permitted. However, permission to reprint/republish this material for advertising or promotional purposes or for creating new collective works for resale or redistribution must be obtained from the IEEE by writing to [email protected]. By choosing to view this document, you agree to all provisions of the copyright laws protecting it.”Since the late 90's IP telephony, commonly referred to as Voice over IP (VoIP), has been presented as a revolution on communications enabling the possibility to converge historically separated voice and data networks, reducing costs, and integrating voice, data and video on applications. This paper presents a study over the standard VoIP protocols H.323, Session Initiation Protocol (SIP), Media Gateway Control Protocol (MGCP), and H.248/Megaco. Given the fact that H.323 and SIP are more widespread than the others, we focus our study on them. For each of these protocols we describe and discuss its main capabilities, architecture, stack protocol, and characteristics. We also briefly point their technical limitations. Furthermore, we present the Advanced Multimedia System (AMS) project, a new system that aims to operate on Next Generation Networks (NGN) taking the advantage of its features, and it is viewed as the successor to H.323 and SIP
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