Covert Channels in SIP for VoIP signalling

In this paper, we evaluate available steganographic techniques for SIP (Session Initiation Protocol) that can be used for creating covert channels during signaling phase of VoIP (Voice over IP) call. Apart from characterizing existing steganographic methods we provide new insights by introducing new techniques. We also estimate amount of data that can be transferred in signalling messages for typical IP telephony call.Comment: 8 pages, 4 figure

Statistical Analysis and Modeling of SIP Traffic for Parameter Estimation of Server Hysteretic Overload Control, Journal of Telecommunications and Information Technology, 2013, nr 4

The problem of overload control in Session Initiation Protocol (SIP) signaling networks gives rise to many questions which attract researchers from theoretical and practical point of view. Any mechanism that is claimed to settle this problem down demands estimation of local (control) parameters on which its performance is greatly dependent. In hysteretic mechanism these parameters are those which define hysteretic loops. In order to find appropriate values for parameters one needs adequate model of SIP traffic flow circulating in the network under consideration. In this paper the attempt is made to address this issue. Analysis of SIP traffic collected from telecommunication operator’s network is presented. Traffic profile is built. It is shown that fitting with Markov Modulated Poisson Process with more than 2 phases is accurate. Estimated values of its parameters are given

Гистерезисное управление сигнальной нагрузкой в сети SIP-серверов

This review deals with the research of load control mechanisms in signaling networks that use three types of thresholds for congestion control. The main objective of this paper is toanalyze the congestion control mechanisms and mathematical models for SIP-servers. Thestudy is based on hysteresial techniques of flow control, which originally was developed forSignaling System 7. We propose general methods for describing hysteresis signaling flow control techniques. We study the current situation and problems of SIP built-in overloadcontrol mechanism, proposed by IETF. Our approaches to mathematical models constructionin the form of queuing systems with hysteresis control are presented.В статье, являющейся по сути обзором, исследуются механизмы управления нагрузкой в сетях сигнализации, которые используют три типа порогов для контроля перегрузок. Целью обзора является анализ механизмов и моделей контроля перегрузок SIP-серверов. В основе исследований лежит гистерезисное управление нагрузкой, которое исходно было разработано для общеканальной системы сигнализации №7. Разработаны унифицированные методы описания процедур гистерезисного управления сигнальной нагрузкой. Исследовано современное состояние и проблемы базового механизма контроля перегрузок SIP-серверов, предложенного комитетом IETF. Изложены подходы к построению математических моделей SIP-серверов в виде систем массового обслуживания с гистерезисным управлением

Design and implementation of SIP VoIP Adapter

Projecte fet en col.laboració amb École Central d'Eléctronique de ParisThe SIP VoIP Adapter is a Java application that is able to establish a SIP communication acting as a User Agent, which uses an external device as a sound device, to play and acquire the audio from the call established through Asterisk PBX. The SIP VoIP Adapter has been also implemented to be able to use and independent audio data codecs for both audio communications, the RTP communications between SIP clients, and the UDP communication between the SIP VoIP Adapter and the 6lowPAN end device.

Test generation algorithm for the All-Transition-State criteria of Finite State Machines

In the current article a novel test generation algorithm is presented for deterministic finite state machine specifications based on the recently introduced All-Transition-State criteria. The size of the resulting test suite and the time required for test suite generation are investigated through analytical and practical analyses and are also compared to the Transition Tour, Harmonized State Identifiers and random walk test generation methods. The fault detection capabilities of the different approaches are also investigated with simulations applying randomly injected transfer faults

Μελέτη προηγμένων τηλεπικοινωνιακών συστημάτων με πλατφόρμες ανοικτού λογισμικού

Σκοπός της παρούσης διπλωματικής εργασίας είναι η μελέτη συστημάτων κινητής τηλεφωνίας και η υλοποίησή τους με πλατφόρμες Software Defined Radio και προγράμματα ανοιχτού λογισμικού. Εξετάζονται τα συστήματα GSM/GPRS/UMTS/LTE, προτεινόμενες κυματομορφές των δικτύων νέας γενιάς 5G και υλοποιείται η εφαρμογή τους με συστήματα ανοιχτού λογισμικού σε πλατφόρμες SDR.The purpose of the current thesis is to study advanced mobile telecommunication systems and their implementation using Software Defined Radio platforms and open source software. The thesis examines the GSM/GPRS/UMTS/LTE systems, suggested waveforms for the upcoming 5G new technologies and their implementation using open source software and SDR platforms

Sistema de monitoramento de qualidade em serviços de telefonia IP

Dissertação (mestrado) - Universidade Federal de Santa Catarina, Centro Tecnológico, Programa de Pós-Graduação em Ciência da Computação, Florianópolis, 2015.A telefonia IP é um serviço consolidado que tem crescido de forma constante, impulsionada por suas diversas vantagens, como redução de custos, facilidade de integração com outros serviços, dentre muitas outras. No entanto, manter a qualidade deste serviço ainda é um desafio principalmente em locais de redes congestionadas. Como a telefonia é um serviço essencial para várias organizações, é fundamental manter as chamadas em níveis razoáveis de qualidade. Para tal, a qualidade oferecida pela telefonia IP deve ser constantemente monitorada de maneira a orientar as ações de novos investimentos e manutenção. Nesta direção, a presente dissertação propõe um sistema de monitoramento da qualidade fim-a-fim para os serviços de telefonia IP com base no pacote de relatórios de qualidade RTCP XR (Real Time Control Protocol - Extended Reports) e no protocolo de sinalização SIP (Session Initiation Protocol). Um caso de uso do sistema proposto, em um serviço de produção de telefonia IP de uma universidade, utilizando métricas objetivas e metodologias de monitoramento não intrusivas, demonstrou a efetividade e versatilidade do sistema proposto.Abstract : IP telephony is a consolidated service that has been growing steadily, driven by its various advantages, such as cost reduction, ease of integration with other services, among many others. However, maintaining the quality of this service is still a challenge especially in places of congested networks. As the telephony is an essential service for various organizations, it is essential to maintain the call quality at reasonable levels. To this end, the quality offered by IP telephony should be constantly monitored in order to guide the actions of new investments and maintenance. In this direction, this thesis proposes a quality end-to-end monitoring system for IP telephony services based on quality reporting package RTCP XR (Real Time Control Protocol - Extended Reports) and SIP signaling protocol (Session Initiation Protocol). A use case of the proposed system in an IP telephony production service of a university, using objective metrics and non-intrusive monitoring methodologies, demonstrated the effectiveness and versatility of the proposed system