Study, Analysis and Modeling of IP Multimedia systems on next generation networks providing mobile and fixed multimedia services

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Security aspects in voice over IP systems

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

A method for forensic artifact collection, analysis and incident response in environments running Session Initiation Protocol (SIP) and Session Description protocol

In this paper, we perform an analysis of SIP, a popular voice over IP (VoIP) protocol and propose a framework for capturing and analysing volatile VoIP data in order to determine forensic readiness requirements for effectively identifying an attacker. The analysis was performed on real attack data and the findings were encouraging. It seems that if appropriate forensic readiness processes and controls are in place, a wealth of evidence can be obtained. The type of the end user equipment of the internal users, the private IP, the software that is used can help build a reliable baseline information database. On the other hand the private IP addresses of the potential attacker even during the presence of NAT services, as well as and the attack tools employed by the malicious parties are logged for further analysis

Options for Securing RTP Sessions

The Real-time Transport Protocol (RTP) is used in a large number of different application domains and environments. This heterogeneity implies that different security mechanisms are needed to provide services such as confidentiality, integrity, and source authentication of RTP and RTP Control Protocol (RTCP) packets suitable for the various environments. The range of solutions makes it difficult for RTP-based application developers to pick the most suitable mechanism. This document provides an overview of a number of security solutions for RTP and gives guidance for developers on how to choose the appropriate security mechanism

The generic context sharing protocol GCSP : Application to signaling in a cross-network and multi-provider environment

This paper proposes a new signaling paradigm and a new signaling protocol called the Generic Context Sharing Protocol (GCSP) for the construction of a global control plane over present and future communication networks. After identifying the special nature of the control plane software involved in the setup of a conversational service instance it examines the various mechanisms for information sharing which leads to our new proposal. We show that this new data-based protocol is better suited to control plane requirements than the present day’s command-oriented signaling mechanisms. We indicate the basic principles of the protocol and we give a brief description of the generic context. We show the place of this proposal in the present day research efforts and we mention a practical implementation case.8th IFIP/IEEE International conference on Mobile and Wireless CommunicationRed de Universidades con Carreras en Informática (RedUNCI

SNOM provisioning

The Internet was not designed in the beginning, to be used as a real-time transportation medium. Today, it is used to send any kind of service. My project is to add an application to AskoziaPBX that provisions SNOM phones. I use the protocol as a transport medium for creating, sending, and termination of communication with the SNOM phone is SIP protocol. Asterisk is a free software application that provides the functionality of a PBX telephone exchange. Like any PBX, you can connect a number of phone calls to each other and even connect to VoIP providers. The most interesting aspect of Asterisk is that it provides many VoIP protocols such as SIP, H.323, IAX, etc. AskoziaPBX is a very comprehensive system based on Asterisk. My job is to automate the provisioning of a SNOM 360 phone (telephones being used for this project) via SIP multicast. We can say that SIP is today one of the safer and better protocols for VoIP. Finally, I should add AskoziaPBX with my applications The purpose of this application is to make it simpler and faster for users to configure their phones.Ingeniería Técnica en Telemátic

DCCP Simultaneous-Open Technique to Facilitate NAT/Middlebox Traversal

https://datatracker.ietf.org/doc/rfc5595/Publisher PD

A novel mechanism for anonymizing Global System for Mobile Communications calls using a resource-based Session Initiation Protocol community network

Considering the widespread adoption of smartphones in mobile communications and the well-established resource sharing use in the networking community, we present a novel mechanism to achieve anonymity in the Global System for Mobile Communications (GSM). We propose a Voice over Internet Protocol infrastructure using the Session Initiation Protocol (SIP) where a smartphone registers on a SIP registrar and can start GSM conversation through another smartphone acting as a GSM gateway, by using a SIP intermediate without an extra cost. The testbed that we developed for empirical evaluation revealed no significant quality of service degradation