Usage control in SIP-based multimedia delivery

The Session Initiation Protocol (SIP) is an application layer signaling protocol for the creation, modification and termination of multimedia sessions and VoIP calls with one or more participants.SIP is widely accepted as the protocol that will dominate multimedia communications in the future and one of the reasons is that it can inherently support multidomain heterogeneous networks.While SIP operates in highly dynamic environments, in the current version its authorization support is based on traditional access control models.The main problem these models face is that they were designed many years ago, and under some circumstances tend to be inadequate in modern highly dynamic environments.Usage Control (UCON), instead, is a model that supports the same operations as traditional access control models do, but it further enhances them with novel ones.In previous work, an architecture supporting continuous authorizations on SIP, based on the UCON model, was presented.In this paper, an authorization support implementing the whole UCON model, including authorizations, obligations and conditions, has been integrated in a SIP system.Moreover, a testbed has been set up to experimentally evaluate the performance of the proposed security mechanism

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

XMPP and iDTV or how to make television a social medium

Instant Messaging (IM) has the potential to become one of the killer applications for interactive Digital Television (iDTV) [12]. However, several factors make it difficult to provide a good implementation of IM services, among which the limited resources of a set-top box and the different user experience compared to computer environments. This paper proposes the XMPP (Extensible Messaging and Presence Protocol) standard as a solution for implementing IM. When we compare XMPP with other technologies, it reveals itself to be very well adapted to the specific needs of iDTV middleware platforms like the Multimedia Home Platform (MHP) [5]. Moreover, the use of XMPP doesn't limit itself to IM The flexible architecture of XMPP opens a window of opportunities like the ease of adding new interactive services. To demonstrate the possibilities of XMPP on MHP, an IM client -IM4MHP- is presented in this paper

Delivering Live Multimedia Streams to Mobile Hosts in a Wireless Internet with Multiple Content Aggregators

We consider the distribution of channels of live multimedia content (e.g., radio or TV broadcasts) via multiple content aggregators. In our work, an aggregator receives channels from content sources and redistributes them to a potentially large number of mobile hosts. Each aggregator can offer a channel in various configurations to cater for different wireless links, mobile hosts, and user preferences. As a result, a mobile host can generally choose from different configurations of the same channel offered by multiple alternative aggregators, which may be available through different interfaces (e.g., in a hotspot). A mobile host may need to handoff to another aggregator once it receives a channel. To prevent service disruption, a mobile host may for instance need to handoff to another aggregator when it leaves the subnets that make up its current aggregator�s service area (e.g., a hotspot or a cellular network).\ud In this paper, we present the design of a system that enables (multi-homed) mobile hosts to seamlessly handoff from one aggregator to another so that they can continue to receive a channel wherever they go. We concentrate on handoffs between aggregators as a result of a mobile host crossing a subnet boundary. As part of the system, we discuss a lightweight application-level protocol that enables mobile hosts to select the aggregator that provides the �best� configuration of a channel. The protocol comes into play when a mobile host begins to receive a channel and when it crosses a subnet boundary while receiving the channel. We show how our protocol can be implemented using the standard IETF session control and description protocols SIP and SDP. The implementation combines SIP and SDP�s offer-answer model in a novel way