The Virtual Device: Expanding Wireless Communication Services Through Service Discovery and Session Mobility

We present a location-based, ubiquitous service architecture, based on the Session Initiation Protocol (SIP) and a service discovery protocol that enables users to enhance the multimedia communications services available on their mobile devices by discovering other local devices, and including them in their active sessions, creating a 'virtual device.' We have implemented our concept based on Columbia University's multimedia environment and we show its feasibility by a performance analysis

Towards a scalable video interactivity solution over the IMS

Includes bibliographical references (leaves 72-76).Rapid increase in bandwidth and the interactive and scalability features of the Internet provide a precedent for a converged platform that will support interactive television. Next Generation Network platforms such as the IP Multimedia Subsystem (IMS) support Quality of Service (QoS), fair charging and possible integration with other services for the deployment of IPTV services. IMS architecture supports the use of the Session Initiation Protocol (SIP) for session control and the Real Time Streaming Protocol (RTSP) for media control. This study aims to investigate video interactivity designs over the Internet using an evaluation framework to examine the performance of both SIP and RTSP protocols over the IMS over different access networks. It proposes a Three Layered Video Interactivity Framework (TLVIF) to reduce the video processing load on a server

Unified Messaging using SIP and RTSP

Traditional answering machines and voice mail services are closed systems, tightly coupled to a single end system,the local PBX or local exchange carrier. Even simple services, such as forwarding voice mail to another user outside the local system, are hard to provide. With the advent of Internet telephony, we need to provide voice and video mail services. This also offers the opportunity to address some of the shortcomings of existing voice mail systems. We list general requirements for a multimedia mail system for Internet telephony. We then propose an architecture using SIP (Session Initiation Protocol) and RTSP (Real-Time Streaming Protocol) and compare various alternative approaches to solving call forwarding, reclaiming and retrieval of messages. We also briefly describe our prototype implementation

Efficient User Controlled Inter-Domain SIP Mobility: Authentication, Registration, and Call Routing

Over the past decade, multimedia services have gained significant acceptance and played an important role in the convergence of IP networks. Supporting mobility in IP (Internet Protocol) networks is a crucial step towards satisfying the nomadic communication paradigms on the current Internet. The Session Initiation Protocol (SIP) presents one approach towards supporting IP mobility. Additionally, SIP is increasingly gaining in popularity as the next generation multimedia signaling and session establishment protocol. It is anticipated that the SIP infrastructure will be extensively deployed all over the Internet. In this paper, we explore an efficient approach to inter-domain SIP mobility in an attempt to improve personal and terminal mobility schemes. We succeed in applying a persistent identification framework to application level SIP addressing by introducing a level of indirection on top of the traditional SIP architecture. We refer to our approach as the Handle SIP (H-SIP). H-SIP leverages the current SIP architecture abstracting any domain binding from users. Our approach to mobility is user-controlled. We experimentally prove the efficiency of H-SIP in achieving inter-domain authentication and call routing through modeling and real-time measurements

An ICT-oriented Management Solution for NGNs

NGN architecture reused several standards from the IP world, as exemplified by the Session Initiation Protocol SIP, which is ubiquitous in the majority of these network components. However, the NGN management architecture simply presented a very generic management model that follows TMN. Several management technologies are proposed, such as Web services, CORBA and SNMP, to implement management solutions. Network and systems management standardizing bodies currently promote newer technologies that aim to solve known shortcomings to these. This paper proposes a management solution for NGNs based on recent IP world technologies. The presented solution was implemented in the form of a middleware to manage NGN elements. This middleware was used in the management of an element belonging to the IP Multimedia Subsystem platform, namely the Policy and Charging Rules Function

Representing New Voice Services and Their Features

New voice services are investigated in the fields of Internet telephony (SIP – Session Initiation Protocol) and interactive voice systems (VoiceXML – Voice Extended Markup Language). It is explained how CRESS (Chisel Representation Employing Systematic Specification) can graphically represent services and features in these domains. CRESS is a front-end for detecting feature interactions and for implementing features. The nature of service architecture and feature composition are presented. CRESS descriptions are automatically compiled into LOTOS (Language Of Temporal Ordering Specification) and SDL (Specification and Description Language), allowing automated analysis of service behaviour and feature interaction. For implementation, CRESS diagrams can be compiled into Perl (for SIP) and VoiceXML. The approach combines the benefits of an accessible graphical notation, underlying formalisms, and practical realisation

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

Security in peer-to-peer communication systems

P2PSIP (Peer-to-Peer Session Initiation Protocol) is a protocol developed by the IETF (Internet Engineering Task Force) for the establishment, completion and modi¿cation of communication sessions that emerges as a complement to SIP (Session Initiation Protocol) in environments where the original SIP protocol may fail for technical, ¿nancial, security, or social reasons. In order to do so, P2PSIP systems replace all the architecture of servers of the original SIP systems used for the registration and location of users, by a structured P2P network that distributes these functions among all the user agents that are part of the system. This new architecture, as with any emerging system, presents a completely new security problematic which analysis, subject of this thesis, is of crucial importance for its secure development and future standardization. Starting with a study of the state of the art in network security and continuing with more speci¿c systems such as SIP and P2P, we identify the most important security services within the architecture of a P2PSIP communication system: access control, bootstrap, routing, storage and communication. Once the security services have been identi¿ed, we conduct an analysis of the attacks that can a¿ect each of them, as well as a study of the existing countermeasures that can be used to prevent or mitigate these attacks. Based on the presented attacks and the weaknesses found in the existing measures to prevent them, we design speci¿c solutions to improve the security of P2PSIP communication systems. To this end, we focus on the service that stands as the cornerstone of P2PSIP communication systems¿ security: access control. Among the new designed solutions stand out: a certi¿cation model based on the segregation of the identity of users and nodes, a model for secure access control for on-the-¿y P2PSIP systems and an authorization framework for P2PSIP systems built on the recently published Internet Attribute Certi¿cate Pro¿le for Authorization. Finally, based on the existing measures and the new solutions designed, we de¿ne a set of security recommendations that should be considered for the design, implementation and maintenance of P2PSIP communication systems.Postprint (published version

Mobility Management in beyond 3G-Environments

Beyond 3G-environments are typically defined as environments that integrate different wireless and fixed access network technologies. In this paper, we address IP based Mobility Management (MM) in beyond 3G-environments with a focus on wireless access networks, motivated by the current trend of WiFi, GPRS, and UMTS networks. The GPRS and UMTS networks provide countrywide network access, while the WiFi networks provide network access in local areas such as city centres and airports. As a result, mobile end-users can be always on-line and connected to their preferred network(s), these network preferences are typically stored in a user profile. For example, an end-user who wishes to be connected with highest bandwidth could be connected to a WiFi network when available and fall back to GPRS when moving outside the hotspot area.\ud In this paper, we consider a combination of MM for legacy services (like web browsing, telnet, etc.) using Mobile IP and multimedia services using SIP. We assume that the end-user makes use of multi-interface terminals with the capability of selecting one or more types of access networks\ud based on preferences. For multimedia sessions, like VoIP or streaming video, we distinguish between changes in network access when the end-user is in a session or not in a session. If the end-user is not in a session, he or she needs to be able to start new sessions and receive invitations for new sessions. If the end-user is in a session, the session needs to be handed over to the new access network as seamless as possible from the perspective of the end-user. We propose an integrated but flexible solution to these problems that facilitates MM with a customizable transparency to applications and end-users