A new scheme to reduce session establishment time in session initiation protocol (SIP)

The session Initiation Protocol (SIP) has been developed by Internet Engineering Taskforce standard (IETF) with the main purpose of establishing and managing sessions between two or more parties wishing to communicate. SIP is a signaling protocol which is used for the current and future Internet Protocol (IP) telephony services, video services, and integrated web and multimedia services. SIP is an application layer protcol, thus it can run over Transmission Control Protocol(TCP) or User Datagram Protocol (UDP). When the packets are sent over the network, a form of congestion control mechanism is necessary to prevent from network collapse. TCP is a reliable protocl and provides the congestion control by adjusting the size of the congestion windows. UDP is an unreliable protocol and no flow control mechanism is provided. Many applications of the Internet require the establishment and management of sessions. The purpose of the thesis is to study the session establishnment procedure in SIP and try to reduce the time taken for the session setup in two different conditions. One, when there is no congestion in the network, and the other is when there is a network congestion. We have simulated the behaviour of session establishment in SIP using Network Simulator (NS2). UDP is used as the transport protocol. We have created different network topologies. In the topology we had created SIP user agents who wants to communicte, proxy servers for forwarding the requests on behalf of the user agents, and a Domain Name Server (DNS) which maintains the location information of all proxy servers. We tried to reduce the time taken for the session establishment. As UDP does not provide any congestion control mechanisms, we used the binary exponential backoff (BEB) algorithm to set the timers. In our network topolgy when there is no packet loss in the network, the time taken for the session establishment is reduced from 0.86 sec to 0.574 sec. In case of network congestion the setup time is reduced from 4.55 sec to 2.86 sec. From the simulation, we conclude that the session establishment time can be reduced by reducing the number of message exchanges required for session setup

Mobility and Handoff Management in Wireless Networks

With the increasing demands for new data and real-time services, wireless networks should support calls with different traffic characteristics and different Quality of Service (QoS)guarantees. In addition, various wireless technologies and networks exist currently that can satisfy different needs and requirements of mobile users. Since these different wireless networks act as complementary to each other in terms of their capabilities and suitability for different applications, integration of these networks will enable the mobile users to be always connected to the best available access network depending on their requirements. This integration of heterogeneous networks will, however, lead to heterogeneities in access technologies and network protocols. To meet the requirements of mobile users under this heterogeneous environment, a common infrastructure to interconnect multiple access networks will be needed. In this chapter, the design issues of a number of mobility management schemes have been presented. Each of these schemes utilizes IP-based technologies to enable efficient roaming in heterogeneous network. Efficient handoff mechanisms are essential for ensuring seamless connectivity and uninterrupted service delivery. A number of handoff schemes in a heterogeneous networking environment are also presented in this chapter.Comment: 28 pages, 11 figure

Roaming Real-Time Applications - Mobility Services in IPv6 Networks

Emerging mobility standards within the next generation Internet Protocol, IPv6, promise to continuously operate devices roaming between IP networks. Associated with the paradigm of ubiquitous computing and communication, network technology is on the spot to deliver voice and videoconferencing as a standard internet solution. However, current roaming procedures are too slow, to remain seamless for real-time applications. Multicast mobility still waits for a convincing design. This paper investigates the temporal behaviour of mobile IPv6 with dedicated focus on topological impacts. Extending the hierarchical mobile IPv6 approach we suggest protocol improvements for a continuous handover, which may serve bidirectional multicast communication, as well. Along this line a multicast mobility concept is introduced as a service for clients and sources, as they are of dedicated importance in multipoint conferencing applications. The mechanisms introduced do not rely on assumptions of any specific multicast routing protocol in use.Comment: 15 pages, 5 figure

Signaling for Internet Telephony

Internet telephony must offer the standard telephony services.However, the transition to Internet-based telephony services also provides an opportunity to create new services more rapidly and with lower complexity than in the existing public switched telephone network(PSTN). The Session Initiation Protocol (SIP) is a signaling protocol that creates, modifies and terminates associations between Internet end systems, including conferences and point-to-point calls. SIP supports unicast, mesh and multicast conferences, as well as combinations of these modes. SIP implements services such as call forwarding and transfer, placing calls on hold, camp-on and call queueing by a small set of call handling primitives. SIP implementations can re-use parts of other Internet service protocols such as HTTP and the Real-Time Stream Protocol (RTSP). In this paper, we describe SIP, and show how its basic primitives can be used to construct a wide range of telephony services

An interoperable and secure architecture for internet-scale decentralized personal communication

Interpersonal network communications, including Voice over IP (VoIP) and Instant Messaging (IM), are increasingly popular communications tools. However, systems to date have generally adopted a client-server model, requiring complex centralized infrastructure, or have not adhered to any VoIP or IM standard. Many deployment scenarios either require no central equipment, or due to unique properties of the deployment, are limited or rendered unattractive by central servers. to address these scenarios, we present a solution based on the Session Initiation Protocol (SIP) standard, utilizing a decentralized Peer-to-Peer (P2P) mechanism to distribute data. Our new approach, P2PSIP, enables users to communicate with minimal or no centralized servers, while providing secure, real-time, authenticated communications comparable in security and performance to centralized solutions.;We present two complete protocol descriptions and system designs. The first, the SOSIMPLE/dSIP protocol, is a P2P-over-SIP solution, utilizing SIP both for the transport of P2P messages and personal communications, yielding an interoperable, single-stack solution for P2P communications. The RELOAD protocol is a binary P2P protocol, designed for use in a SIP-using-P2P architecture where an existing SIP application is modified to use an additional, binary RELOAD stack to distribute user information without need for a central server.;To meet the unique security needs of a fully decentralized communications system, we propose an enrollment-time certificate authority model that provides asserted identity and strong P2P and user-level security. In this model, a centralized server is contacted only at enrollment time. No run-time connections to the servers are required.;Additionally, we show that traditional P2P message routing mechanisms are inappropriate for P2PSIP. The existing mechanisms are generally optimized for file sharing and neglect critical practical elements of the open Internet --- namely link-level security and asymmetric connectivity caused by Network Address Translators (NATs). In response to these shortcomings, we introduce a new message routing paradigm, Adaptive Routing (AR), and using both analytical models and simulation show that AR significantly improves message routing performance for P2PSIP systems.;Our work has led to the creation of a new research topic within the P2P and interpersonal communications communities, P2PSIP. Our seminal publications have provided the impetus for subsequent P2PSIP publications, for the listing of P2PSIP as a topic in conference calls for papers, and for the formation of a new working group in the Internet Engineering Task Force (IETF), directed to develop an open Internet standard for P2PSIP

A Survey of Satellite Communications System Vulnerabilities

The U.S. military’s increasing reliance on commercial and military communications satellites to enable widely-dispersed, mobile forces to communicate makes these space assets increasingly vulnerable to attack by adversaries. Attacks on these satellites could cause military communications to become unavailable at critical moments during a conflict. This research dissected a typical satellite communications system in order to provide an understanding of the possible attacker entry points into the system, to determine the vulnerabilities associated with each of these access points, and to analyze the possible impacts of these vulnerabilities to U.S. military operations. By understanding these vulnerabilities of U.S. communications satellite systems, methods can be developed to mitigate these threats and protect future systems. This research concluded that the satellite antenna is the most vulnerable component of the satellite communications system’s space segment. The antenna makes the satellite vulnerable to intentional attacks such as: RF jamming, spoofing, meaconing, and deliberate physical attack. The most vulnerable Earth segment component was found to be the Earth station network, which incorporates both Earth station and NOC vulnerabilities. Earth segment vulnerabilities include RF jamming, deliberate physical attack, and Internet connection vulnerabilities. The most vulnerable user segment components were found to be the SSPs and PoPs. SSPs are subject to the vulnerabilities of the services offered, the vulnerabilities of Internet connectivity, and the vulnerabilities associated with operating the VSAT central hub. PoPs are susceptible to the vulnerabilities of the PoP routers, the vulnerabilities of Internet and Intranet connectivity, and the vulnerabilities associated with cellular network access

Optimizing IETF multimedia signaling protocols and architectures in 3GPP networks : an evolutionary approach

Signaling in Next Generation IP-based networks heavily relies in the family of multimedia signaling protocols defined by IETF. Two of these signaling protocols are RTSP and SIP, which are text-based, client-server, request-response signaling protocols aimed at enabling multimedia sessions over IP networks. RTSP was conceived to set up streaming sessions from a Content / Streaming Server to a Streaming Client, while SIP was conceived to set up media (e.g.: voice, video, chat, file sharing, …) sessions among users. However, their scope has evolved and expanded over time to cover virtually any type of content and media session. As mobile networks progressively evolved towards an IP-only (All-IP) concept, particularly in 4G and 5G networks, 3GPP had to select IP-based signaling protocols for core mobile services, as opposed to traditional SS7-based protocols used in the circuit-switched domain in use in 2G and 3G networks. In that context, rather than reinventing the wheel, 3GPP decided to leverage Internet protocols and the work carried on by the IETF. Hence, it was not surprise that when 3GPP defined the so-called Packet-switched Streaming Service (PSS) for real-time continuous media delivery, it selected RTSP as its signaling protocol and, more importantly, SIP was eventually selected as the core signaling protocol for all multimedia core services in the mobile (All-)IP domain. This 3GPP decision to use off-the-shelf IETF-standardized signaling protocols has been a key cornerstone for the future of All-IP fixed / mobile networks convergence and Next Generation Networks (NGN) in general. In this context, the main goal of our work has been analyzing how such general purpose IP multimedia signaling protocols are deployed and behave over 3GPP mobile networks. Effectively, usage of IP protocols is key to enable cross-vendor interoperability. On the other hand, due to the specific nature of the mobile domain, there are scenarios where it might be possible to leverage some additional “context” to enhance the performance of such protocols in the particular case of mobile networks. With this idea in mind, the bulk of this thesis work has consisted on analyzing and optimizing the performance of SIP and RTSP multimedia signaling protocols and defining optimized deployment architectures, with particular focus on the 3GPP PSS and the 3GPP Mission Critical Push-to-Talk (MCPTT) service. This work was preceded by a detailed analysis work of the performance of underlying IP, UDP and TCP protocol performance over 3GPP networks, which provided the best baseline for the future work around IP multimedia signaling protocols. Our contributions include the proposal of new optimizations to enhance multimedia streaming session setup procedures, detailed analysis and optimizations of a SIP-based Presence service and, finally, the definition of new use cases and optimized deployment architectures for the 3GPP MCPTT service. All this work has been published in the form of one book, three papers published in JCR cited International Journals, 5 articles published in International Conferences, one paper published in a National Conference and one awarded patent. This thesis work provides a detailed description of all contributions plus a comprehensive overview of their context, the guiding principles beneath all contributions, their applicability to different network deployment technologies (from 2.5G to 5G), a detailed overview of the related OMA and 3GPP architectures, services and design principles. Last but not least, the potential evolution of this research work into the 5G domain is also outlined as well.Els mecanismes de Senyalització en xarxes de nova generació es fonamenten en protocols de senyalització definits per IETF. En particular, SIP i RTSP són dos protocols extensibles basats en missatges de text i paradigma petició-resposta. RTSP va ser concebut per a establir sessions de streaming de continguts, mentre SIP va ser creat inicialment per a facilitar l’establiment de sessions multimèdia (veu, vídeo, xat, compartició) entre usuaris. Tot i així, el seu àmbit d’aplicació s’ha anat expandint i evolucionant fins a cobrir virtualment qualsevol tipus de contingut i sessió multimèdia. A mesura que les xarxes mòbils han anat evolucionant cap a un paradigma “All-IP”, particularment en xarxes 4G i 5G, 3GPP va seleccionar els protocols i arquitectures destinats a gestionar la senyalització dels serveis mòbils presents i futurs. En un moment determinat 3GPP decideix que, a diferència dels sistemes 2G i 3G que fan servir protocols basats en SS7, els sistemes de nova generació farien servir protocols estandarditzats per IETF. Quan 3GPP va començar a estandarditzar el servei de Streaming sobre xarxes mòbils PSS (Packet-switched Streaming Service) va escollir el protocol RTSP com a mecanisme de senyalització. Encara més significatiu, el protocol SIP va ser escollit com a mecanisme de senyalització per a IMS (IP Multimedia Subsystem), l’arquitectura de nova generació que substituirà la xarxa telefònica tradicional i permetrà el desplegament de nous serveis multimèdia. La decisió per part de 3GPP de seleccionar protocols estàndards definits per IETF ha representat una fita cabdal per a la convergència del sistemes All-IP fixes i mòbils, i per al desenvolupament de xarxes NGN (Next Generation Networks) en general. En aquest context, el nostre objectiu inicial ha estat analitzar com aquests protocols de senyalització multimèdia, dissenyats per a xarxes IP genèriques, es comporten sobre xarxes mòbils 3GPP. Efectivament, l’ús de protocols IP és fonamental de cara a facilitar la interoperabilitat de solucions diferents. Per altra banda, hi ha escenaris a on és possible aprofitar informació de “context” addicional per a millorar el comportament d’aquests protocols en al cas particular de xarxes mòbils. El cos principal del treball de la tesi ha consistit en l’anàlisi i optimització del rendiment dels protocols de senyalització multimèdia SIP i RTSP, i la definició d’arquitectures de desplegament, amb èmfasi en els serveis 3GPP PSS i 3GPP Mission Critical Push-to-Talk (MCPTT). Aquest treball ha estat precedit per una feina d’anàlisi detallada del comportament dels protocols IP, TCP i UDP sobre xarxes 3GPP, que va proporcionar els fonaments adequats per a la posterior tasca d’anàlisi de protocols de senyalització sobre xarxes mòbils. Les contribucions inclouen la proposta de noves optimitzacions per a millorar els procediments d’establiment de sessions de streaming multimèdia, l’anàlisi detallat i optimització del servei de Presència basat en SIP i la definició de nous casos d’ús i exemples de desplegament d’arquitectures optimitzades per al servei 3GPP MCPTT. Aquestes contribucions ha quedat reflectides en un llibre, tres articles publicats en Revistes Internacionals amb índex JCR, 5 articles publicats en Conferències Internacionals, un article publicat en Congrés Nacional i l’adjudicació d’una patent. La tesi proporciona una descripció detallada de totes les contribucions, així com un exhaustiu repàs del seu context, dels principis fonamentals subjacents a totes les contribucions, la seva aplicabilitat a diferents tipus de desplegaments de xarxa (des de 2.5G a 5G), així una presentació detallada de les arquitectures associades definides per organismes com OMA o 3GPP. Finalment també es presenta l’evolució potencial de la tasca de recerca cap a sistemes 5G.Postprint (published version

Context transfer support for mobility management in all-IP networks.

This thesis is a description of the research undertaken in the course of the PhD and evolves around a context transfer protocol which aims to complement and support mobility management in next generation mobile networks. Based on the literature review, it was identified that there is more to mobility management than handover management and the successful change of routing paths. Supportive mechanisms like fast handover, candidate access router discovery and context transfer can significantly contribute towards achieving seamless handover which is especially important in the case of real time services. The work focused on context transfer motivated by the fact that it could offer great benefits to session re-establishment during the handover operation of a mobile user and preliminary testbed observations illustrated the need for achieving this. Context transfer aims to minimize the impact of certain transport, routing, security-related services on the handover performance. When a mobile node (MN) moves to a new subnet it needs to continue such services that have already been established at the previous subnet. Examples of such services include AAA profile, IPsec state, header compression, QoS policy etc. Re-establishing these services at the new subnet will require a considerable amount of time for the protocol exchanges and as a result time- sensitive real-time traffic will suffer during this time. By transferring state to the new domain candidate services will be quickly re-established. This would also contribute to the seamless operation of application streams and could reduce susceptibility to errors. Furthermore, re-initiation to and from the mobile node will be avoided hence wireless bandwidth efficiency will be conserved. In this research an extension to mobility protocols was proposed for supporting state forwarding capabilities. The idea of forwarding states was also explored for remotely reconfiguring middleboxes to avoid any interruption of a mobile users' sessions or services. Finally a context transfer module was proposed to facilitate the integration of such a mechanism in next generation architectures. The proposals were evaluated analytically, via simulations or via testbed implementation depending on the scenario investigated. The results demonstrated that the proposed solutions can minimize the impact of security services like authentication, authorization and firewalls on a mobile user's multimedia sessions and thus improving the overall handover performance