97 research outputs found

    Reliable Session Initiation Protocol

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    Evaluating Extensions to IMS Session Setup for Multicast-based Many-to-Many Services

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    Telecommunication networks are converging towards an all-IP paradigm that integrates a broad set of value-added services. In this context, the IP Multimedia Subsystem (IMS) is being developed by the 3GPP as a key element to achieve the convergence. Additionally, multiparty services are nowadays acquiring an increasing interest from the industry. In this respect, network multicast provides a cost-effective solution to deliver these services to the user. Nevertheless, although network multicast is being considered as an enabler for one-to-many services (e.g. IPTV) in the IMS, the specifications for many-to-many services still follow a unicast approach (e.g. push-to-talk and conference). This paper describes extensions to the session control procedures in the IMS, to support multicast based multi-user services. The idea was first described in a prior work, but this paper presents enhancements to provide a comprehensive solution and to improve the grade of service (GOS) perceived by the users. In addition, the GOS achieved by the proposal is evaluated. First, the bandwidth utilization for the multicast-based multi-user services is analyzed and compared against the unicast scenario. Next, the GOS is evaluated using an analytical approach, by obtaining the mathematical expressions for the session and user plane setup delays. Finally, the GOS is also evaluated using an experimental approach, and the results are compared with values recommended by the ITU-T

    VoLTE service implementation in EPS-IMS networks

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    DiplomovĂĄ prĂĄce popisuje VoLTE sluĆŸbu, vĂœvoj a nasazenĂ­ LTE (zavĂĄděcĂ­ fĂĄzi, skutečnĂœ LTE stav a vĂœhledy do budoucna atd.), EPC-IMS architekturu (popis funkce uzlu, rozhranĂ­ atd.) Komunikace mezi uzly a funkce, rozhranĂ­ a protokoly jsou pouĆŸĂ­vĂĄny v prĆŻběhu signalizace (SIP SDP) a datovĂœ tok (RTCP RTP). PrĂĄce stručně popisuje zĂĄkladnĂ­ toky hovorĆŻ, typy nosičƯ (GBR and N-GBR), a to vytvoƙenĂ­ / mazanĂ­ nosičƯ během komunikace. DalĆĄĂ­ část diplomovĂ© prĂĄce o implementaci volte, instalace a konfigurace IMS. ZĂĄvěrečnĂĄ část diplomovĂ© prĂĄce popisuje zkouĆĄky sĂ­tě a, analĂœzu protokolu.The master's thesis describes VoLTE service, LTE evolution and deployment (deployment phases, actual LTE state and future perspectives etc.), EPC-IMS architecture (functional node description, interfaces etc.). Communications between nodes and functions, interfaces and protocols which are used during signaling (SIP-SDP) and data flow (RTCP RTP). Thesis briefly describe basic call flows, bearers types (GBR and N-GBR) and their establishment/delete during communication. The next part of master's thesis is about VoLTE implementation solutions, IMS installation and configuration. The final part of master's thesis describes the network and protocols tests, analyzes.

    Quality of Service optimisation framework for Next Generation Networks

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    Within recent years, the concept of Next Generation Networks (NGN) has become widely accepted within the telecommunication area, in parallel with the migration of telecommunication networks from traditional circuit-switched technologies such as ISDN (Integrated Services Digital Network) towards packet-switched NGN. In this context, SIP (Session Initiation Protocol), originally developed for Internet use only, has emerged as the major signalling protocol for multimedia sessions in IP (Internet Protocol) based NGN. One of the traditional limitations of IP when faced with the challenges of real-time communications is the lack of quality support at the network layer. In line with NGN specification work, international standardisation bodies have defined a sophisticated QoS (Quality of Service) architecture for NGN, controlling IP transport resources and conventional IP QoS mechanisms through centralised higher layer network elements via cross-layer signalling. Being able to centrally control QoS conditions for any media session in NGN without the imperative of a cross-layer approach would result in a feasible and less complex NGN architecture. Especially the demand for additional network elements would be decreased, resulting in the reduction of system and operational costs in both, service and transport infrastructure. This thesis proposes a novel framework for QoS optimisation for media sessions in SIP-based NGN without the need for cross-layer signalling. One key contribution of the framework is the approach to identify and logically group media sessions that encounter similar QoS conditions, which is performed by applying pattern recognition and clustering techniques. Based on this novel methodology, the framework provides functions and mechanisms for comprehensive resource-saving QoS estimation, adaptation of QoS conditions, and support of Call Admission Control. The framework can be integrated with any arbitrary SIP-IP-based real-time communication infrastructure, since it does not require access to any particular QoS control or monitoring functionalities provided within the IP transport network. The proposed framework concept has been deployed and validated in a prototypical simulation environment. Simulation results show MOS (Mean Opinion Score) improvement rates between 53 and 66 percent without any active control of transport network resources. Overall, the proposed framework comes as an effective concept for central controlled QoS optimisation in NGN without the need for cross-layer signalling. As such, by either being run stand-alone or combined with conventional QoS control mechanisms, the framework provides a comprehensive basis for both the reduction of complexity and mitigation of issues coming along with QoS provision in NGN

    Service composition based on SIP peer-to-peer networks

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    Today the telecommunication market is faced with the situation that customers are requesting for new telecommunication services, especially value added services. The concept of Next Generation Networks (NGN) seems to be a solution for this, so this concept finds its way into the telecommunication area. These customer expectations have emerged in the context of NGN and the associated migration of the telecommunication networks from traditional circuit-switched towards packet-switched networks. One fundamental aspect of the NGN concept is to outsource the intelligence of services from the switching plane onto separated Service Delivery Platforms using SIP (Session Initiation Protocol) to provide the required signalling functionality. Caused by this migration process towards NGN SIP has appeared as the major signalling protocol for IP (Internet Protocol) based NGN. This will lead in contrast to ISDN (Integrated Services Digital Network) and IN (Intelligent Network) to significantly lower dependences among the network and services and enables to implement new services much easier and faster. In addition, further concepts from the IT (Information Technology) namely SOA (Service-Oriented Architecture) have largely influenced the telecommunication sector forced by amalgamation of IT and telecommunications. The benefit of applying SOA in telecommunication services is the acceleration of service creation and delivery. Main features of the SOA are that services are reusable, discoverable combinable and independently accessible from any location. Integration of those features offers a broader flexibility and efficiency for varying demands on services. This thesis proposes a novel framework for service provisioning and composition in SIP-based peer-to-peer networks applying the principles of SOA. One key contribution of the framework is the approach to enable the provisioning and composition of services which is performed by applying SIP. Based on this, the framework provides a flexible and fast way to request the creation for composite services. Furthermore the framework enables to request and combine multimodal value-added services, which means that they are no longer limited regarding media types such as audio, video and text. The proposed framework has been validated by a prototype implementation

    Linking session based services with transport plane resources in IP multimedia subsystems.

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    The massive success and proliferation of Internet technologies has forced network operators to recognise the benefits of an IP-based communications framework. The IP Multimedia Subsystem (IMS) has been proposed as a candidate technology to provide a non-disruptive strategy in the move to all-IP and to facilitate the true convergence of data and real-time multimedia services. Despite the obvious advantages of creating a controlled environment for deploying IP services, and hence increasing the value of the telco bundle, there are several challenges that face IMS deployment. The most critical is that posed by the widespread proliferation ofWeb 2.0 services. This environment is not seen as robust enough to be used by network operators for revenue generating services. However IMS operators will need to justify charging for services that are typically available free of charge in the Internet space. Reliability and guaranteed transport of multimedia services by the efficient management of resources will be critical to differentiate IMS services. This thesis investigates resource management within the IMS framework. The standardisation of NGN/IMS resource management frameworks has been fragmented, resulting in weak functional and interface specifications. To facilitate more coherent, focused research and address interoperability concerns that could hamper deployment, a Common Policy and Charging Control (PCC) architecture is presented that defines a set of generic terms and functional elements. A review of related literature and standardisation reveals severe shortcomings regarding vertical and horizontal coordination of resources in the IMS framework. The deployment of new services should not require QoS standardisation or network upgrade, though in the current architecture advanced multimedia services are not catered for. It has been found that end-to-end QoS mechanisms in the Common PCC framework are elementary. To address these challenges and assist network operators when formulating their iii NGN strategies, this thesis proposes an application driven policy control architecture that incorporates end-user and service requirements into the QoS negotiation procedure. This architecture facilitates full interaction between service control and resource control planes, and between application developers and the policies that govern resource control. Furthermore, a novel, session based end-to-end policy control architecture is proposed to support inter-domain coordination across IMS domains. This architecture uses SIP inherent routing information to discover the routes traversed by the signalling and the associated routes traversed by the media. This mechanism effectively allows applications to issue resource requests from their home domain and enable end-to-end QoS connectivity across all traversed transport segments. Standard interfaces are used and transport plane overhaul is not necessary for this functionality. The Common PCC, application driven and session based end-to-end architectures are implemented in a standards compliant and entirely open source practical testbed. This demonstrates proof of concept and provides a platform for performance evaluations. It has been found that while there is a cost in delay and traffic overhead when implementing the complete architecture, this cost falls within established criteria and will have an acceptable effect on end-user experience. The open nature of the practical testbed ensures that all evaluations are fully reproducible and provides a convenient point of departure for future work. While it is important to leave room for flexibility and vendor innovation, it is critical that the harmonisation of NGN/IMS resource management frameworks takes place and that the architectures proposed in this thesis be further developed and integrated into the single set of specifications. The alternative is general interoperability issues that could render end-to-end QoS provisioning for advanced multimedia services almost impossible

    A Unified Mobility Management Architecture for Interworked Heterogeneous Mobile Networks

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    The buzzword of this decade has been convergence: the convergence of telecommunications, Internet, entertainment, and information technologies for the seamless provisioning of multimedia services across different network types. Thus the future Next Generation Mobile Network (NGMN) can be envisioned as a group of co-existing heterogeneous mobile data networking technologies sharing a common Internet Protocol (IP) based backbone. In such all-IP based heterogeneous networking environments, ongoing sessions from roaming users are subjected to frequent vertical handoffs across network boundaries. Therefore, ensuring uninterrupted service continuity during session handoffs requires successful mobility and session management mechanisms to be implemented in these participating access networks. Therefore, it is essential for a common interworking framework to be in place for ensuring seamless service continuity over dissimilar networks to enable a potential user to freely roam from one network to another. For the best of our knowledge, the need for a suitable unified mobility and session management framework for the NGMN has not been successfully addressed as yet. This can be seen as the primary motivation of this research. Therefore, the key objectives of this thesis can be stated as: To propose a mobility-aware novel architecture for interworking between heterogeneous mobile data networks To propose a framework for facilitating unified real-time session management (inclusive of session establishment and seamless session handoff) across these different networks. In order to achieve the above goals, an interworking architecture is designed by incorporating the IP Multimedia Subsystem (IMS) as the coupling mediator between dissipate mobile data networking technologies. Subsequently, two different mobility management frameworks are proposed and implemented over the initial interworking architectural design. The first mobility management framework is fully handled by the IMS at the Application Layer. This framework is primarily dependant on the IMS’s default session management protocol, which is the Session Initiation Protocol (SIP). The second framework is a combined method based on SIP and the Mobile IP (MIP) protocols, which is essentially operated at the Network Layer. An analytical model is derived for evaluating the proposed scheme for analyzing the network Quality of Service (QoS) metrics and measures involved in session mobility management for the proposed mobility management frameworks. More precisely, these analyzed QoS metrics include vertical handoff delay, transient packet loss, jitter, and signaling overhead/cost. The results of the QoS analysis indicates that a MIP-SIP based mobility management framework performs better than its predecessor, the Pure-SIP based mobility management method. Also, the analysis results indicate that the QoS performances for the investigated parameters are within acceptable levels for real-time VoIP conversations. An OPNET based simulation platform is also used for modeling the proposed mobility management frameworks. All simulated scenarios prove to be capable of performing successful VoIP session handoffs between dissimilar networks whilst maintaining acceptable QoS levels. Lastly, based on the findings, the contributions made by this thesis can be summarized as: The development of a novel framework for interworked heterogeneous mobile data networks in a NGMN environment. The final design conveniently enables 3G cellular technologies (such as the Universal Mobile Telecommunications Systems (UMTS) or Code Division Multiple Access 2000 (CDMA2000) type systems), Wireless Local Area Networking (WLAN) technologies, and Wireless Metropolitan Area Networking (WMAN) technologies (e.g., Broadband Wireless Access (BWA) systems such as WiMAX) to interwork under a common signaling platform. The introduction of a novel unified/centralized mobility and session management platform by exploiting the IMS as a universal coupling mediator for real-time session negotiation and management. This enables a roaming user to seamlessly handoff sessions between different heterogeneous networks. As secondary outcomes of this thesis, an analytical framework and an OPNET simulation framework are developed for analyzing vertical handoff performance. This OPNET simulation platform is suitable for commercial use

    Developing a cross platform IMS client using the JAIN SIP applet phone

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    Since the introduction of the IP Multimedia Subsystem (IMS) by the Third Generation Partnership Project (3GPP) in 2002, a lot of research has been conducted aimed at designing and implementing IMS capable clients and network elements. Though considerable work has been done in the development of IMS clients, there is no single, free and open source IMS client that provides researchers with all the required functionality needed to test the applications they are developing. For example, several open and closed source SIP/IMS clients are used within the Rhodes University Conver- gence Research Group (RUCRG) to test applications under development, as a result of the fact that the various SIP/IMS clients support different subsets of SIP/IMS features. The lack of a single client and the subsequent use of various clients comes with several problems. Researchers have to know how to deploy, configure, use and at times adapt the various clients to suit their needs. This can be very time consuming and, in fact, contradicts the IMS philosophy (the IMS was proposed to support rapid service creation). This thesis outlines the development of a Java-based, IMS compliant client called RUCRG IMS client, that uses the JAIN SIP Applet Phone (JSAP) as its foundation. JSAP, which originally offered only basic voice calling and instant messaging (IM) capabilities, was modified to be IMS compliant and support video calls, IM and presence using XML Configuration Access Protocol (XCAP)

    Peer-to-peer television for the IP multimedia subsystem

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    Peer-to-peer (P2P) video streaming has generated a significant amount of interest in both the research community and the industry, which find it a cost-effective solution to the user scalability problem. However, despite the success of Internet-based applications, the adoption has been limited for commercial services, such as Internet Protocol Television (IPTV). With the advent of the next-generation-networks (NGN) based on the IP Multimedia Subsystem (IMS), advocating for an open and inter-operable architecture, P2P emerges as a possible alternative in situations where the traditional mechanisms are not possible or economically feasible. This work proposes a P2P IPTV architecture for an IMS-based NGN, called P2PTV, which allows one or more service providers to use a common P2P infrastructure for streaming the TV channels to their subscribers. Instead of using servers, we rely on the uploading capabilities of the user equipments, like set-top boxes, located at the customers’ premise. We comply with the existing IMS and IPTV standards from the 3rd Generation Partnership Project (3GPP) and the Telecommunications and Internet converged Services and Protocols for Advanced Networking (TISPAN) bodies, where a centralized P2PTV application server (AS) manages the customer access to the service and the peer participation. Because watching TV is a complex and demanding user activity, we face two significant challenges. The first is to accommodate the mandatory IMS signaling, which reserves in the network the necessary QoS resources during every channel change, establishing a multimedia session between communicating peers. The second is represented by the streaming interruptions, or churn, when the uploading peer turns off or changes its current TV channel. To tackle these problems, we propose two enhancements. A fast signaling method, which uses inactive uploading sessions with reserved but unused QoS, to improve the tuning delay for new channel users. At every moment, the AS uses a feedback based algorithm to compute the number of necessary sessions that accommodates well the demand, while preventing the over-reservation of resources. We approach with special care mobility situations, where a proactive transfer of the multimedia session context using the IEEE 802.21 standard offers the best alternative to current methods. The second enhancement addresses the peer churn during channel changes. With every TV channel divided into a number of streams, we enable peers to download and upload streams different from their current channel, increasing the stability of their participation. Unlike similar work, we benefit from our estimation of the user demand and propose a decentralized method for a balanced assignment of peer bandwidth. We evaluate the performance of the P2PTV through modeling and large-scale computer simulations. A simpler experimental setting, with pure P2P streaming, indicates the improvements over the delay and peer churn. In more complex scenarios, especially with resource-poor peers having a limited upload capacity, we envision P2P as a complementary solution to traditional approaches like IP multicast. Reserving P2P for unpopular TV channels exploits the peer capacity and prevents the necessity of a large number of sparsely used multicast trees. Future work may refine the AS algorithms, address different experimental scenarios, and extend the lessons learned to non-IMS networks. ----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------La transmisiĂłn de vĂ­deo con tecnologĂ­as peer-to-peer (P2P) ha generado un gran interĂ©s, tanto en la industria como en la comunidad cientĂ­fica, quienes han encontrado en dicha uniĂłn la soluciĂłn para afrontar los problemas de escalabilidad de la transmisiĂłn de vĂ­deo, reduciendo al mismo tiempo sus costes. A pesar del Ă©xito de estos mecanismos en Internet, la transmisiĂłn de vĂ­deo mediante tĂ©cnicas P2P no se ha utilizado en servicios comerciales como puede ser el de televisiĂłn por IP (IPTV). Con la apariciĂłn de propuestas de redes de prĂłxima generaciĂłn basadas en el IP Multimedia Subsystem (IMS), que permite una arquitectura abierta e interoperable, los mecanismos basados en P2P emergen como posibles alternativas en situaciones donde los mecanismos tradicionales de transmisiĂłn de vĂ­deo no se pueden desplegar o no son econĂłmicamente viables. Esta tesis propone una arquitectura de servicio de televisiĂłn peer-to-peer para una red de siguiente generaciĂłn basada en IMS, que abreviaremos como P2PTV, que permite a uno o mĂĄs proveedores de servicio utilizar una infraestructura P2P comĂșn para la transmisiĂłn de canales de TV a sus suscriptores. En vez de utilizar varios servidores, proponemos utilizar la capacidad de envĂ­o de los equipos de usuario, como los set-top boxes, localizados en el lado del cliente. En esta tesis extendemos los trabajos de estandarizaciĂłn sobre IMS IPTV de los organismos 3rd Generation Partnership Project (3GPP) y del Telecommunications and Internet converged Services and Protocols for Advanced Networking (TISPAN), donde un servidor de aplicaciĂłn (AS) central de P2PTV administra el acceso de los clientes al servicio y permite compartir los recursos de los equipos. Debido a que el acceso a los canales de TV por parte de los usuarios es una actividad compleja, nos enfrentamos a dos retos importantes. El primero es administrar la señalizaciĂłn de IMS, con la cual se reservan los recursos de QoS necesarios durante cada cambio de canal, estableciendo una sesiĂłn multimedia entre los diferentes elementos de la comunicaciĂłn. El segundo estĂĄ representado por las interrupciones de la reproducciĂłn de video, causado por los equipos que sirven dicho vĂ­deo cuando estos se desconectan del sistema o cuando cambian de canal. Para afrontar estos retos, proponemos dos mejoras al sistema. La primera mejora introduce el mĂ©todo de señalizaciĂłn rĂĄpida, en la cual se utilizan sesiones multimedia inactivas pero con recursos reservados para acelerar las conexiones entre usuarios. En cada momento, el AS utiliza la informaciĂłn extraĂ­da del algoritmo propuesto, que calcula el nĂșmero de sesiones necesarias para administrar la demanda de conexiones, pero sin realizar una sobre-estimaciĂłn, manteniendo bajo el uso de los recursos. Hemos abordado con especial cuidado la movilidad de los usuarios, donde se ha propuesto una transferencia de sesiĂłn pro-activa utilizando el estĂĄndar IEEE 802.21, el cual brinda una mejor alternativa que los mĂ©todos propuestos hasta la fecha. La segunda mejora se enfoca en las desconexiones de usuarios durante cambios de canal. Dividiendo los canales de TV en varios segmentos, permitimos a los equipos descargar o enviar diferentes partes de cualquier canal, aumentando la estabilidad de su participaciĂłn. A diferencia de otros trabajos, nuestra propuesta se beneficia de la estimaciĂłn de la demanda futura de los usuarios, proponiendo un mĂ©todo descentralizado para una asignaciĂłn balanceada del ancho de banda de los equipos. Hemos evaluado el rendimiento del sistema P2PTV a travĂ©s de modelado y de simulaciones de ordenador en sistemas IPTV de gran escala. Una configuraciĂłn simple, con envĂ­o P2P puro, indica mejoras en el retardo y nĂșmero de desconexiones de usuarios. En escenarios mĂĄs complejos, especialmente con equipos con pocos recursos en la subida, sugerimos el uso de P2P como una soluciĂłn complementaria a las soluciones tradicionales de multicast IP. Reservando el uso de P2P para los canales de TV poco populares, se permite explotar los recursos de los equipos y se previene la necesidad de un alto nĂșmero de ĂĄrboles multicast dispersos. Como trabajo futuro, se propone refinar los algoritmos del AS, abordar diferentes escenarios experimentales y tambiĂ©n extender las lecciones aprendidas en esta tesis a otros sistemas no basados en IMS
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