Internames: a name-to-name principle for the future Internet

We propose Internames, an architectural framework in which names are used to identify all entities involved in communication: contents, users, devices, logical as well as physical points involved in the communication, and services. By not having a static binding between the name of a communication entity and its current location, we allow entities to be mobile, enable them to be reached by any of a number of basic communication primitives, enable communication to span networks with different technologies and allow for disconnected operation. Furthermore, with the ability to communicate between names, the communication path can be dynamically bound to any of a number of end-points, and the end-points themselves could change as needed. A key benefit of our architecture is its ability to accommodate gradual migration from the current IP infrastructure to a future that may be a ubiquitous Information Centric Network. Basic building blocks of Internames are: i) a name-based Application Programming Interface; ii) a separation of identifiers (names) and locators; iii) a powerful Name Resolution Service (NRS) that dynamically maps names to locators, as a function of time/location/context/service; iv) a built-in capacity of evolution, allowing a transparent migration from current networks and the ability to include as particular cases current specific architectures. To achieve this vision, shared by many other researchers, we exploit and expand on Information Centric Networking principles, extending ICN functionality beyond content retrieval, easing send-to-name and push services, and allowing to use names also to route data in the return path. A key role in this architecture is played by the NRS, which allows for the co-existence of multiple network "realms", including current IP and non-IP networks, glued together by a name-to-name overarching communication primitive.Comment: 6 page

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

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

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

Enterprise network convergence: path to cost optimization

During the past two decades, telecommunications has evolved a great deal. In the eighties, people were using television, radio and telephone as their communication systems. Eventually, the introduction of the Internet and the WWW immensely transformed the telecommunications industry. This internet revolution brought about a huge change in the way businesses communicated and operated. Enterprise networks now had an increasing demand for more bandwidth as they started to embrace newer technologies. The requirements of the enterprise networks grew as the applications and services that were used in the network expanded. This stipulation for fast and high performance communication systems has now led to the emergence of converged network solutions. Enterprises across the globe are investigating new ways to implement voice, video, and data over a single network for various reasons – to optimize network costs, to restructure their communication system, to extend next generation networking abilities, or to bridge the gap between their corporate network and the existing technological progress. To date, organizations had multiple network services to support a range of communication needs. Investing in this type of multiple communication infrastructures limits the networks ability to provide resourceful bandwidth optimization services throughout the system. Thus, as the requirements for the corporate networks to handle dynamic traffic grow day by day, the need for a more effective and efficient network arises. A converged network is the solution for enterprises aspiring to employ advanced applications and innovative services. This thesis will emphasize the importance of converging network infrastructure and prove that it leads to cost savings. It discusses the characteristics, architecture, and relevant protocols of the voice, data and video traffic over both traditional infrastructure and converged architecture. While IP-based networks present excellent quality for non real-time data networking, the network by itself is not capable of providing reliable, quality and secure services for real-time traffic. In order for IP networks to perform reliable and timely transmission of real-time data, additional mechanisms to reduce delay, jitter and packet loss are required. Therefore, this thesis will also discuss the important mechanisms for running real-time traffic like voice and video over an IP network. Lastly, it will also provide an example of an enterprise network specifications (voice, video and data), and present an in depth cost analysis of a typical network vs. a converged network to prove that converged infrastructures provide significant savings

Mechanisms for AAA and QoS Interaction

Proceedings of Third IEEE Workshop on Applications and Services in Wireless Networks, ASWN 2003. Bern, Switzerland, July 2-4, 2003.The interaction between Authentication, Authorization and Accounting (AAA) systems and the Quality of Service (QoS) infrastructure is to become a must in the near future. This interaction will allow rich control and management of both users and networks. DIAMETER and DiffServ are likely to turn into the future standards in AAA and QoS systems, but they are not designed to interact with each other. To face this, we propose a new Diameter-Diffserv interaction model and describe the Application Specific Module (ASM) implemented to allow this interaction. The ASM has been implemented and tested in a complete AAA-QoS IPv6 scenario

Unicast UDP Usage Guidelines for Application Designers

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