IMS signalling for multiparty services based on network level multicast

3rd EURO-NGI Conference on Next Generation Internet Networks. Norwegian University of Science and Technology, Trondheim, Norway, 21-23 may 2007.The standardization process of the UMTS technology has led to the development of the IP Multimedia Subsystem (IMS). IMS provides a framework that supports the negotiation of the next generation multimedia services with QoS requirements that are envisioned for 3G networks. But even though many of these services involve the participation of multiple users in a multiparty arrangement, the delivery technology at network level is still unicast based. This approach is not optimum, in terms of transmission efficiency. In this paper, a new approach is presented proposing to use a network level multicast delivery technology for the multiparty services that are signalled through IMS. The main advantages and drawbacks related with this new approach are analyzed in the article. Finally, as a starting point in the development of the presented solution, a new SIP signalling dialogue is proposed allowing the negotiation of a generic multiparty service, and supporting at the same time the configuration of the corresponding network level multicast delivery service with QoS requirements that will be used in the user plane.Publicad

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

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)

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

Designing and Implementation of SIP-ALG with State Recovery Signaling

SIP (Session Initiation Protocol)による通信では，シグナリングがエンティティの主要な動作を決定する．そのため，シグナリングがないままに通信途中でエラーが発生すると，ユーザ端末を含めパス上に存在するサーバは，適切に初期状態へ移行することができない．したがって，これらは復旧までの間，様々なリソースを浪費することになる．従来までこの問題に対する解決策は，エンティティに対して独自にタイマを持たせることでのみ実現されていた．通信セッションに異常が発生しても，エンティティはリソースを解放できないことがあった．本論文では，この問題を解決をするために，状態正常化シグナリングに基づくSIP-ALGを提案する．提案方式に基づいて実装を行い，その動作を検証する．またエンティティに対して定義されているオートマトンを利用して，状態正常化シグナリングの有効性について検討する．修士論

Multimedia session continuity in the IP multimedia subsystem : investigation and testbed implementation

Includes bibliographical references (leaves 91-94).The advent of Internet Protocol (IP) based rich multimedia services and applications has seen rapid growth and adoption in recent years, with an equally increasing user base. Voice over IP (VoIP) and IP Television (IPTV) are key examples of services that are blurring the lines between traditional stove-pipe approach network infrastructures. In these, each service required a different network technology to be provisioned, and could only be accessed through a specific end user equipment (UE) technology. The move towards an all-IP core network infrastructure and the proliferation of multi-capability multi-interface user devices has spurred a convergence trend characterized by access to services and applications through any network, any device and anywhere