SIP-Based IMS Signaling Analysis for WiMax-3G Interworking Architectures

The 3rd generation partnership project (3GPP) and 3GPP2 have standardized the IP multimedia subsystem (IMS) to provide ubiquitous and access network independent IP-based services for next generation networks via merging cellular networks and the Internet. The application layer Session Initiation Protocol (SIP), standardized by 3GPP and 3GPP2 for IMS, is responsible for IMS session establishment, management, and transformation. The IEEE 802.16 worldwide interoperability for microwave access (WiMax) promises to provide high data rate broadband wireless access services. In this paper, we analyze the SIP-based IMS registration signaling delay for 3rd generation (3G) cellular and WiMax networks with specific reference to their interworking architectures. We also explore the effects of different WiMax-3G interworking architectures on the IMS registration signaling delay.

Within the 3rd Generation Partnership Project (3GPP), the thrust of the session initiation protocol (SIP)-based Internet protocol (IP) multimedia subsystem (IMS) is envisaged to allow a swift progression towards the provision of multimedia applications for increasingly demanding end users. The paradigm of service programmability using open network application programming interfaces (APIs), with open service access (OSA) as its main exponent, is helping to drive this development together with the use of SIP. The focal point of this paper will be the multimedia services architecture in the IMS by providing details of the interaction of the IMS and the application servers in the form of the OSA gateway and the SIP application server. The paper aims to assess the value of the IMS service control (ISC) interface on application server interaction in the IMS. The paper will provide an OSA application use case, and will also present the presence server as an example of a SIP application server that fits in with the IMS. © 2003 Lucent Technologies Inc.

The IP Multimedia Subsystem (IMS) has been considered to be exclusive to big players of the telecommunication world. Session Initiation Protocol (SIP) has been standardized by IETF and solutions based on it have been in the market at least for a decade. It is also an essential part of the IMS, as it is the control protocol of IMS. Recent development of the Linux platform and the Open source IMS system has lead to a situation where the IMS is also an affordable solution for smaller telecommunication companies. The obvious question has been whether the level of performance with Open source IMS is really suitable for business usage. ETSI-TISPAN has unified IMS performance measurements by defining a common framework which to follow in measurements. The paper presents comparisons between IMS and SIP based networks in call and messaging performance. the Open IMS solution by the Fraunhover Focus Institute is compared with the OpenSIPS, an open source SIP server. The number of users in our measurements was chosen to be 100 000, a large enough number to be considered as a suitable number of users for small or moderate business usage needs. Our study shows that both of the systems are able to handle this many users and, in the case of OpenSIPS, with a clear margin. It was also discovered that both systems are reliable and are suitable for small and moderate business usage.

As the next generation network NGN core network solution, IMS (IP multimedia subsystem) can support a variety of terminal access method and provide a foundation for the fixed network and mobile network integration. IMS is using the session initiation protocol (SIP) as its core control protocol to establish, manage and terminate multimedia services. IMS provides multimedia communication services based on data, voice and video, it gives users an unprecedented experience. Satellite mobile communication system has become an integral part of the global communication network. It can be seamlessly connected to the global communication network, which provides a strong complement to the ground network coverage. In recent years, the research and implementation of multimedia services through satellite access to the IMS core network has also been concerned. However, the problem of propagation delay of satellite channel has a great influence on the QoS (quality of service) of IMS access. Therefore, the application of IMS streaming media transmission and SIP signaling interaction must also consider the propagation delay of the satellite channel. In this paper, the process of satellite access to IMS and the transmission delay of satellite communication was simulated; the complete IMS voice and video service access and communication process was designed and implemented. In order to solve the problem of the propagation delay of the satellite in the communication process, the session signaling process was optimized.

In this paper we propose a novel architecture for converged multimedia SIP (session initiation protocol) services in wireline and 3G networks. The work on the future evolution of 3G systems is taking momentum with ongoing research on access to the IMS (IP Multimedia Subsystem) from any broadband access network. In particular WLAN (wireless local area network) access to the 3G services located in the IMS is being standardized though major points still need to be addressed. Wireline access to 3G services is not yet standardized and no deployable solution exists today. We propose in this paper a new architecture that allows an xDSL (digital subscriber line) access to 3G SIP based services, enabling a new generation of converged fixed/mobile services. This would also allow network operators to mutualise equipments and therefore reduce costs. In this paper, we analyze the high level requirements of this architecture, and we propose to introduce two new functional entities in the xDSL network: a SIP proxy and a border gateway. We study the impact of this new architecture with several transversal requirements: connectivity, QoS and security

IP multimedia subsystems (IMS) is using session initiation protocol (SIP) as signalling protocol. SIP is becoming computing intensive and hence the ability of entry and exit proxies to process SIP messages has great impact on the overall performance of the IMS core network. This paper shows that the flow of SIP messages during handover process in the IMS core network complies with the well known central server model. The later is validated via discrete event simulation results. The performance prediction study reveals that S-CSCF is the bottleneck device of the IMS core network processing a large number of SIP messages.

This article describes a novel architecture signaling network so that an IP Multimedia Subsystem (IMS) user, flying in an aircraft, be able to register with his IMS network of origin and have access to personal services and multimedia, through a Wide-Band Code Division Multiple Access (W-CDMA) link. The proposed architecture integrates the Session Initialization Protocol (SIP), the Network Mobility Basic Support (NEMO BS) protocol and the Mobile Internet Protocol (MIPv6) to enable the usage of IMS services by passenger on board an aircraft, as a type of Aeronautical Passenger Communication (APC) services. In this novel architecture, NEMO and MIPv6 protocols are responsible for the mobility management of smartphones and user notebook and the SIP protocol is responsible for the register and the management of sessions of an IMS user. This novel architecture enables an Air Communication Service Provider (ACSP) be able to supply to an on-board passenger (The IMS user) some APC services through an Universal Mobile Telecommunication System (UMTS). The simulation results indicate the accuracy of the proposed architecture into aeronautical scenarios.

IP Multimedia Subsystem (IMS) network peering is a key enabler that will help accelerate deployment of next-generation IMS-based networks. Today's early deployments of dispersed IMS networks require public switched telephone network (PSTN)/public land mobile network (PLMN) bridges for network interconnection between IMS islands. The PSTN/PLMN bridging arrangement is inefficient, however, in that it results in unnecessary settlements for the carriers. It further impedes the implementation of rich multimedia and Voice over IP (VoIP)-related services that require end-to-end Internet Protocol (IP) connectivity. Last, it perpetuates the reliance on the existing PSTN/PLMN network for voice calls among subscribers served by different IMS-based carriers. This paper analyzes in detail the IMS peering challenges from the perspective of Session Initiation Protocol (SIP) signaling peering. It discusses issues related to the routing of SIP messages, and addressing, address resolution, and discovery of peering points for IMS signaling peering. It further establishes that a new routing algorithm is needed that will allow signaling peering points to dynamically discover the “best” transit network among others for reaching a destination. In closing, it presents a high-level IMS signaling routing process that includes, among other benefits, support for number portability as a key function for inter-carrier IMS peering.

This chapter describes the support for IP based multimedia services on Third Generation (3G) wireless cellular networks. While earlier cellular networks offered only basic IP connectivity on a best-effort basis, 3G networks provide explicit support for real time multimedia services with guaranteed Quality of Service (QoS). As a result, they can offer traditional voice telephony, rich telephony services and multimedia streaming over IP, both inside a 3G network and in conjunction with external networks such as the Internet, the Public Switched Telephone Network (PSTN) and the Integrated Services Digital Network (ISDN). In fact, 3G networks will deploy for the first time in a large scale technologies such as IP version 6 (IPv6) and policy based IP QoS, spearheading the introduction of such technologies into the Internet. The two most important aspects of 3G networks with respect to IP based multimedia services are the IP Multimedia Subsystem (IMS) and the Multimedia Broadcast / Multicast Service (MBMS). The IMS enables complex IP based multimedia sessions to be created with guaranteed QoS for each media component. Example applications include voice telephony and video conferencing. The IMS interoperates with both traditional telephony services and external IP based multimedia services. The MBMS provides native IP broadcast and multicast support in 3G networks, allowing high bandwidth services to be economically offered to multiple users. Example applications include video streaming via multicast and location based services via broadcast. The MBMS interoperates directly with IP multicasting. While both the IMS and the MBMS are IP based, their standardization is proceeding independently. It is however clear that their combination would allow numerous new services to be provided. The outline of this chapter is as follows. In Section II we give an overview of cellular networks and their evolution, while in Section III we describe in detail a specific 3G network, the Universal Mobile Telecommunications System (UMTS). Section IV presents an introductory description of the features and services provided by the IMS and the MBMS. In Section V we provide the details of the IMS, including service architecture, session setup and control and interworking issues. Section VI describes the MBMS in the same manner as for the IMS. Finally, Section VII discusses the QoS issues for IP based multimedia services, describing the overall QoS concept, the policy based QoS control scheme and its application to IMS sessions. At the end of the chapter, Tables II and III list the acronyms used. An extensive UMTS vocabulary is provided in [1].

Session Initiation Protocol (SIP) is a new signaling protocol adopted by the 3rd Generation Partnership Project (3GPP) for the IP Multimedia Subsystem (IMS). SIP is used to establish and release sessions between end systems. SIP will play a determinant role in the future as one of the most important signaling protocols for session's establishment and management. In this paper we explore the use of SIP to manage sessions between SIP users in a DiffServ Domain. We use SIP to enhance the performance of TCP sessions by stochastic scheduling of sessions according to their size and duration.

A popular application that runs on a Voice-over-IP network is the Interactive Voice Response (IVR), which provides a way for interacting with an end user's phone via a telephony interface by presenting a set of audible menu options, and collecting the user's Dual Tone Multi-Frequency responses as the user presses the telephone numeric keypad, and recording the user's voice. This thesis takes a closer look into IVR systems on an IP-based network, specifically in networks that support the Session Initiation Protocol (SIP) for controlling the telephony signalling, and more specifically on IVR systems that have their SIP signalling controlled by a SIP Application Server (SIPAS) implementing the SipServlet 1.0 or 1.1 specifications and where a SIP-based Media Server is used to stream IVR media to the end-user. We describe existing ways for supporting an IVR development in a SIP network, and then we propose an alternative way of accomplishing the same task: the IVRObject. With the help of prototypes, the IVRObject is compared with the existing state of the art against three criteria: a) how easy it is to develop, b) how portable the development solution is, c) and how scalable the solution is in order to sustain a high call volume. As a conclusion, it will be highlighted that the IVRObject provides an easy mechanism for development of IVR-based applications running on a SIP AS, that it is portable to different media server vendors, and that it supports a test strategy that can be leveraged to improve software development quality and faster development. This makes the IVRObject a good alternative especially for enterprise-based IVR applications where scalability is less of an issue than in carrier-space applications.

The IP Multimedia Subsystem (IMS) is envisioned as a solution for the next generation multimedia rich communication. The 3rd Generation Partnership Project (3GPP) has chosen Session Initiation Protocol (SIP) as signaling protocol for IMS. The main cause of performance degradation in SIP network is the overload. SIP does not provide sufficient mechanism to handle this problem. In this paper, we propose a hierarchical modeling for performance evaluation of the network for both normal and overload conditions. An analytical model is developed using cut-off priority queueing for a SIP server and the Generalized Stochastic Petri Nets (GSPN) technique is used to study the performance of the complete network. The proposed model is suitable for handling two different types of signaling traffic and protects them from overloading against each-other. Based on our performance analysis, it is observed that the system reduces retransmissions of SIP messages, thereby reducing congestion and increasing throughput.

The 3rd Generation Partnership Project (3GPP) defines IP Multimedia Core Network Subsystem (IMS) to support IP‐based multimedia services. IMS utilizes protocols such as IP version 6 (IPv6), Session Initiation Protocol (SIP), and Real‐time Transport Protocol (RTP) to deliver the multimedia content. This paper proposes an analysis tool referred to as SIPv6 Analyzer to investigate the IMS‐related protocols during IMS service deployment. The SIPv6 Analyzer not only dissects the protocol headers but also provides user‐friendly functions such as message flow generation and audio/video replay for IPv6, SIP, and RTP. We describe the design of the SIPv6 Analyzer and use examples to illustrate how it works. Copyright © 2006 John Wiley & Sons, Ltd.

This letter describes the design and implementation of a prototype application intended to demonstrate how the IP Multimedia Subsystem (IMS) platform could be used to create blended services that help enhance a user's lifestyle. Communication applications (e.g., group instant messaging and voice conferencing) are blended with collaboration services (e.g., shared Web browsing and shared streaming video) and location-based services to create a compelling scenario that makes planning a night out with friends both convenient and fun. Several IMS components and protocols are used to achieve the blending. A seamless user interface is created through the use of an integrated Session Initiation Protocol (SIP) client and a unified graphical user interface (GUI). This prototype, which is being showcased in major tradeshows around the world, is serving as a foundation for planning new features for Lucent Technologies' IMS portfolio.

Chapter 16 - The IP Multimedia Subsystem: True Converged Communications