Call and messaging performance comparison between IMS and SIP networks

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.

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.

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.

Distributed intrusion detection scheme for next generation networks

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.

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.

Selective parsing to enhance SIP performance

The IP Multimedia Subsystem (IMS) in all-IP based next generation communication environment can offer network/service providers various multimedia services (e.g.,voice, audio, video and data etc). The IMS is worthy of notice in that it is possible to make wired/wireless convergence and quick service development. The IMS needs an efficient Session Initiation Protocol (SIP)-based routing among Call Session Control Functions (CSCF) in charge of call and session processing. In this paper, we propose a dynamic routing algorithm for managing CSCF agents using management system based on Fault, Configuration, Accounting, Performance and Security (FCAPS) in the IMS network.

Nowadays, session initiation protocol (SIP) is the most important application layer protocol for multi-media to create, modify, and terminate the sessions. SIP is the platform of next generation networks. In this way, SIP should be able to respond to the needs of such a largely-used network. One of the major problems in SIP networks is overload. The basic mechanism embedded in SIP cannot appropriately resolve the due problem by purely rejecting requests, which may cause collapse in network. Many methods have been proposed to overcome overload in SIP, among which, multi-agent systems are new agent-based and increasingly growing approaches. Multi-agent system is a powerful tool to model and develop the complex large scale distributed systems. In SIP network, each agent may point out to a server which is in interaction with the others. As a result, a distributed and complex system is simply modeled. The due complexity can be reduced by holonic organization. In this paper, SIP network is divided into geographical areas in which each holon controls an area. The entire network is controlled in hierarchical structure of holons. Holonics structure can be dynamically changed based on the circumstances parameters and the status of the network during execution. Holons implement an end-to-end window-based method. This method offers an appropriate window size for edge servers to control load from the beginning of the network and prevent network overload. The simulation results confirm improved throughput, as well as better reduced call setup delay, rapid response to change and fairness.

IMS (IP multimedia subsystem) is a main trend of development NGN. The base of IMS is an Internet Protocol which initially did not provide support for quality of service. However, for the transmission of multimedia streams is necessary to ensure a given level of quality of service (QoS). Therefore, IMS QoS estimation is a necessary task but a high complexity of network topologies and a large number of information flows makes it difficult to solve this problem. In IMS network, QoS support is provided by the interaction of systems of IMS on control and application planes. This interaction based on the signalling message exchange that also has delay. This paper presents the results to create a nodal tensor model of IMS network for delay analysis. In this model, the main systems of the IMS network such as CSCF (Call Session Control Function), HSS (Home Subscriber Server) etc. are represented as a single-server queuing systems. Model is formed in accordance to the postulates of the tensor analysis of networks. Moreover, proposed model allow estimating the delay time as for the individual routes, and for the whole network. Research of the model allowed determining such distribution of traffic that resulted to the minimum value of the delay that in turn leads to increased QoS of information flows in an IMS network.

A novel mobile peer-to-peer (MP2P) in cellular networks is presented in this paper. The proposed architecture is based on Session Initiation Protocol (SIP) that is part of the IP Multimedia Subsystem (IMS) in the third-generation mobile cellular networks. In this architecture, P2P control messages are transported in SIP messages, so called P2PoverSIP. By way of SIP, the operators can have control on security, group management and chargeability. We describe the target architecture for IMS network. The performance of the architecture is analyzed mathematically.

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.

The performance characteristics of Session Initiation Protocol (SIP) network elements determine what services can be supported by SIP networks. These elements must be able to provide customers with appropriate response times for new services, e.g., push-to-talk service. In IP-telephony converged networks, SIP-based signaling elements must meet the requirements of telephony signaling — that is, SIP-based servers will be required to process messages according to real-time constraints. Performance of SIP network elements also determines how much SIP networks cost. To support large customer bases, service providers must provide adequate performance on economical components. In this paper, we examine the main processing requirements of SIP elements and describe a number of performance metrics. We support our analyses with measurements of four different implementations of selected SIP functions.