Session Initiation Protocol Networks Research Articles

Application of session initiation protocol to networked sensor interfaces

Network of measurement and control is now converging with network of information tightly more and more, which improves the development of network-distributed measurement and control technology. And networked sensors are important unit in networked measurement and control system. This paper applied Session Initiation Protocol (SIP) to networked sensors system aimed for a kind of unified IEEE 1451 NCAP model of networked sensors using SIP as the interface. We analyzed the features of SIP and sensor network interface, presented a frame of networked sensor based on SIP according to IEEE1451 sensor standards, and studied on-line register of networked sensor based on SIP. Devices supported by SIP in distributed control system can communicate easily with IP-based information network, and can also provide conveniently multimedia service. SIP and P2P SIP are worthy of deep research for networked sensor system.

A common SIP profile for next-generation networks

The telephony network is evolving to an all Internet Protocol (IP)-based network. The 3rd Generation Partnership Project (3GPP) and 3rd Generation Partnership Project 2 (3GPP2) standards organizations are currently defining the network architecture for the next generation IP Multimedia Subsystem (IMS) network, which is being applied to multiple access technologies and networks, including the next-generation network (NGN). Concurrently, an effort lead by International Telecommunications Union Telecommunications standardization sector study group 13 (ITU-T SG 13), and presently carried in the focus group on NGN, is adopting IMS for wireline networks. The European Telecommunications Standards Institute's (ETSI) Telecommunication and Internet converged Services and Protocols for Advanced Networks (TISPAN) group is also developing NGN specifications. An essential part of the next-generation network is the seamless interaction with the existing telephony network, enabled by interworking between integrated services digital network user part (ISUP) in the public switched telephone network (PSTN) and Session Initiation Protocol (SIP) in the IMS. Several SIP profiles have been defined in the Internet Engineering Task Force (IETF) and ITU-T for specific interworking applications. SIP is a very flexible protocol with numerous options and extensions. A SIP profile is a specific combination of options and extensions as used by an application. No profile currently supports the interworking of all existing services between SIP networks. This paper analyzes existing SIP profiles against requirements for interworking between different networks and identifies the main SIP profile characteristics necessary to effectively interoperate between multiple networks while maintaining existing services. Two SIP profiles used in ITU-T discussions — IMS SIP and SIP with encapsulated ISUP (SIP-I) — are highlighted to explain ISUP signaling situations. Requirements for a common SIP profile are then described as they relate to these two profiles.

SIP message prioritization and its applications

Session Initiation Protocol (SIP) signaling is an integral part of the IP Multimedia Subsystem (IMS) to support services such as Voice over IP (VoIP), multimedia sessions, presence, and instant messaging (IM). All of these services share a common SIP signaling infrastructure. In spite of careful network engineering, SIP network elements may experience high-load situations, in which SIP messages may experience too much delay or even get dropped. This paper addresses how such undesired high-load situations can be handled effectively by introducing a message prioritization mechanism in SIP servers. To validate the mechanism, it has been implemented in a high-performance JAIN <sup xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">∗</sup> -SIP-compliant Java <sup xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">∗</sup> SIP stack. This SIP stack incorporates several other innovations, such as a very efficient SIP message parser, and it can be used for a variety of SIP-based applications. Its design enables service providers to define custom message prioritization and rejection strategies based on SIP message characteristics, system state, and statistics. Measurements show that high-priority emergency messages can indeed be serviced without significant delay or loss in a high-load situation. The concept of and techniques for SIP prioritization are the subjects of a Lucent Technologies patent application.

The deployment of IPv6 SIP-based VoIP applications

This paper presents an IP version 6 (IPv6) Session Initiation Protocol (SIP)-based Voice over IP (VoIP) deployment in Taiwan. This deployment supported by National Information and Communication Initiative (NICI) IPv6 R&D and Promotion divisions contains the SIPv6 User Agents (UAs), Socket-layer Translator, SIPv6 Translator and SIPv6 Translator. The major contributions of this paper are introducing the IPv6 translation mechanisms for SIP-based applications and exercising the ENUM on IPv6 SIP networks, to establish the communications between the SIP-based VoIP and the Public Switched Telephone Network (PSTN) users.

An IPv4-IPv6 translation mechanism for SIP overlay network in UMTS all-IP environment

Both IPv6 and session initiation protocol (SIP) are default protocols for Universal Mobile Telecommunications System (UMTS) all-Internet protocol (IP) network. In the existing mobile telecommunications environments, an IPv6-based UMTS all-IP network needs to interwork with other Internet protocol version 4 (IPv4)-based SIP networks. Therefore, mobile SIP applications are typically offered through an overlay structure over the IPv4-Internet protocol version 6 (IPv6) interworking environments. Based on 3GPP 23.228, we propose an IPv4-IPv6 translation mechanism (i.e., SIPv6 translator) that integrates different IP infrastructures (i.e., IPv4 and IPv6) to provide an overlay network for transparent SIP application deployment. In this paper, we present the architecture and the call flows of the SIPv6 translator. An analytic model is proposed to investigate the fault tolerance issue of our approach. Our study provides guidelines to select appropriate number of processors for fault tolerance.

On SIP performance

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.

Issues in supporting multimedia services in SIP networks with mixed endpoint types

The evolution from traditional circuit-based networks to packet-based networks is a salient feature of today's telecommunications industry. As new types of Session Initiation Protocol (SIP) networks and endpoints (e.g., 3rd Generation Partnership Project Internet Protocol Multimedia Subsystem [3GPP IMS], cable, public switched telephone network [PSTN] gateways, and 802.11-capable personal digital assistant [PDAs]) are introduced, SIP extensions are being introduced to solve the problems unique to each of them. As SIP networks mature to support mixed endpoint types, interesting issues relating to interoperability among the various endpoints arise. This paper includes a discussion of some of these issues, including out-of-band versus in-band call progress information, handling early media, support of Internet Protocol (IP) and asynchronous transfer mode (ATM) transport networks, interworking with ISDN User Part (ISUP), and support of different SIP extensions. It also explores issues of performance, reliability, and service quality that arise as networks serving mixed endpoints are deployed.

Approaches to implementing services in SIP networks

The Session Initiation Protocol (SIP) provides powerful and flexible signaling capabilities for building services without enforcing a specific service model. As such, it provides the opportunity for service manufacturers to design a service model that is well suited for the targeted services and network environment. In this letter, we evaluate common service models and introduce the new cooperative service model.

Media services in SIP networks

Media services have played an important role in the traditional time division multiplexing (TDM)-based telephone network. As the network migrates to an Internet Protocol (IP)-based environment, media services are also moving to new environments. This paper reviews the background of media services and examines current trends in the telecommunications and Internet industries to identify implications for the future of such services in IP networks, especially in Session Initiation Protocol (SIP)-based environments. One of the most exciting trends is the emergence and adoption of complementary modular standards that leverage the Internet to enable media services to be developed, deployed, and updated more rapidly than before in a network architecture that supports concepts we call provisioning on demand and scaling on demand. This paper discusses how these new standards facilitate these concepts and how the concepts can be applied to create a media server that requires no preprovisioned service-specific logic or service-specific resources (e.g., announcements and grammars). It also provides an overview of the Bell Labs media server prototype, which has been developed to support and explore this vision.

A SIP-enabled all-IP architecture for converged next-generation networks

The success of converged next-generation networks (NGNs) will be contingent on the ability of subscribers to access data, voice, and other real-time multimedia services seamlessly, whether they are roaming wireline wide area networks (WWANs) or wireless local area network (WLAN) hot spots. Session Initiation Protocol (SIP) is one of the underlying technologies required to implement an access-agnostic Internet Protocol (IP) multimedia network infrastructure. In this paper, we investigate the key challenges involved in supporting SIP-based real-time (e.g., voice over IP [VoIP]) and multimedia services over heterogeneous networks and we describe a SIP-enabled all-IP architecture (SEAA) that leverages 3rd Generation Partnership Project/3rd Generation Partnership Project2 (3GPP/3GPP2) IP Multimedia Subsystem (IMS) framework, and IETF's SIP profile to support access-independent NGNs. We develop a multilayered policy-based quality of service (QoS) control mechanism and propose an interaction mechanism involving the application service layer, the session control layer, and the transport network layer to support end-to-end consistent QoS and service level agreements (SLAs). We address a SIP-based session control mechanism and the implications for unified service management and describe in detail the end-to-end SIP call flow in an example policy-based network to demonstrate the validity of our solution. We also analyze the interworking architecture between SIP and legacy networks.

Network overload and congestion: A comparison of ISUP and SIP

For years, telephony service providers have supported voice calls using circuit networks controlled by ISDN User Part (ISUP) signaling. ISUP signaling is based on a mature standard, is widely deployed, supports a variety of services, and has grown to include extensive congestion control mechanisms. With the move to packet networks, Session Initiation Protocol (SIP) signaling is being introduced to support voice over packet networks. As service providers begin this migration, comparisons between SIP and ISUP signaling are inevitable. Specifically, service providers want assurance that SIP provides congestion control mechanisms for packet networks similar to those ISUP provides for circuit networks. This paper is a result of a study to anticipate the congestion control needs of packet networks based on Signaling System 7 (SS7) network experience. It provides a comparison of SIP and ISUP congestion control functions as well as an analysis of mechanisms that could be deployed in SIP networks to enable them to achieve parity with ISUP networks.

Transport protocol considerations for session initiation protocol networks

The session initiation protocol (SIP), an application-layer protocol in the Internet protocol (IP) model, is increasingly being used for signaling Internet telephony multimedia sessions. The widespread applicability and use of SIP has engendered the use of diverse transport-layer protocols underneath it (e.g., user datagram protocol [UDP], transmission control protocol [TCP], stream control transmission protocol [SCTP], transport layer security [TLS] over TCP, and TLS over SCTP). In carrier-grade networks, SIP must meet stringent scaling, reliability, and post-dial delay requirements. To some extent, these requirements can be met by optimally transporting SIP. The right choice of a transport protocol is primarily determined by the characteristics of the relevant transport protocols and by how those protocols interplay with individual SIP entities. Additionally, prudent management of transport-layer connections can help lower signaling latencies and facilitate scaling. This paper focuses on SIP transport as a decisive factor in engineering low-latency, scalable, carrier-grade SIP entities. It also presents the concept of SIP connection management and the persistent connection model.

Session initiation protocol: Service residency and resiliency

As the session initiation protocol (SIP) gets deployed, attention invariably needs to be paid to which SIP entities, user agents (UAs), proxies, back-to-back UAs (B2BUA), host services, and to what the impact is on the entity if the service causes it to fail. This paper discusses providing SIP services in terms of individual entities. We address how each of the entities participates in a SIP network and provides services in an individual capacity as well as part of the collective SIP network. © 2003 Lucent Technologies Inc.