Efficient User Controlled Inter-Domain SIP Mobility: Authentication, Registration, and Call Routing

Over the past decade, multimedia services have gained significant acceptance and played an important role in the convergence of IP networks. Supporting mobility in IP (Internet Protocol) networks is a crucial step towards satisfying the nomadic communication paradigms on the current Internet. The Session Initiation Protocol (SIP) presents one approach towards supporting IP mobility. Additionally, SIP is increasingly gaining in popularity as the next generation multimedia signaling and session establishment protocol. It is anticipated that the SIP infrastructure will be extensively deployed all over the Internet. In this paper, we explore an efficient approach to inter-domain SIP mobility in an attempt to improve personal and terminal mobility schemes. We succeed in applying a persistent identification framework to application level SIP addressing by introducing a level of indirection on top of the traditional SIP architecture. We refer to our approach as the Handle SIP (H-SIP). H-SIP leverages the current SIP architecture abstracting any domain binding from users. Our approach to mobility is user-controlled. We experimentally prove the efficiency of H-SIP in achieving inter-domain authentication and call routing through modeling and real-time measurements

Efficient User Controlled Inter-Domain SIP Mobility Authentication, Registration, and Call Routing

Over the past decade, multimedia services have gained significant acceptance and played an important role in the convergence of IP networks. The proliferation of mobile devices and the nomadic user and computing lifestyles on current networks make mobility support a crucial ingredient of current IP-based multimedia systems. The Session Initiation Protocol (SIP) presents one approach towards supporting IP mobility. Additionally, SIP is increasingly gaining in popularity as the next generation multimedia signaling and session establishment protocol, and the SIP infrastructure is anticipated to be extensively deployed all over the Internet. We have lately proposed an approach to inter-domain SIP mobility which we call H-SIP. H-SIP is a user-controlled mobility scheme that improves personal and terminal mobility. H-SIP uses persistent identifiers and leverages the traditional SIP architecture to abstract any domain binding from users. This paper expands on our previous work and experimentally proves the efficiency of H-SIP in achieving inter-domain authentication and call routing through modeling and real-time measurements

IPv6 mobility support for real-time multimedia communications: A survey

Mobile Internet protocol version 6(MIPv6) route optimization improves triangular routing problem that exists in MIPv4 environment.Route optimization of Session Initiation Protocol (SIP) over MIPv6 provides ef�cient real-time multimedia applications to users. This article provides a survey of SIP over MIPv6. We review the processes involved during the setting up of a SIP call and during mid-call SIP mobility. When SIP transmits real-time multimedia applications in a wireless environment, the mobile node (MN) may move from one access router (AR) to another AR, handing over control from one AR to the other. High handover latency degrades the quality of real-time multimedia applications due to the fact that real-time multimedia applications are delay-sensitive.Handover latency is an important issue to discuss.Reduction of handover latency can be made possible with the use of SIP's hierarchical registration. On the other hand, hybrid hierarchical and fast handover SIP's registration performs better compared to hierarchical registration. Finally, we present the directions for future research

Analisa dan Implementasi Session Initiation Protocol dan<br /> Real-Time Transport Protocol Pada Telephony Over<br /> Internet Protocol<br /> Analisys and Implementation of Session Initiation<br /> Protocol and Real-Time Transport Protocol to Telephony<br

ABSTRAKSI: SIP (Session Initiation Protocol) ialah protokol kontrol sinyal pada application layer yang membentuk, memodifikasi, dan menghentikan sesi multimedia seperti internet multimedia conferences, internet telephone calls, dan multimedia distribution. Protokol SIP merupakan protokol berbasis teks dan dapat dikembangkan dengan fitur dan layanan tambahan seperti layanan pengontrolan panggilan dan ketersediaan user, instant messages, mobility, dan interoperability dengan sistem telephony. Adanya teknologi ini diharapkan dapat memenuhi kebutuhan akan koneksi panggilan ke berbagai wilayah selama masih dalam cakupan jaringan internet. Protokol RTP (Real-Time Transport Protocol) membentuk fungsi yang tepat pada transportasi jaringan end-to-end system untuk aplikasi transmisi data real-time seperti audio, video, dan data simulasi melalui layanan jaringan multicast atau unicast. Penerapan Session Initiation Protocol sebagai protokol kontrol pensinyalan dan Real-Time Transport Protocol sebagai protokol yang mengatur pengiriman media pada system yang akan diimplementasikan. Penggunaan teknologi Internet Telephony dengan menggunakan protokol SIP dan RTP dintegrasikan dengan sistem administrasi akan lebih praktis dalam proses pembangunan dan pemeliharaan aplikasi tersebut. Analisis pada SIP dan RTP dilakukan dengan mengaplikasikan penggunaan Type of Service dan codec yang digunakan oleh klien dalam melakukan koneksi pada system sehingga akan didapat behavior pesan dan call flow SIP dan delay, jitter, dan konsumsi bandwidth pada RTP.Kata Kunci : Internet Telephony, SIP, RTP, linux.ABSTRACT: The Session Initiation Protocol is an application layer control signaling protocol for creating, modifying and terminating sessions, these include internet multimedia conferences, internet telephone calls and multimedia distribution. SIP is text-based protocol and can be developed with additional feature and service, these include call control service, presence, instant messages, mobility, and interoperability with other telephony system. This technology can fulfilled the needed of call connection to other area in internet coverage. The Real-time Transport Protocol provides end-to-end network transport functions suitable for applications transmitting real-time data such as audio, video or simulation data, over multicast or unicast network services. By using Session Initiation Protocol for control signaling protocol and RTP for controlling media transport on a system that will be implemented. Internet Telephony technology with SIP and RTP will be integrated with administration system will be more effective in the building and maintenance of the system. Analize processing to SIP and RTP with using of type of service and codecs that used by client in order to connecting to the system so it can be obtain SIP message behavior and call flow and RTP delay, jitter, and bandwidth compsumptions.Keyword: Internet Telephony, SIP, RTP, linux

Generic Models for Mobility Management in Next Generation Networks

In the network community different mobility management techniques have been proposed over the years. However, many of these techniques share a surprisingly high number of similarities. In this technical report we analyze and evaluate the most relevant mobility management techniques, pointing out differences and similarities. For macro-mobility we consider Mobile IP (MIP), the Session Initiation Protocol (SIP) and mobility management techniques typical of a GSM network; for micro-mobility we describe and analyze several protocols such as: Hierarchical MIP, TeleMIP, IDMP, Cellular IP and HAWAII

An IPtel architecture based on the SIP protocol

More and better accesses to the Internet increase the interest in using it to carry not only data but also voice and video. The IP Telephony (IPTel) was born in this context and offers a framework to create multimedia communication systems. The Session Initiation Protocol, used in the IPTel architecture, is a protocol for signaling and call control between two or more participants. This paper presents telephony over IP service. Different protocols typically used in IPtel are analyzed and the architecture and functionality of SIP protocol are explained. Different mobility modes provided by SIP through the application layer are described. Finally we present the sIPtel, a Java application that supports real audio and video communications and uses the SIP protocol for call signaling

Taxonomy and analysis of IP micro-mobility protocols in single and simultaneous movements scenarios

The micro-mobility is an important aspect in mobile communications, where the applications are anywhere and used anytime. One of the problems of micro-mobility is the hand-off latency. In this paper, we analyse two solutions for IP micro-mobility by means of a general taxonomy. The first one is based on the Stream Control Transmission Protocol (SCTP), which allows the dynamic address configuration of an association. The second one is based on the Session Initiation Protocol (SIP), which is the most popular protocol for multimedia communications over IP networks. We show that for the SCTP solution, there is room for further optimisations of the hand-off latency by adding slight changes to the protocol. However, as full end-to-end solution, SCTP is not able to handle simultaneous movement of hosts, whose probability in general cannot be neglected. On the other hand, the SIP can handle both single and simultaneous movements cases, although the hand-off latency can increase with respect to the SCTP solution. We show that for a correct and fast hand-off, the SIP server should be statefull

Multicast mobility support in session layer using SIP

Due to the development of computer hardware technology, the use of portable computers has become more popular. Research workers have become more and more interested in wireless mobility related to these new hardware technologies. Much work focusing on Mobile IP has been done. However, there exist some issues such as "Triangle Routing" in Mobile IP. It has been found that SIP-based (Session Initiation Protocol) mobility has advantages that can be used to avoid the issues in Mobile IP. Based on this idea, this thesis first introduces the previous work related to mobility as well as mobility with multicast, and presents some issues that exist in IP mobility with multicast. Then we provide an approach called "Multicast Mobility Support in Session Layer Using SIP", which is independent of the IP layer and automatically avoids the issues in IP layer. The structure of the SIP mobility with multicast is presented and a proposal to extend SIP functions for supporting SIP multicast mobility is provided. Based on the system model built by using SDL (ITU Specification and Description Language), we used ObjectGeode tool set to simulate the proposed idea presented in this thesis. This work may be helpful to shift the concept of multicasting and mobility from IP layer to session and application layers for real time multimedia

A Unified Mobility Management Architecture for Interworked Heterogeneous Mobile Networks

The buzzword of this decade has been convergence: the convergence of telecommunications, Internet, entertainment, and information technologies for the seamless provisioning of multimedia services across different network types. Thus the future Next Generation Mobile Network (NGMN) can be envisioned as a group of co-existing heterogeneous mobile data networking technologies sharing a common Internet Protocol (IP) based backbone. In such all-IP based heterogeneous networking environments, ongoing sessions from roaming users are subjected to frequent vertical handoffs across network boundaries. Therefore, ensuring uninterrupted service continuity during session handoffs requires successful mobility and session management mechanisms to be implemented in these participating access networks. Therefore, it is essential for a common interworking framework to be in place for ensuring seamless service continuity over dissimilar networks to enable a potential user to freely roam from one network to another. For the best of our knowledge, the need for a suitable unified mobility and session management framework for the NGMN has not been successfully addressed as yet. This can be seen as the primary motivation of this research. Therefore, the key objectives of this thesis can be stated as: To propose a mobility-aware novel architecture for interworking between heterogeneous mobile data networks To propose a framework for facilitating unified real-time session management (inclusive of session establishment and seamless session handoff) across these different networks. In order to achieve the above goals, an interworking architecture is designed by incorporating the IP Multimedia Subsystem (IMS) as the coupling mediator between dissipate mobile data networking technologies. Subsequently, two different mobility management frameworks are proposed and implemented over the initial interworking architectural design. The first mobility management framework is fully handled by the IMS at the Application Layer. This framework is primarily dependant on the IMS’s default session management protocol, which is the Session Initiation Protocol (SIP). The second framework is a combined method based on SIP and the Mobile IP (MIP) protocols, which is essentially operated at the Network Layer. An analytical model is derived for evaluating the proposed scheme for analyzing the network Quality of Service (QoS) metrics and measures involved in session mobility management for the proposed mobility management frameworks. More precisely, these analyzed QoS metrics include vertical handoff delay, transient packet loss, jitter, and signaling overhead/cost. The results of the QoS analysis indicates that a MIP-SIP based mobility management framework performs better than its predecessor, the Pure-SIP based mobility management method. Also, the analysis results indicate that the QoS performances for the investigated parameters are within acceptable levels for real-time VoIP conversations. An OPNET based simulation platform is also used for modeling the proposed mobility management frameworks. All simulated scenarios prove to be capable of performing successful VoIP session handoffs between dissimilar networks whilst maintaining acceptable QoS levels. Lastly, based on the findings, the contributions made by this thesis can be summarized as: The development of a novel framework for interworked heterogeneous mobile data networks in a NGMN environment. The final design conveniently enables 3G cellular technologies (such as the Universal Mobile Telecommunications Systems (UMTS) or Code Division Multiple Access 2000 (CDMA2000) type systems), Wireless Local Area Networking (WLAN) technologies, and Wireless Metropolitan Area Networking (WMAN) technologies (e.g., Broadband Wireless Access (BWA) systems such as WiMAX) to interwork under a common signaling platform. The introduction of a novel unified/centralized mobility and session management platform by exploiting the IMS as a universal coupling mediator for real-time session negotiation and management. This enables a roaming user to seamlessly handoff sessions between different heterogeneous networks. As secondary outcomes of this thesis, an analytical framework and an OPNET simulation framework are developed for analyzing vertical handoff performance. This OPNET simulation platform is suitable for commercial use

Vertical Handoff Characterization for SIP and mSCTP Based UMTS-WLAN Integration Solutions

It is desirable to integrate 3G Universal Mobile Telecommunication System (UMTS) and 802.11 wireless local area networks, especially at hot-spot locations such as hotels and airports. The efficiency of wireless data services can be maximized if the integration provides users with seamless roaming across the two types of networks. Seamless handoff between these two networks to maintain session continuity is a major challenge in WLAN-3G integration. To achieve this goal, integration architectures together with mobility solutions such mobile stream control transmission protocol (mSCTP) and session initiation protocol (SIP) have been proposed in the literature. In this paper, we implement through simulations an integration architecture and characterize the vertical handoff delay for both mobility solutions mSCTP and SIP as a function of network parameters. This study finds that mSCTP perform better in terms of handoff delay compared to SIP for the assumptions specified in this paper