A new scheme to reduce session establishment time in session initiation protocol (SIP)

The session Initiation Protocol (SIP) has been developed by Internet Engineering Taskforce standard (IETF) with the main purpose of establishing and managing sessions between two or more parties wishing to communicate. SIP is a signaling protocol which is used for the current and future Internet Protocol (IP) telephony services, video services, and integrated web and multimedia services. SIP is an application layer protcol, thus it can run over Transmission Control Protocol(TCP) or User Datagram Protocol (UDP). When the packets are sent over the network, a form of congestion control mechanism is necessary to prevent from network collapse. TCP is a reliable protocl and provides the congestion control by adjusting the size of the congestion windows. UDP is an unreliable protocol and no flow control mechanism is provided. Many applications of the Internet require the establishment and management of sessions. The purpose of the thesis is to study the session establishnment procedure in SIP and try to reduce the time taken for the session setup in two different conditions. One, when there is no congestion in the network, and the other is when there is a network congestion. We have simulated the behaviour of session establishment in SIP using Network Simulator (NS2). UDP is used as the transport protocol. We have created different network topologies. In the topology we had created SIP user agents who wants to communicte, proxy servers for forwarding the requests on behalf of the user agents, and a Domain Name Server (DNS) which maintains the location information of all proxy servers. We tried to reduce the time taken for the session establishment. As UDP does not provide any congestion control mechanisms, we used the binary exponential backoff (BEB) algorithm to set the timers. In our network topolgy when there is no packet loss in the network, the time taken for the session establishment is reduced from 0.86 sec to 0.574 sec. In case of network congestion the setup time is reduced from 4.55 sec to 2.86 sec. From the simulation, we conclude that the session establishment time can be reduced by reducing the number of message exchanges required for session setup

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

Open Access to Resource Management in Multimedia Networks

The paper is dedicated to mechanisms for open access to resource management in the Internet Protocol (IP) multimedia networks. First we present the concept of IP Multimedia Subsystem (IMS) and explain the IMS functional architecture, principles of quality of service management and service control in IMS. Then we describe the idea behind the opening of network interfaces for third parties so that others besides the network operator can create and deploy services. Open Service Access (OSA) and Parlay appear to be the technologies for value-added service delivery in multimedia networks. In the paper we take a closer look to the Parlay/OSA interfaces that allow third party applications to access the resource management functions in IMS. OSA "Connectivity Manager" interfaces and OSA "Policy Management" interfaces are considered. Parlay X Web Services interfaces provide a higher level of abstraction than Parlay/OSA interfaces and gain an amazing amount of support among service developers. We address "Applicationdriven Quality of Service" Parlay X Web Service and "Policy" Parlay X Web Service also

Signaling for Internet Telephony

Internet telephony must offer the standard telephony services.However, the transition to Internet-based telephony services also provides an opportunity to create new services more rapidly and with lower complexity than in the existing public switched telephone network(PSTN). The Session Initiation Protocol (SIP) is a signaling protocol that creates, modifies and terminates associations between Internet end systems, including conferences and point-to-point calls. SIP supports unicast, mesh and multicast conferences, as well as combinations of these modes. SIP implements services such as call forwarding and transfer, placing calls on hold, camp-on and call queueing by a small set of call handling primitives. SIP implementations can re-use parts of other Internet service protocols such as HTTP and the Real-Time Stream Protocol (RTSP). In this paper, we describe SIP, and show how its basic primitives can be used to construct a wide range of telephony services

Interworking Between SIP/SDP and H.323

There are currently two standards for signaling and control of Internet telephone calls, namely ITU-T Recommendation H.323 and the IETF Session Initiation Protocol (SIP). We describe how an interworking function (IWF) can allow SIP user agents to call H.323 terminals and vice versa. Our solution addresses user registration, call sequence mapping and session description. We also describe and compare various approaches for multi-party conferencing and call transfer

A method for forensic artifact collection, analysis and incident response in environments running Session Initiation Protocol (SIP) and Session Description protocol

In this paper, we perform an analysis of SIP, a popular voice over IP (VoIP) protocol and propose a framework for capturing and analysing volatile VoIP data in order to determine forensic readiness requirements for effectively identifying an attacker. The analysis was performed on real attack data and the findings were encouraging. It seems that if appropriate forensic readiness processes and controls are in place, a wealth of evidence can be obtained. The type of the end user equipment of the internal users, the private IP, the software that is used can help build a reliable baseline information database. On the other hand the private IP addresses of the potential attacker even during the presence of NAT services, as well as and the attack tools employed by the malicious parties are logged for further analysis

Service composition based on SIP peer-to-peer networks

Today the telecommunication market is faced with the situation that customers are requesting for new telecommunication services, especially value added services. The concept of Next Generation Networks (NGN) seems to be a solution for this, so this concept finds its way into the telecommunication area. These customer expectations have emerged in the context of NGN and the associated migration of the telecommunication networks from traditional circuit-switched towards packet-switched networks. One fundamental aspect of the NGN concept is to outsource the intelligence of services from the switching plane onto separated Service Delivery Platforms using SIP (Session Initiation Protocol) to provide the required signalling functionality. Caused by this migration process towards NGN SIP has appeared as the major signalling protocol for IP (Internet Protocol) based NGN. This will lead in contrast to ISDN (Integrated Services Digital Network) and IN (Intelligent Network) to significantly lower dependences among the network and services and enables to implement new services much easier and faster. In addition, further concepts from the IT (Information Technology) namely SOA (Service-Oriented Architecture) have largely influenced the telecommunication sector forced by amalgamation of IT and telecommunications. The benefit of applying SOA in telecommunication services is the acceleration of service creation and delivery. Main features of the SOA are that services are reusable, discoverable combinable and independently accessible from any location. Integration of those features offers a broader flexibility and efficiency for varying demands on services. This thesis proposes a novel framework for service provisioning and composition in SIP-based peer-to-peer networks applying the principles of SOA. One key contribution of the framework is the approach to enable the provisioning and composition of services which is performed by applying SIP. Based on this, the framework provides a flexible and fast way to request the creation for composite services. Furthermore the framework enables to request and combine multimodal value-added services, which means that they are no longer limited regarding media types such as audio, video and text. The proposed framework has been validated by a prototype implementation