The Session Initiation Protocol (SIP) was first published in 1999, by the Internet Engineering Task Force (IETF), to be the standard for multimedia transfers. SIP is a peer-to-peer signaling protocol that is capable of initiating, modifying, and terminating media sessions. SIP utilizes existing Internet Protocols (IP) such as Domain Name Service (DNS) and the Session Description Protocol (SDP), allowing it to seamlessly integrate into existing IP networks. As SIP has matured and gained acceptance, its deficiencies when functioning as a multipoint communications protocol have become apparent. SIP currently supports two modes of operation referred to as conferencing and multicasting. Conferencing is the unicast transmission of session information between conference members. Multicasting uses IP multicast to distribute session information. This thesis proposes an extension for the Session Initiation Protocol that improves functionality for multipoint communications. When using conferencing, a SIP user-agent has limited information about the conference it is taking part in. This extension increases the awareness of a SIP node by providing it with complete conference membership information, the ability to detect neighboring node failures, and the ability to automatically repair conference partitions. Signaling for conferencing was defined and integrated into a standard SIP implementation where it was used to demonstrate the above capabilities. Using a prototype implementation, the additional functionality was shown to come at the cost of a modest increase in transaction message size and processing complexity. IP multicast has limited deployment in today's networks reducing the usability of this useful feature. Since IP multicast support is not guaranteed, the use of application layer multicast protocols is proposed to replace the use of IP multicast. An efficient means of negotiating an application layer protocol is proposed as well as the ability to provide the protocol with session information to begin operation. A ring protocol was defined and implemented using the proposed extension. Performance testing revealed that the application layer protocol had slightly higher processing complexity than conferencing, but on average had a smaller transaction message size. / Master of Science

Estágio realizado na Novabase e orientado pelo Eng.º Pedro Faúlha / Tese de mestrado integrado. Engenharia Informátca e Computação. Faculdade de Engenharia. Universidade do Porto. 2008

Thesis (M. Sc.)--University of Canterbury, 2008. / Typescript (photocopy). Includes bibliographical references (leaves 103-110). Also available via the World Wide Web.

Thesis (MS)--Montana State University--Bozeman, 2009. / Typescript. Chairperson, Graduate Committee: Brock LaMeres. Includes bibliographical references (leaves 44-47).

The switched telephony network was designed for voice calls. Expansion of datacommunication has lead to a wide range of experimentation to create new services. Theses services take place outside the network. When adding new services we currently encounter problems due to limitations of the simple devices at end points. Theo Kanter has proposed a new model to remove these limitations; this model is called “Adaptive Personal Mobile Communication”. The model consists of several components in the application layer of ISO standard. This model is based on peer to peer connections and the purpose of this model is to move services from within the networks to end point devices and avoid using central servers within the network. The Session Initiation Protocol (SIP) for establishing multimedia sessions allows us to move the point of integration for multimedia service integration out to the end-points. This project concerns the implementing of a prototype of this model as an SIP extension along with it evaluation. SIP offer addressing, naming, and localization of resources in this project. This report presents different design alternatives for XSP as an SIP extension, and the chosen model presents as a result of comparing of these design alternatives.

<p>SIP is receiving much attention these days and it seems to be the most promising candidate as a signaling protocol for the current and future IP telephony services. Realizing this, there is the obvious need to provide a certain level of quality comparable to the traditional telephone service signalling system. Thus, we identified the major costs of SIP, which were found to be delay and security. This thesis discusses the costs of SIP, the solutions for the major costs, and the development of a low cost SIP application. The literature review of the components used to develop such a service is discussed, the networks in which the SIP is used are outlined, and some SIP applications and services previously designed are discussed. A simulation environment is then designed and implemented for the instant messaging service for wireless devices. This environment simulates the average delay in LAN and WLAN in different scenarios, to analyze in which scenario the system has the lowest costs and delay constraints.</p>

This thesis looks in depth at telephony server clusters, the modern switchboards at the core of a packet-based telephony service. The most widely used de facto standard protocols for telecommunications are the Session Initiation Protocol (SIP) and the Real Time Protocol (RTP). SIP is a signaling protocol used to establish, maintain, and tear down communication channel between two or more parties. RTP is a media delivery protocol that allows packets to carry digitized voice, video, or text. SIP telephony server clusters that provide communications services, such as an emergency calling service, must be scalable and highly available. We evaluate existing commercial and open source telephony server clusters to see how they differ in scalability and high availability. We also investigate how a scalable SIP server cluster can be built on a cloud computing platform. Elasticity of resources is an attractive property for SIP server clusters because it allows the cluster to grow or shrink organically based on traffic load. However, simply deploying existing clusters to cloud computing platforms is not good enough to take full advantage of elasticity. We explore the design and implementation of clusters that scale in real-time. The database tier of our cluster was modified to use a scalable key-value store so that both the SIP proxy tier and the database tier can scale separately. Load monitoring and reactive threshold-based scaling logic is presented and evaluated. Server clusters also need to reduce processing latency. Otherwise, subscribers experience low quality of service such as delayed call establishment, dropped calls, and inadequate media quality. Cloud computing platforms do not guarantee latency on virtual machines due to resource contention on the same physical host. These extra latencies from resource contention are temporary in nature. Therefore, we propose and evaluate a mechanism that temporarily distributes more incoming calls to responsive SIP proxies, based on measurements of the processing delay in proxies. Availability of SIP server clusters is also a challenge on platforms where a node may fail anytime. We investigated how single component failures in a cluster can lead to a complete system outage. We found that for single component failures, simply having redundant components of the same type are enough to mask those failures. However, for client-facing components, smarter clients and DNS resolvers are necessary. Throughout the thesis, a prototype SIP proxy cluster is re-used, with variations in the architecture or configuration, to demonstrate and address issues mentioned above. This allows us to tie all of our approaches for different issues into one coherent system that is dynamically scalable, is responsive despite latency varations of virtual machines, and is tolerant of single component failures in cloud platforms.

<p>Session Initiation Protocol (SIP) arises as a new standard of establishing and releasing connections for vast variety of multimedia applications. The protocol may be used for voice calls, video calls, video conferencing, gaming and many more.</p><p>The 3GPP (3^rd Generation Partnership Project) suggests SIP as the signalling solution for 3^rd generation telephony. Thereby, this purely IP-centric protocol appears as a promising alternative to older signalling systems such as H.323, SS7 or analog signals in PSTN. In contrast to them, SIP does not focus on communication with PSTN network. It is more similar to HTTP than to any of the mentioned protocols. </p><p>The main standardisation body behind Session Initiation Protocol is The Internet Engineering Task Force (IETF). The most recent paper published on SIP is RFC 3261 [5]. Moreover, there are working groups within IETF that publish suggestions and extensions to the main standard. One of those extensions is “Caller Preferences for the Session Initiation Protocol (SIP)” [1]. </p><p>This document describes a set of new rules that allow a caller to express preferences about request handling in servers. They give ability to select which Uniform Resource Identifiers (URI) a request gets routed to, and to specify certain request handling directives in proxies and redirect servers. It does so by defining three new request header fields, Accept-Contact, Reject-Contact, and Request-Disposition, which specify the caller preferences. [1]. </p><p>The aim of this project is to extend the existing software with caller preferences and evaluate the new functionality.</p>

<p>SIP is receiving much attention these days and it seems to be the most promising candidate as a signaling protocol for the current and future IP telephony services. Realizing this, there is the obvious need to provide a certain level of quality comparable to the traditional telephone service signalling system. Thus, we identified the major costs of SIP, which were found to be delay and security. This thesis discusses the costs of SIP, the solutions for the major costs, and the development of a low cost SIP application. The literature review of the components used to develop such a service is discussed, the networks in which the SIP is used are outlined, and some SIP applications and services previously designed are discussed. A simulation environment is then designed and implemented for the instant messaging service for wireless devices. This environment simulates the average delay in LAN and WLAN in different scenarios, to analyze in which scenario the system has the lowest costs and delay constraints.</p>

Magister Scientiae - MSc / SIP is receiving much attention these days and it seems to be the most promising candidate as a signaling protocol for the current and future IP telephony services. Realizing this, there is the obvious need to provide a certain level of quality comparable to the traditional telephone service signalling system. Thus, we identified the major costs of SIP, which were found to be delay and security. This thesis discusses the costs of SIP, the solutions for the major costs, and the development of a low cost SIP application. The literature review of the components used to develop such a service is discussed, the networks in which the SIP is used are outlined, and some SIP applications and services previously designed are discussed. A simulation environment is then designed and implemented for the instant messaging service for wireless devices. This environment simulates the average delay in LAN and WLAN in different scenarios, to analyze in which scenario the system has the lowest costs and delay constraints. / South Africa