The generic context sharing protocol GCSP : Application to signaling in a cross-network and multi-provider environment

This paper proposes a new signaling paradigm and a new signaling protocol called the Generic Context Sharing Protocol (GCSP) for the construction of a global control plane over present and future communication networks. After identifying the special nature of the control plane software involved in the setup of a conversational service instance it examines the various mechanisms for information sharing which leads to our new proposal. We show that this new data-based protocol is better suited to control plane requirements than the present day’s command-oriented signaling mechanisms. We indicate the basic principles of the protocol and we give a brief description of the generic context. We show the place of this proposal in the present day research efforts and we mention a practical implementation case.8th IFIP/IEEE International conference on Mobile and Wireless CommunicationRed de Universidades con Carreras en Informática (RedUNCI

The generic context sharing protocol GCSP : Application to signaling in a cross-network and multi-provider environment

This paper proposes a new signaling paradigm and a new signaling protocol called the Generic Context Sharing Protocol (GCSP) for the construction of a global control plane over present and future communication networks. After identifying the special nature of the control plane software involved in the setup of a conversational service instance it examines the various mechanisms for information sharing which leads to our new proposal. We show that this new data-based protocol is better suited to control plane requirements than the present day’s command-oriented signaling mechanisms. We indicate the basic principles of the protocol and we give a brief description of the generic context. We show the place of this proposal in the present day research efforts and we mention a practical implementation case.8th IFIP/IEEE International conference on Mobile and Wireless CommunicationRed de Universidades con Carreras en Informática (RedUNCI

Study of voice quality in IP networks

Orientador: Helio WaldmanDissertação (mestrado) - Universidade Estadual de Campinas, Faculdade de Engenharia Eletrica e de ComputaçãoAbstract: This work describes the study of voice quality in IP networks based on a revision of quality of service (QoS) protocols and mechanisms and aspects of the system impact associated with the presence or absence of them; revision of the diverse evaluation methods of voice quality with emphasis in the automatic methods (objective and repetitive) used to analyze the effects in the voice due to diverse factors presented in packet networks, such as packet loss, delay and jitter, as well as the proper voice coding at low rate; revision of the main protocols of signalling for implementation of voice over IP (VoIP) or IP telephony, considering its strong and weak points with regard to implementation facility, scalability and adequacy for some network applications and quality of service (QoS) and accomplishment of tests in simulated and experimental IP networks. The main objective is the characterization of voice service in IP networks taking into account the effect of the network factors in call set-up (connection) and in voice quality . For the simulation of the IP network the Shunra¿s Cloud software was used where it is possible to deal with, in isolated form, the influence of packet loss, fixed delay, delay variation ( jitter), as well as the composed effect of packet loss and jitter. Solutions to such effects are investigated in experimental tests. Results of system simulations are presented and discussed. Degradations in voice due to such effects are evaluated and a practical method to solve them is considered. The experimental results demonstrate the technical feasibility of using voice over IP (or IP telephony) by service providers, as well as private corporations being able to forward voice and data over the same converged IP networkResumo: Este trabalho descreve o estudo da qualidade de voz em redes IP a partir de uma revisão dos protocolos e mecanismos relativos a qualidade de serviço (QoS) e os aspectos do impacto sistêmico na presença ou ausência destes; revisão dos diversos métodos de avaliação da qualidade da voz com ênfase nos métodos automáticos (objetivos e repetitivos) para auxiliar na análise dos efeitos na voz dos diversos fatores presentes em uma rede de pacotes, tais como perda de pacote, atraso e jitter, bem como a própria codificação da voz em baixas taxas; revisão dos principais protocolos de sinalização utilizados para implementação de voz sobre IP (VoIP) ou telefonia sobre IP, evidenciando-se seus pontos fortes e fracos com relação a facilidade de implementação, extensibilidade e adequabilidade para várias aplicações de rede e qualidade de serviço (QoS) e realização de testes em redes IP simulada e experimental. O principal objetivo é a caracterização do serviço de voz em redes IP levando-se em consideração os efeitos dos fatores de rede e gateway no tempo de estabelecimento de uma chamada (conexão) e na qualidade da voz. Para simulação da rede IP foi utilizado o software Cloud da Shunra onde é possível tratar, de forma isolada, a influência da perda de pacote, do atraso fixo, do atraso variável (jitter), bem como do efeito conjunto da perda de pacote e jitter. Soluções a tais efeitos são investigadas em testes experimentais. Resultados de simulações sistêmicas são apresentados e discutidos. As degradações na voz devidas a tais efeitos são avaliadas e um método prático para solucionar é testado. Os resultados experimentais demonstram a viabilidade técnica da utilização da voz sobre IP (ou telefonia IP) pelos provedores de serviço, bem como pelas corporações privadas, podendo trafegar voz e dados em uma mesma rede convergente IPMestradoTelecomunicações e TelemáticaMestre em Engenharia Elétric

On the development of Voice over IP

This record of study documents the experience acquired during my internship at Sonus Networks, Inc. for the Doctor of Engineering Program. In this record of study, I have surveyed and analyzed the current standardization status of Voice over Internet Protocol (VoIP) security and proposed an Internet draft on secure retargeting and response identity. The draft provides a simple and comprehensive solution to the response identity, call recipient identity and intermediate server retargeting problems in the Session Initiation Protocol (SIP) call setup process. To support product line development and enable product evolution in the quickly growing VoIP market, I have proposed a generic development framework for SIP application servers. The common and open architecture of the framework supports multiple products development and facilitates integration of new service modules. The systematical reuse of proven software design and implementation enables companies to reduce the development cost and shorten the time-to-market. As the development and diffusion of VoIP can never be isolated from the social sphere, I have investigated the current status, influence and interaction of three most important factors: standardization, market forces and government regulation on the development and diffusion of VoIP. The worldwide deregulation and market privatization have caused the transition of the standards development model. This transition in turn influences the market diffusion. Other than standardization, market forces including customer needs, the revenue pressure on carriers and vendors, competitive and economic environment, social culture and regulation uncertainties create both threats and opportunities. I have examined market drivers and obstacles in the current VoIP adoption stage, analyzed current VoIP market players and their strategies, and predicted the direction of VoIP business. The regulation creates the macro environment in which VoIP develops and diffuses. I have explored modern telecommunications regulation principles based on which government makes decisions on most current issues, including 911 support, mergers and acquisitions, interconnection obligation and leasing rights, rate structure and universal service fees

A distributed intelligent network based on CORBA and SCTP

The telecommunications services marketplace is undergoing radical change due to the rapid convergence and evolution of telecommunications and computing technologies. Traditionally telecommunications service providers’ ability to deliver network services has been through Intelligent Network (IN) platforms. The IN may be characterised as envisioning centralised processing of distributed service requests from a limited number of quasi-proprietary nodes with inflexible connections to the network management system and third party networks. The nodes are inter-linked by the operator’s highly reliable but expensive SS.7 network. To leverage this technology as the core of new multi-media services several key technical challenges must be overcome. These include: integration of the IN with new technologies for service delivery, enhanced integration with network management services, enabling third party service providers and reducing operating costs by using more general-purpose computing and networking equipment. In this thesis we present a general architecture that defines the framework and techniques required to realise an open, flexible, middleware (CORBA)-based distributed intelligent network (DIN). This extensible architecture naturally encapsulates the full range of traditional service network technologies, for example IN (fixed network), GSM-MAP and CAMEL. Fundamental to this architecture are mechanisms for inter-working with the existing IN infrastructure, to enable gradual migration within a domain and inter-working between IN and DIN domains. The DIN architecture compliments current research on third party service provision, service management and integration Internet-based servers. Given the dependence of such a distributed service platform on the transport network that links computational nodes, this thesis also includes a detailed study of the emergent IP-based telecommunications transport protocol of choice, Stream Control Transmission Protocol (SCTP). In order to comply with the rigorous performance constraints of this domain, prototyping, simulation and analytic modelling of the DIN based on SCTP have been carried out. This includes the first detailed analysis of the operation of SCTP congestion controls under a variety of network conditions leading to a number of suggested improvements in the operation of the protocol. Finally we describe a new analytic framework for dimensioning networks with competing multi-homed SCTP flows in a DIN. This framework can be used for any multi-homed SCTP network e.g. one transporting SIP or HTTP

SIPBIO : biometrics SIP extension

During the last few decades biometric technologies have become an important research field in computer security. Their deployment, however, in heterogeneous enterprise systems, is complex due to the lack of standardisation. Session Initiation Protocol (SIP) is a popular communication protocol widely used in voice over Internet protocol networks; due to its flexibility, SIP has been broadly adopted in telecommunications for carrier level and telephony systems. This thesis proposes the use of SIPBIO, an extension to SIP, to establish and control multimedia sessions for biometric interactions. For biometric usage in telecommunications networks, a synthesis of techniques to use human characteristics as challenge tokens for access to network resources is first presented. An overview of the SIP protocol is then exposed, by focusing on understanding SIP messages and their component elements. Posteriorly, advanced concepts, such as extensions to the default protocol are introduced. After the technology background review, the core of the proposal is presented with extensive use-case scenarios of biometric operations and the introduction of necessary SIPBIO requirements. Formal processes are defined along with the method to extend SIP to the proposed SIPBIO protocol. It follows a detailed outline of all headers and body components that give form to SIPBIO and define its nature. These stages provide the fundamentals for the protocol implementation. Finally, simulations of some common cases are presented to show the feasibility of SIPBIO. This can be used as a sample flow for full implementations and applications. This thesis corroborates the viability of using a SIP-based protocol for establishing, maintaining and tearing down biometric multimedia sessions