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

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

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

Programming Internet Telephony Services

Internet telephony enables a wealth of new service possibilities. Traditional telephony services, such as call forwarding, transfer, and 800 number services, can be enhanced by interaction with email, web, and directory services. Additional media types, like video and interactive chat, can be added as well. One of the challenges in providing these services is how to effectively program them. Programming these services requires decisions regarding where the code executes, how it interfaces with the protocols that deliver the services, and what level of control the code has. In this paper, we consider this problem in detail. We develop requirements for programming Internet telephony services, and we show that at least two solutions are required --- one geared for service creation by trusted users (such as administrators), and one geared for service creation by untrusted users (such as consumers). We review existing techniques for service programmability in the Internet and in the telephone network,and extract the best components of both. The result is a Common Gateway Interface (CGI) that allows trusted users to develop services, and the Call Processing Language (CPL) that allows untrusted users to develop services

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

An analysis of IP Telephony Signaling using the Session Initiation Protocol (SIP)

This paper examines both the emergence of IP telephony in the telecommunications industry and the Session Initiation Protocol (SIP) as a method for providing signaling services for IP telephony networks. The technical as well as cost advantages of IP telephony are addressed and the various SIP components, addressing mechanisms, protocol messages, and protocol functionality are discussed. The technical benefits of SIP are examined and a brief comparison is made between SIP and H.323, an ITU umbrella specification and a leading alternative to SIP. SIP and the latest version of H.323 are found to be relatively comparable, although due to the inherent simplicity in developing SIP implementations and applications, SIP has the potential to challenge H.323's dominance in the IP telephony signaling market space

An asynchronous time division multiplexing scheme for voice over IP.

by Yip Chung Sun Danny.Thesis (M.Phil.)--Chinese University of Hong Kong, 2000.Includes bibliographical references (leaves 52-54).Abstracts in English and Chinese.Chapter Chapter 1 --- Introduction --- p.1Chapter 1.1 --- Motivation --- p.1Chapter 1.2 --- Organization of Thesis --- p.5Chapter Chapter 2 --- Background --- p.6Chapter 2.1 --- Speech Codec --- p.6Chapter 2.2 --- RTP/UDP/IP Header Compression --- p.7Chapter 2.2.1 --- Real-Time Transport Protocol --- p.7Chapter 2.2.2 --- RTP/UDP/IP Header Compression --- p.8Chapter Chapter 3 --- Scenario and Assumptions --- p.10Chapter Chapter 4 --- Asynchronous Time Division Multiplexing Scheme --- p.14Chapter 4.1 --- Basic Idea --- p.14Chapter 4.1.1 --- Bandwidth Efficiency Improvement --- p.16Chapter 4.1.2 --- Delay Reduction --- p.18Chapter 4.2 --- Header Compression --- p.19Chapter 4.2.1 --- Header Compression Process --- p.21Chapter 4.2.2 --- Context Mapping Table --- p.23Chapter 4.3 --- Protocol --- p.28Chapter 4.3.1 --- UNCOMPRESSED_RTP Mini-Header --- p.30Chapter 4.3.2 --- SYNCHRONIZATION Mini-header --- p.31Chapter 4.3.3 --- COMPRESSED´ؤRTP Mini-header --- p.32Chapter 4.4 --- Connection Establishment --- p.33Chapter 4.4.1 --- Addressing Phase --- p.34Chapter 4.4.2 --- Connection Phase --- p.36Chapter 4.5 --- Software Implementation --- p.38Chapter Chapter 5 --- Simulation Results --- p.39Chapter 5.1 --- Simulation Model --- p.39Chapter 5.2 --- Voice Source Model --- p.41Chapter 5.3 --- Simulation Results --- p.43Chapter 5.3.1 --- Network Utilization and Delay Performance --- p.43Chapter 5.3.2 --- Number of Supported Connections --- p.45Chapter Chapter 6 --- Conclusion and Future Work --- p.49Bibliography --- p.5

Protocolos para telefonia IP

Orientador: Nelson Luis Saldanha da fonsecaDissertação (mestrado) - Universidade Estadual de Campinas, Instituto de ComputaçãoResumo: A telefonia IP, também chamada de VoIP (Voice over IP), pode ser definida como qualquer aplicação telefônica usada em uma rede de comutação de pacotes de dados que utiliza o protocolo Internet Protocol (IP). Engloba novas aplicações que exploram a integração da comunicação de voz, imagens e de dados simultaneamente. Protocolos vêm sendo propostos para telefonia IP. No entanto, um grande desafio a ser transposto por estes protocolos é a garantia de qualidade de voz similar à da telefonia comutada por circuitos. Este trabalho apresenta os protocolos H.323, SIP, MGCP e Megaco/H.248 para telefonia IP, faz uma comparação destes protocolos e aborda fatores que afetam a Qualidade de Serviço (QoS) de telefonia IPAbstract: IP telephony can be defined as any telephonic application over the Internet Protocol and is one of the new applications that explore the integration of voice, image and data communication. Protocols have been proposed for IP telefony. However, one of the challenges in the IP telephony is to assure that the voice quality has similar quality of the one in circuit-switched telephony. This work presents the protocols H323, SIP, MGCP and MegacoIH.248 for IP telephony and compare them. It also describes the issues which impact the Quality of Service (QoS) in IP telephonyMestradoEngenharia de ComputaçãoMestre em Computaçã

Multiparty/Multimedia Conferencing in Mobile Ad-Hoc Networks for Improving Communications between Firefighters

In current practice, firefighters’ communications systems are verbal, using a simplex Radio Frequency (RF) system (walkie-talkie). They use a push-to-talk mechanism in which only one person can talk at any time and all other firefighters will hear the messages. They use special codes (e.g. 1008, 1009, etc.) to express their current situation. Firefighters of the same team need to be in visual contact with each other at all times. This RF system does not support other functionalities (e.g. video communications, conference calls). In addition, because communication between firefighters is a flat structure, private communications is not possible. Mobile Ad-Hoc Networks (MANETs) are infrastructure-less and self-organized wireless networks of mobile devices, which are not based on any centralized control. MANETs are suitable for the hosting of a wide range of applications in emergency situations, such as natural or human-induced disasters, and military and commercial settings. Multimedia conferencing is an important category of application that can be deployed in MANETs. This includes well-known sets of applications, such as audio/video conferencing, data communications, and multiplayer games. Conferencing can be defined as the conversational exchange of data content between several parties. Conferencing requires, at the very least, the opening of two sessions: a call signaling session, and a media handling session. Call signaling is used to set up, modify, and terminate the conference. Media handling is used to cover the transportation of the media, and to control/manage the media mixers and media connections. So far, very little attention has been devoted to the firefighters’ communication system. In the present work, we focus on building a new communication system for firefighters using multimedia conferencing/sub-conferencing in MANETs. The background information for the firefighters’ current communications system and MANETs, along with the multimedia conferencing, is provided. The limitations of this system are determined, and the requirements are derived to determine the functionalities of a better communication system that will overcome current limitations. We have proposed a cluster-based signaling architecture that meets our requirements. We have also identified a state-of-the-art media handling and mixing system that meets most of our requirements, and have adapted it to inter-work with our signaling system. We have implemented the proposed architecture using SIP signaling protocol. Performance measurements have been performed on the prototype. Through experiments, we have found that the new multimedia communication system is a very promising approach to solve the current firefighters’ communication problems