Recursos VoIP - Enlaces

Fuente: Nortel Networks
Formato: PDF

Nortel Networks commissioned The Tolly Group, the IT industry's foremost independent evaluator of Enterprise-class IP Telephony equipment, to observe and to validate the migration of a Meridian 1 PBX to the Succession 1000M IP PBX. The goal was to chronicle the effort required to transition the Meridian 1 to the Succession 1000M, plus identify the hardware and software components that are necessary to make the migration a success.

Fuente: ShoreTel
Formato: PDF

InfoTech conducted in-depth interviews with three enterprises regarding their IP telephony deployments. These enterprises span financial services, education, and professional services markets. The intent of the research was to focus on the challenges and success factors encountered after deploying an IP telephony solution. According to InfoTech, the ability of an enterprise to achieve the desired business benefits from VoIP is directly tied to three inter-related activities:

1. Choosing the converged solution vendor and support model
2. Deciding on a specific IP Telephony solution
3. Implementing the IP Telephony solution

Read this report to identify migration strategies and see candid insights from companies that have deployed VoIP solutions.

Fuente: Avaya
Formato: PDF

The business benefits of converged communications come hand-in-hand with additional technological complexity that poses challenges- and potential rewards - to IT organizations. Incorporating the experience and lessons learned from businesses that have successfully implemented IP Telephony (IPT), this Executive Briefing recommends a new organizational approach for planning, implementing, and managing converged networks.

Fuente: PowerDsine
Formato: PDF

Power over LAN is a revolutionary technology that integrates data, voice and power over standard LAN infrastructure. It is the means to supply reliable, uninterrupted power to Internet Protocol (IP) telephones, wireless LAN access points, network cameras and other Ethernet devices, using existing, commonly used Category 3 (4 pairs) and Category 5 cable infrastructure. This paper provides an overview of this technology and describes how to install IP Telephony network using Power over LAN technology.

Fuente: AT&T
Formato: PDF

Whereas the first wave of enterprise convergence projects focused on creating a single infrastructure based on the Internet protocol (IP), companies are now focusing on the challenge of integrating major business applications with real-time communications running over IP. This is one of the main findings from a survey of 236 executives (including 77 CEOs) on network convergence issues conducted in 2005 by The Economist Intelligence Unit in co-operation with AT&T.

The analysis in this white paper suggests that it will become increasingly difficult to divide communications and information processing into separate categories, as organizations load as much corporate traffic as possible onto their IP-based.

Fuente: VCON
Formato: PDF

IP Multicasting is defined (RFC 1112, ‘Host Extension for IP Multicasting’, Steve Deering, 1989) as the transmission of datagrams (UDP packets) to a set of hosts through a single IP destination address. Each member of the set, known as a host group, then picks up the datagrams from the host-group address. Because IP Multicasting uses a virtual IP destination address to reach the host group, the sender does not need to know the connection details of, or have a direct connection to, all the members. Unicast, on the other hand, requires that the identification of each connection be known. Comparing the two methods of transferring data to multiple users emphasizes the benefits of IP Multicasting. While the Unicast method sends multiple copies of data, one copy for each receiver, IP Multicast sends only one copy for all of the receivers at once. Thus, multicast transmission reduces traffic load significantly, allowing multi-point conferencing to occur over any existing network infrastructure, typically with little extra cost or replacement of equipment. Bandwidth is more efficiently utilized, resources are saved, and datagrams are transmitted with less processing.

Fuente: International Engineering Consortium
Formato: PDF

Internet telephony refers to communications services—voice, facsimile, and/or voice-messaging applications—that are transported via the Internet, rather than the public switched telephone network (PSTN). The basic steps involved in originating an Internet telephone call are conversion of the analog voice signal to digital format and compression/translation of the signal into Internet protocol (IP) packets for transmission over the Internet; the process is reversed at the receiving end. This tutorial discusses the ongoing but rapid evolution of Internet telephony, the market forces fueling that evolution and the benefits that users can realize, as well as the underlying technologies. It also examines the hurdles that must be overcome before Internet telephony can be adopted on a widespread basis.

Fuente: University of Washington
Formato: WORD

During the past three years, Voice-over-IP (or VOIP) has evolved from a ham-radio hacker toy to one of the numerous "next great things" facilitated by the Internet. Today, Net calling not only works between PCs, connected via the Internet, but also between standard phones in most parts of the world. In this technology’s corner are several Internet heavyweights like Cisco Systems and Lucent Technologies who bring VOIP tremendous credibility, as well as start-ups like Qwest and Level 3 Communications, who are currently building out nationwide next-generation data networks powered by the Internet. Clearly, this is a technology that has a great deal of money and exciting momentum behind it.

Fuente: Nuera Communications
Formato: PDF

Service Providers have a choice of implementing ETSI-TIPHON OSP (Open Settlement Protocol) and/or ITU-T H.225 Annex G based protocol for VoIP (Voice Over Internet Protocol) applications. In order for services to work in an environment that encompasses both protocols, as for example in an end-to-end service that involves connectivities among multiple SP’s, an inter-working function must be available. This document addresses SP to SP interoperability when dealing with exchange of inter-domain pricing, authorization and settlement information between different SP’s using OSP and Annex G. Mapping between messages between the two protocols is a key function of such interoperability.

Fuente: University of Waterloo
Formato: PDF

This article discusses technical issues related to delivery and control of IP multimedia services, such as video-conferencing, involving heterogeneous end terminals. In particular, it describes the design and implementation of an experimental system for interworking between IETF SIP (Session Initiation Protocol) and ISO MPEG-4 DMIF (Delivery Multimedia Integration Framework) session and call control signaling protocols. This IP videoconferencing interworking system is composed of two core units for supporting delivery of audio-video streams from a DMIF domain to a SIP domain (i.e. DMIF2SIP unit) and from a SIP domain to a DMIF domain (i.e. SIP2DMIF unit). These units perform various translation functions for transparent establishment and control of multimedia sessions across IP networking environment, including, session protocol conversion, service gateway conversion and address translation.

Fuente: Cisco Systems
Formato: PDF

The Interworking Signaling Enhancements for H.323 and SIP VoIP feature enables VoIP networks to properly signal the setup and tear-down of calls, including generating in-band tones and announcements when needed at the originating or terminating switch. When a tone (for example, ringback, busy, reorder) or announcement (for example, "The number you have dialed is no longer in service") is played at the destination switch, the backward voice path from the called party to the calling party is cut-through early, so that the calling party can hear the tone or announcement.

Fuente: Performance Technologies
Formato: PDF

A VoIP network carries voice traffic cheaper than a switched circuit telephone network because IP telephony networks make better use of available bandwidth. In a public switched telephone network, for example, a dedicated 64 kilobits per second (kbps) end-to-end circuit is allocated for each call. In a VoIP network, digitized voice data is highly compressed and carried in packets over IP networks. Using the same bandwidth, a VoIP network can carry many times the number of voice calls as a switched circuit network with better voice quality. The savings realized in using VoIP networks are often passed onto users in the form of lower costs.

Fuente: Network General
Formato: PDF

This is the first of six technical briefing papers that examine the concepts, operation and analysis of Voice over Internet Protocol (VoIP) and converged networks. The concept of a converged network - one that combines voice, data, and other signal transmissions into a single, higher-speed network interface - has been around for several decades. We will look at:
- What is a converged network.
- Enabling technologies and standards.
- Migrating the legacy network.
- Benefits and applications for the enterprise.
- Challenges for converged networks.

Fuente: Anixter
Formato: PDF

Voice and data integration is currently a hot topic within the business community, with its promise of major cost savings and consolidation of data and voice infrastructures. But there remains some scepticism and the old adage ‘if it ain’t broke, don’t fix it’ certainly has a few adherents. However, while PBX based telephone systems have certainly proved their worth in the past, there is growing evidence that the demands of modern business will increasingly necessitate the implementation of integrated voice and data systems. Up to now, small sites were the ones that showed themselves willing to adopt the new standard, but now it is the large enterprise sites with existing PBX based systems that are beginning to be seduced by the significant business and cost benefits that a migration to VoIP technology can bring. The aim of this paper is to explain how and why this migration can be carried out, and to highlight to the user the myriad of benefits that such a migration entails.

Fuente: Cisco Systems
Formato: PDF

As deployments of voice over IP (VoIP) networks continue at a rapid pace, network professionals need to understand the real-world implementation issues surrounding voice gateways, which handle the many tasks involved in translating between transmission formats and protocols and act as the interface between an IP telephony network and the PSTN or PBX. Focus on the Session Initiation Protocol (SIP) in this sample chapter from Cisco Voice Gateways and Gatekeepers. SIP is an Internet Engineering Task Force (IETF) standard call control protocol designed to provide signaling and session management for voice and multimedia connections over packet-based networks. It is a peer-to-peer protocol with intelligent endpoints and distributed call control. SIP performs four basic tasks:

- Locating users, resolving their SIP address to an IP address.

- Negotiating capabilities and features among all the session participants.

- Changing the session parameters during the call.

- Managing the setup and teardown of calls for all users in the session.

Learn how SIP works and how to configure SIP gateways. Investigate SIP call flow, SIP pros and cons, dial plan considerations, security issues, and troubleshooting tools.