Recursos VoIP - Enlaces

Fuente: Hewlett-Packard
Formato: PDF

Extensive call-quality tests conducted independently by The Tolly Group in September 2003 show that Hewlett-Packard Co.'s ProCurve enterprise LAN switching infrastructure is a fertile network transport for voice over IP (VoIP) traffic generated by a diverse set of IP PBX products from multiple vendors. HP ProCurve Networking products gracefully supports call-quality levels on par with, or better than, those provided by a pure Cisco Catalyst network or a pure 3Com Corp. switch fabric when transporting VoIP alongside data traffic.

Fuente: 3Com
Formato: PDF

The demanding and forward-looking enterprise chooses 3Com solutions because they work well today and are ready for the future. 3Com bases its products on advanced technologies, designs them for efficient technology migration and service expansion, and tests them rigorously for reliability, performance, and interoperability. 3Com deliberately designs its products to decrease network complexity and raise network performance, availability, and user productivity. It helps remove superfluous product costs and drive down the enterprise's recurring network overhead - including administration, management, service, and support staff expenses.

Fuente: Columbia University Department of Computer Science
Formato: PDF

This contribution compares and contrasts SIP to H.323v4 to help aid operators and vendors in the selection of a single least common denominator control protocol for the ps “domain” or perhaps more appropriately “plane” of UMTS Release 2000. The format anticipates the concerns, in the form of questions, which may arise from 3GPP members. Two standards have emerged for signaling and control of VoIP telephony: ITU-T H.323 and the IETF Session Initiated Protocol (SIP). These protocols, although resulting in the same end-user service (telephony), differ in the approach to providing signaling functions. H.323 is based more on a monolithic bloc derived from H.320 for traditional of the traditional circuit-switched ISDN multimedia, and SIP favors a more lightweight approach based on HTTP.

Fuente: Microsoft
Formato: PDF

Many applications require fast data transfer over high speed and long distance networks. However, standard TCP fails to fully utilize the network capacity due to the limitation in its conservative Congestion Control (CC) algorithm. This paper proposes a novel Compound TCP (CTCP) approach, which is a synergy of delay-based and loss-based approach. Specifically, it adds a scalable delay-based component into the standard TCP Reno congestion avoidance algorithm (a.k.a., the loss-based component). The sending rate of CTCP is controlled by both components. This new delay-based component can rapidly increase sending rate when network path is under utilized, but gracefully retreat in a busy network when bottleneck queue is built.

Fuente: NetIQ
Formato: PDF

Deploying Voice over IP (VoIP) successfully in a data network has some unexpected pitfalls. In previous papers, we’ve explored how to do a Voice Readiness Assessment and looked at focused planning and design tips. This paper describes changes you can make to improve how a data network handles VoIP traffic – that is, how you can reduce one-way delay, jitter, and data loss for VoIP traffic, while retaining the performance of your other business-critical network applications.

Fuente: Ashton, Metzler & Associates
Formato: PDF

There is an unstoppable movement on the part of network organizations to deploy IP Telephony. However, while the advantages of deploying IP Telephony are compelling, there are some challenging tasks that need to be accomplished in order to have a successful IP Telephony deployment. In addition to ensuring the appropriate level of security, these tasks include assessing the readiness of the network to support voice, designing and optimizing the network to support voice and implementing and supporting the network.

Fuente: Lucent Technologies
Formato: PDF

Voice over Internet Protocol (VoIP) holds the promise of providing customers with exciting new end user features and services while reducing costs. Today, enterprises continue to seek VoIP solutions that will meet these requirements. This white paper examines the long-term enduring values of VoIP that can be leveraged for sustainable cost and revenue advantages as well as a solution for delivering on this value.

Fuente: Adomo
Formato: PDF

Legacy voicemail systems were built with system reliability and enterprise-grade voice response times in mind, and for years they've served the enterprise well. But most legacy phone systems were designed before the introduction of e-mail and other robust business applications, so they can't capitalize on the same advancements that offer today's businesses enormous productivity gains. Download this Adomo white paper to learn more about the evolution of enterprise phone systems and to find out how converged messaging architectures can provide the key advantages of legacy phone systems while lowering your IT costs and boosting end user productivity.

Fuente: Award Solutions
Formato: PDF

The Session Initiation Protocol has been described as a "simple, extensible" IP Telephony signaling protocol. This discussion will explore the second of those two adjectives. We will show that SIP is extensible in three major ways: 1) the general-purpose nature of the message body; 2) the introduction of new headers into SIP messages; and 3) the introduction of entirely new message types. This discussion will demonstrate how these three techniques can be used to devise new features and services.

Fuente: Nuera Communications
Formato: PDF

New application deployments for VoIP networks can use a variety of network protocols and architectures. The use of MGCP and SIP are possible solutions and this paper discusses the design and tradeoffs of each approach. Today’s Advanced Intelligent Network (AIN) provides many useful features common to callers today. These features include Caller ID, Voice Mail, Call Waiting, Pre-and-post paid calling cards, 911, Call Blocking, and Auto Call-back to name a few. These features represent years of development and investment by vendors and service providers, and are delivered via a proven circuit switched infrastructure.

Fuente: Dittberner Associates
Formato: PDF

The packet switching and cell switching networks inherently perform statistical multiplexing meaning they dynamically allocate bandwidth to various links based on their transmission activity. Since bandwidth is not reserved for any specific path, the available bandwidth is allotted according to network needs at any particular time. For integrating voice and data networks, this paper provides an evaluation of the three packet voice transport technologies, VoATM, VoIP and VoDSL.

Fuente: Brooktrout Technology
Formato: PDF

Originally designed for the purpose of conveying information relating to call establishment and teardown from exchange to exchange, the protocol architecture has been extended to cover a variety of tasks associated with collecting and reporting information necessary for the transmission of telephone calls. The SS7 standards now include specifications for a wide diversity of telephony management tasks and have proven to be extremely successful and resilient. As we have moved towards convergence between the public circuit-switched telephone network and the packet-switched IP world, SS7 has become the subject of significant attention as developers seek to integrate the two worlds and leverage the best of both. An understanding of SS7 is thus a vital component of an understanding of the current and next generation of public networks. The purpose of this white paper is to provide a clear explanation of the role of SS7 in the telephone network today, to explore its origins and architecture and to examine its future in this rapidly changing environment.

Fuente: Adomo
Formato: PDF

This Adomo white paper highlights an open approach to IP telephony that has the power to change how your users communicate. Learn why it's next-generation open applications—rather than IP telephones or a PBX—that will actually deliver the business value that converged networks promise. You'll read about new unified communications functionality that allows users to work to work remotely as easily as they work locally, and to work with voice messages as easily as they work with e-mail. As the perfect complement to your VoIP rollout, these new end-user capabilities can make your organization more productive, more responsive, and ultimately more competitive.

Fuente: Adomo
Formato: PDF

This paper highlights an open approach to IP telephony applications that, running alongside CallManager, has the power to change how your business communicates, inside and out. It is these "next-generation" open applications, and not the phones or the PBX, that will deliver the business value that converged networks promise. New functionality under the umbrella of Unified Communications allows users to work with voice messages as easilyas email and to work remotely as easily as locally. Speech interfaces can save time and mask complexity. As a perfect complement to your Cisco VoIP rollout, these new end-user capabilities make your organization more productive, more responsive, and ultimately more competitive.

Fuente: Cisco Systems
Formato: PDF

This paper discusses Cisco AVVID (Architecture for Voice, Video and Integrated Data). Cisco AVVID brings to multiservice networking a standards-based, open-systems architecture for converged networking cisco, network, protection, secure, threats, intrusion, security, attacks, policy, monitoring, infrastructure.