Quality of Service (QOS) (Suscribirme)
Enlaces
eHealth for Voice - Managing for IP Telephony Quality and Performance
http://www.recursosvoip.com/docs/english/voipmgt_brief.pdf
Formato: PDF
Companies today are striving to contain costs and improve productivity in all areas of the business. One opportunity to rein in expenses is through the deployment of IP Telephony built on a Voice over Internet Protocol (VoIP) infrastructure. Concord's eHealth Suite and eHealth for Voice offer rich support for end-to-end management of VoIP, whether it provides end-to-end voice services or whether it co-exists with legacy voice systems, integrated with network and quality of service management.
Enabling VoIP Services With Effective Operations Management Strategy
http://www.recursosvoip.com/docs/english/AccelerateNOS_wpLtr_0504.pdf
Formato: PDF
Driven by competitive pressure and the desire for new revenue generating opportunities, major service providers across North America have begun rolling out VoIP services to business and consumer markets. However, their key challenge remains how to cost-effectively manage the converged network infrastructure to deliver the full range of rich VoIP features and services that their customers are now demanding, while maintaining the consistent high quality of service (QoS) their customers expect. The answer lies in the service provider’s Operations Support System (OSS) infrastructure. This white paper examines the VoIP network challenges that service providers are now facing, the OSS requirements and scenarios for addressing those challenges and present in more detail in the Lucent OSS Solution for VoIP Networks.
End-to-End QoS Support for SIP Sessions in CDMA2000 Networks
http://www.recursosvoip.com/docs/english/qos-sip-bltj04.pdf
Formato: PDF
This paper discusses how end-to-end QoS is supported in CDMA2000 networks using the IP Multimedia Subsystem (IMS) framework. The focus of this paper is on how Session Initiation Protocol (SIP) and Session Description Protocol (SDP) in the application layer are used for QoS negotiation between endpoints. In particular, the paper presents how IMS elements work together to facilitate QoS negotiation on the control plane and traffic enforcement on the bearer plane. The authors compare and contrast two design choices regarding how SDP should be handled during QoS negotiation process. They also report a prototype test-bed implementation of the IMS framework for end-to-end QoS negotiation and traffic control.
End-to-End Voice Quality - The Impact of VoIP Cable Telephony in the Triple Play
http://www.recursosvoip.com/docs/english/0900940380069d32_White_paper.pdf
Formato: PDF
In this paper, Lucent presents QoS option, analyze various impairment factors impacting voice quality in the PacketCable architecture, identify their sources, and discuss guidelines to provide end-to-end QoS for cable telephony. In addition, it discusses how QoS can be monitored on an ongoing basis and the proactive steps that can be taken to prevent congestion.
Factors in the Success of Voice Quality in Converging Telephony and IP Networks
http://www.recursosvoip.com/docs/english/VoiceQuality1.doc
Formato: PDF
Traditional telephony networks are built to provide an optimal service for time-sensitive voice applications requiring low delay and jitter. Telephone networks provide constant but low bandwidth services. However, Internet protocol (IP) networks are built to support non-real-time applications (file transfers or emails) that are characterized by bursty traffic characteristics with occasional high bandwidth demand and longer delays.
Converging telephony and IP networks demand that IP networks be enhanced with mechanisms that ensure the quality of service (QoS) required to carry VoIP. High QoS is especially important considering that traditional telephony network users are used to high voice quality standards. Providing service quality comparable with traditional telephone networks will drive the initial acceptance and success of VoIP services. Consider the following table demonstrating the primary factors that influence user perception of phone service quality.
Feature-Rich Voice Over IP Gives Crisis Communications New Levels of Survivability and Interoperabil
http://www.recursosvoip.com/docs/english/W1814_00c.pdf
Formato: PDF
As leaders from state and local as well as federal government bodies work to improve crisis communications preparedness and response, IP communications infrastructure and applications stand out as solutions that out-perform legacy TDM solutions in crisis situations. Survivability, interoperability and powerful new features address head on the requirements of ensuring continuity of communications and operations in the public sector. This paper analyzes the most promising developments in convergence communications and how they can fill the gaps in crisis communications.
Impact of link failures on VoIP performance
http://www.recursosvoip.com/docs/english/voip-nossdav.pdf
Formato: PDF
ISPs need to provide a comparable quality in terms of both voice quality and availability of the service. One can identify three major causes of potential degradation of performance for telephone services over the Internet, network congestion, link failures, and routing instabilities. The goal of this paper is to study the frequency of these events and to assess their impact on VoIP performance.
Implementing QoS Solutions for H.323 Video Conferencing Over IP
http://www.recursosvoip.com/docs/english/video-qos.pdf
Formato: PDF
Many IP video conference applications use the H.323 suite of protocols. The International Telecommunications Union (ITU) H.323 defines an international standard for multimedia over IP. ITU approved the first version of the H.323 standard in 1996. The current version is 4. Many applications now commonly deploy LAN-based H.323 video systems. An example application is Microsoft NetMeeting, which utilizes H.323 for video conference and shared collaboration. H.323 is the standard with global acceptance for multimedia conferences in an IP network. This white paper discusses tools to implement Quality of Service (QoS) for H.323 video conferences over an enterprise WAN with relatively low-speed links.
Introduction to Differentiated Services (DiffServ) and HP-UX IPQoS
http://www.recursosvoip.com/docs/english/ipqos_wp.pdf
Formato: PDF
The Differentiated Services (DiffServ or DS) model classifies and, if needed, conditions traffic streams to conform to specified levels of service as defined by Service Level Agreements (SLAs). A DS Domain is defined by a contiguous set of nodes provisioned with the same service policies and PHB definitions. The classification and conditioning of packets occurs on DS boundary nodes (or interior nodes - with limited conditioning) within the organization's core network and other DS Domains. The packets are marked and conditioned by the DS nodes to receive a specific level of service as they traverse the network to their final destination. This white paper focuses on the Differentiated Services (DiffServ) model of QoS.
Investigating the Performance of Audio/Video Service Architecture I: Single Broker
http://www.recursosvoip.com/docs/english/SingleBroker-cts05-submitted.pdf
Formato: PDF
The availability of increasing network bandwidth and computing power provides new opportunities for videoconferencing systems over Internet. The number of homes and small offices with broadband Internet connections are increasing rapidly. Even the cell phones will have broadband Internet access in the near future with the deployment of 3G standards. Therefore, it is not inconceivable to imagine that the trend in the increasing usage of videoconferencing systems will continue by accelerating. This will require universally accessible and scalable videoconferencing systems that can deliver thousands of concurrent audio and video streams. However, developing videoconferencing systems over Internet is a challenging task, since audio and video distribution requires high bandwidth and low latency.
Investigating the Performance of Audio/Video Service Architecture II: Broker Network
http://www.recursosvoip.com/docs/english/BrokerNetwork-cts05-final.pdf
Formato: PDF
Increasing network bandwidth and computing power provide new opportunities for videoconferencing systems over Internet. In addition to homes and small offices, even the cell phones will have broadband Internet access in the near future. Therefore, one can imagine that the trend in the increasing usage of videoconferencing systems will continue. This requires universally accessible and scalable videoconferencing systems that can deliver thousands of concurrent audio and video streams. However, developing videoconferencing systems over Internet is a challenging task, since audio and video distribution requires high bandwidth and low latency. This paper proposes service oriented architecture for videoconferencing, GlobalMMCS, and use an event brokering middleware, NaradaBrokering, to deliver real-time audio and video streams to high number of users.
IP Communications in Public Safety
http://www.recursosvoip.com/docs/english/local_IPCSLG_PublicSafety_OV.pdf
Formato: PDF
Public safety agencies serve a vital role in state and local government's top-level goals: maintaining a safe environment for citizens, providing effective services, and advancing economic development by making the community more appealing to new residents and businesses. To meet aggressive goals for response and prevention, public safety agencies and public officials need to achieve the highest degree of awareness of their operation. The challenge is that they lack the ability to share information quickly enough to achieve the best outcome. Today, public safety agencies across the United States are overcoming this challenge without increasing operational costs. With Cisco IP Communications Solutions for Public Safety, agencies connect the entire chain of command, wherever they may be, to actionable information.
IP SLAs - Analyzing VoIP Service Levels Using the RTP-Based VoIP Operation
http://www.recursosvoip.com/docs/english/htrtp.pdf
Formato: PDF
The IP Service Level Agreements (SLAs) Real-Time Transport Protocol (RTP)-based Voice over IP (VoIP) Operation feature provides the capability to set up and schedule a test call and use Voice gateway Digital Signal Processors (DSPs) to gather network performance-related statistics for the call. Available statistical measurements for VoIP networks include jitter, frame loss, Mean Opinion Score for Conversational Quality (MOS-CQ), and Mean Opinion Score for Listening Quality (MOS-LQ).
IP Telephony Solution: Technical Brief
http://www.recursosvoip.com/docs/english/IP_Telephony_Tech_Brief.pdf
Formato: PDF
This white paper, which addresses the opportunity to provide an integrated voice-over-internet-protocol (VoIP)/cellular service, will discuss the growth in mobility users, advances in cellular and Wi-Fi technologies, and the introduction of dual-mode handsets and how this combination of mobility growth and technology advancements presents an opportunity for landline mobility services. It will also discuss how VeriSign is addressing the converged opportunity with the VeriSign Wireless IP Connect Service currently available for customer evaluation and testing.
Latency Key in Wireless-Net Management
http://www.recursosvoip.com/docs/english/EETimesPaper.030929.pdf
Formato: PDF
Many factors affect the perceived quality in two-way communication. An extremely important parameter is latency. This white paper takes a look at why latency is the key in wireless-net management. In traditional telephony, long delays are basically experienced only for long distance calls and calls to mobile phones. This is not necessarily true for voice over Internet Protocol (VoIP), where the effects of excessive delay have often been overlooked, resulting in significant quality degradation even in short distance calls. Wireless VoIP, typically over a wireless local area network, is becoming increasingly popular, but even further elevates the challenges of delay management.


