Quality of Service (QOS) (Suscribirme)
Enlaces
Supporting VoIP Traffic in IEEE 802.11 WLAN With Enhanced Medium Access Control (MAC) for Quality of
http://www.recursosvoip.com/docs/english/ALR-2002-025-paper.pdf
Formato: PDF
With fast deployment of wireless local area networks (WLANs), the ability of WLAN to support real time services with stringent quality of service (QoS) requirements has come into fore. This paper evaluates the capability of the enhanced point coordination function (EPCF) and enhanced distributed coordination function (EDCF), which are part of the medium access control (MAC) enhancements for QoS in the IEEE 802.11e supplements, to support voice over IP (VoIP) applications. The performance of VoIP under EPCF and EDCF is shown through simulations, and the impact of background traffic on the VoIP performance is also evaluated.
Synchronization for Improving VoIP Quality
http://www.recursosvoip.com/docs/english/wp-sivoipq.pdf
Formato: PDF
The white paper examines influences that contribute to the Quality of Service (QoS) issues surrounding Voice over Internet Protocol (VoIP) and the deployment of Next Generation Networks (NGNs). The migration from circuit - switching to packet switching technologies has several advantages in both cost and architectural aspects that make VoIP attractive to operators.
The Service Exchange Framework: Providing Greater Control for Cisco IP Next-Generation Networks
http://www.recursosvoip.com/docs/english/cdccont_0900aecd8027e382.pdf
Formato: PDF
To help service providers deliver a rich variety of services to a wide range of devices over multiple access means, Cisco Systems offers the Cisco Service Exchange Framework (SEF), which allows service providers to control customer access and use of services, without limiting the types of applications that can be deployed. The access independent, open SEF helps network operators achieve better understanding, visibility, and control of their network by answering such questions as who their subscribers are and what authorized services and policies govern their use. It helps network operators to assess how the network can be dynamically controlled and indicates where the users and their devices are at any given time.
The Winds of Change: Be Prepared for Business Continuity and Disaster Recovery With Avaya Intelligen
http://www.recursosvoip.com/docs/english/mis3146.pdf
Formato: PDF
Business continuity is a collection of disciplines that are closely related and often confused with each other. Disaster management, disaster recovery, crisis management, business recovery, emergency planning, and business continuity are all frequently spoken of in the same breath like siblings or close cousins. The topic of business continuity has, by default, come to represent this collection of disciplines. This collection is called as Business Continuity and Disaster Recovery (BCDR), but which represents the full spectrum. Business continuity has become a core focus in many companies and countries throughout the world, and there are now laws dictating the needs and parameters for these standards. And certifications to apply best practices.
Understanding Delay in Packet Voice Networks
http://www.recursosvoip.com/docs/english/delay-details.pdf
Formato: PDF
This paper explains the sources of delay when using Cisco router/gateways over packet networks. Though the examples are geared to Frame Relay, the concepts are applicable to Voice over IP (VoIP) and Voice over ATM (VoATM) networks as well.
Understanding VoIP Packet Sizing and Traffic Engineering
http://www.recursosvoip.com/docs/english/cdccont_0900aecd802c52e5.pdf
Formato: PDF
This paper discusses service flows in more detail and outlines the modes that can be applied to a DOCSIS 1.1 QoS parameter set. Next, the paper addresses coder/decoders (codecs) that can be selected and how they affect actual data packet size and traffic rates. One of the foremost issues in traffic engineering is to know where the different bottlenecks are in a network.
Using PSQM to Test Packetized Speech
http://www.recursosvoip.com/docs/english/ZAR_WP_PSQM.pdf
Formato: PDF
PSQM (perceptual speech quality measurement) is a means for objectively assessing the quality of speech that has been degraded by a telephony network. It has a high correlation to subjective quality across a range of types of distortion, and is appropriate for testing networks that are subject to different coding types and transmission errors. Defined by ITU-T recommendation P.861, PSQM is used primarily to test networks that have speech compression, digital speech interpolation, and packetization. Networks that carry voice over IP (VoIP), voice over frame relay (VoFR), and voice over ATM (VoATM) have these characteristics. However, the use of PSQM is not limited to these applications, and can be used effectively to test, for example, wireless systems and cable modem systems that carry speech.
Voice Quality in Converging Telephony and IP Networks
http://www.recursosvoip.com/docs/english/47172.pdf
Formato: PDF
As the telephone industry changes — that is, as new technologies and services are added, existing technologies are applied in different ways, and new players become involved — maintaining the basic quality of a telephone call becomes increasingly complex. Although voice quality has evolved over the years to be consistently high and predictable, it is now an important differentiating factor for new voice-over-packet networks and equipment. Consequently, measuring voice quality in a relatively inexpensive, reliable, and objective way becomes very important.
VoIP in 3G Networks: An End-to-End Quality of Service Analysis
Formato: PDF
This paper presents the results of a Quality of Service (QoS) study for VoIP service over 3G WCDMA networks. An end-to-end simulation platform has been used for this purpose. The simulations have been run using Adaptive Multi-Rate (AMR) speech codec at 12.2 kbit/s with combination of RTP, UDP and IPv6 protocols. The simulated transmission path includes two radio links (uplink and downlink), connected with a packet switched core network and UTRAN Radio Access Networks with several different radio transmission conditions. Furthermore, RObust Header Compression is applied in both radio links. The results include buffering statistics, end-to-end delay estimates, and packet loss statistics.
VoIP Quality of Service (QoS) Testing
http://www.recursosvoip.com/docs/english/voip_5.pdf
Formato: PDF
Network switching equipment vendors have become increasingly concerned with Quality of Service (QoS) issues. This white paper focuses on the issue of testing and verifying network equipment in light of QoS issues. Specifically, it will describe testing algorithms that model mixed VoIP and “Best Effort” traffic under worst-case stress conditions.
VoIP Service Quality Monitoring Using Active and Passive Probes
http://www.recursosvoip.com/docs/english/comsware06.pdf
Formato: PDF
Service providers and enterprises all over the world are rapidly deploying Voice over IP (VoIP) networks because of reduced capital and operational expenditure, and easy creation of new services. Voice traffic has stringement requirements on the quality of service, like strict delay and loss requirements, and 99.999% network availability. However, IP networks have not been designed to easily meet the above requirements. Thus, service providers need service quality management tools that can proactively detect and mitigate service quality degradation of VoIP traffic. This paper presents active and passive probes that enable service providers to detect service impairments.
VoIP Stats for Avaya IP Support Services: Extending Availability, Reliability and Performance
http://www.recursosvoip.com/docs/english/voipstatswhitepaper_071106.pdf
Formato: PDF
The implementation of IP Telephony requires both careful planning and continuous operational care. To start, an enterprise's IP converged network must have the right throughput and QoS capacity to effectively and reliably operate IP Telephony as an application. This network infrastructure readiness can be determined by a comprehensive network assessment. Once successfully implemented, IP Telephony also challenges the enterprise IT shop to performance-manage the IP application and the transport network on which it runs. IP Telephony is critically dependent upon the real-time performance of a converged IP network. Lost voice packets cannot just be re-sent; they result in voice quality deterioration, dropped calls, and potentially, even end-user complaints.


