The following pages feature over 30 VoIP test solution manufacturers and their products. For an industry dealing with consolidation at every turn, it appears that this market is still wide open. Some of these vendors offer point solutions for testing specific elements of a VoIP network; others offer complete end-to-end VoIP lifecycle solutions. In the end, all of these vendors offer technology that is designed to help service providers and enterprises who are deploying VoIP deliver a product that meets their requirements.
These solutions feature new levels of testing capacity and scalability, and they are designed to test carrier grade applications, such as VoIP and IPTV, with increasing forays into IMS testing. These vendors test every relevant protocol, and every type of access, including wireless.
We also list among the testing vendors two testing labs that have carved out a name for themselves in the industry.
So, if youre in the market for a testing solution, we invite you to use this list as a starting point.
Voice Quality Tester
The Voice Quality Tester (VQT) is a comprehensive and objective voice quality test system. It enables the design, deployment, and operation of voice services on next generation networks by providing accurate and objective testing of voice service quality. The VQT provides detailed test and analysis capabilities for voice quality on modern telephony networks such as IP.
The VQT Ethernet VoIP interface is available on the J6800A Network Analyzers 10/100 Mbps Ethernet NIC. The VQT Ethernet VoIP interface supports VoIP call generation using the SIP and H.323, and the G.711 and G.729 codecs. The VQT Ethernet VoIP interface enables testing directly into an IP network for the following applications:
Troubleshoot voice quality impairments by segmenting an IP network for fault isolation great for network delay and clarity analysis;
Assess voice performance of Pre-VoIP Networks; and
Test voice quality for VoIP end users VQT emulates an IP phone.
The Fortissimo NLG-IP is a SIP network load generator that simulates high volumes of SIP subscribers placing calls into the network. The unit is designed to give the user flexibility to serve a wide range of applications associated with switch and network testing. This high volume of bulk call generation can be combined with complex call testing capability. Through the use of user defined call scripts and line protocols, users can tailor test scenarios to meet a wide range of testing requirements.
Each unit supports 192 SIP subscribers being simulated with full bearer path testing and support for registration, authentication, G.711/G.729 encoding and proxy servers via software selectable parameters.
The actions of each simulated subscriber are independently controlled through unique parameter fields defined in user programmed Call Scripts. Call Scripts include capabilities for testing signaling, dialing, Voice over Packet, QoS (GMOS, G-PSQM, G-PESQ R-Factor,) digit decoding, tone sending, path verification, and tone receive. The Scripts define calling patterns and can simulate practically any action a live caller can perform. Scripts can also simulate multiple subscribers allowing testing of multiple-party calls such as conference calling.
emutel Harmony is an efficient, cost-effective, and flexible tool, invaluable for use in the development, performance verification, and pre-deployment testing of many types of VoIP, ISDN, and analog equipment. The product combines comprehensive load testing and network simulation capabilities, detailed protocol analysis, extensive call statistics, and quality of service measurements in an easily affordable and user-friendly package.
An advanced Bulk Call generator, the emutel Harmonys modular architecture provides a powerful, robust platform that can be tailored to meet your current testing requirements and expanded to include additional options as your needs evolve. A single system can provide anything from one to 15 Ethernet interfaces, eight to 120 E1/T1 interfaces, 16 to 240 analog interfaces, or any combination of the above.
The emutel Harmony provides an integrated IP and TDM platform designed to enable testers to completely surround a device under test. This wrap-around performance testing enables end-to-end call routing and ensures seamless communication between IP-based equipment and legacy TDM networks.
Brix Networks recently unveiled its IPTV service assurance portfolio, a comprehensive offering designed to give service providers complete visibility into the quality of video content, the underlying delivery infrastructure, and the overall customer experience.
With BrixVision, service providers can perform root-cause analysis to identify IP transmission versus video quality impairments, monitor end-to-end video quality, and proactively monitor and manage their subscribers experience throughout the lifecycle of their services.
Additionally, the IPTV Infrastructure Verification Package measures the key performance indicators (KPIs) of a network, such as jitter, latency, and frame loss, and correlates these metrics with the overall user experience, which is critical when troubleshooting quality problems. The Infrastructure Verification Package also includes an IPTV dashboard that provides network operators with configuration, scheduling, monitoring, and reporting capabilities to enable proactive means of assuring customer satisfaction.
The DCT2000 and MGTS are multi-user, multi-protocol, programmable test systems designed to significantly reduce product development time and costs while ensuring conformance to telecom industry standards. Their power is achieved through a range of hardware platforms, a line of Catapult-designed network interface cards, and a test library of over 700 protocols.
The DCT2000 and MGTS reduce product development time by allowing multiple independent users to perform their tests concurrently. During the early stages of development, these systems can emulate network equipment that is either prohibitively expensive to purchase or not readily available. Later, the DCT2000 and MTGS can support load testing to measure how much traffic the equipment can handle. These systems can be upgraded and expanded as testing needs change.
DCT and MGTS users are able to create complex, multi-protocol test sequences with a set of graphical tools. These tools are particularly useful for developing and executing tests to simulate heavily loaded conditions and allow the user to change test parameters in real-time. Users are able to significantly reduce the time needed to develop comprehensive 3G, VoIP, SS7, and many other telecom tests.
Clarus Systems, recently unveiled a new version of its VoIP testing software designed specifically for mid-size to large enterprise IP networks. Through automated, end-to-end testing down to the individual phone level, ClarusIPC Operations is designed to accelerate network deployments, reduce operations costs, enable remote trouble shooting, and aim for maximum availability of enterprise Cisco IP telephony environments.
ClarusIPC Operations delivers active, end-to-end testing of the functionality of each phone on the enterprise network, throughout the entire network lifecycle. By automatically and systematically testing and validating all user functionality in a Cisco IP telephony system, ClarusIPC enables organizations to conduct the decades old industry best practice of a nightly health check of the entire telephony environment. ClarusIPC Operations gives organizations visibility into their IP networks, providing vital information and assurance. ClarusIPC Operations features include:
Deployment and operations configuration verification.
Automated, scheduled and customized testing.
Automated, standardized reporting.
Certified integration with Network Management Systems.
Remote troubleshooting of individual and group phones dispersed over broad geographic areas.
Trouble identification and isolation tools for the enterprise help desk.
Empirixs Hammer XMS next-generation monitoring system for VoIP service providers Hammer XMS 1.4 contains a host of features intended to help service providers deliver higher quality service to customers at lower cost.
These features include:
Support for new protocols including H.323, ISDN, TCAP;
Improvements to Hammer XMS unique media analysis capabilities with new RTP statistics and improved diagnostics;
New Error by Type Report; and
Enhanced call correlation to better handle NATing in Session Border Controllers.
Hammer XMS combines patent-pending signaling analysis, high-performance probes, and a highly scalable architecture, enabling service providers to quickly identify and troubleshoot problems and efficiently monitor customer Service Level Agreements (SLAs). The system tracks VoIP and TDM protocol activity as well as media quality in real time for every call, 24x7. Data from multiple remote probes can be centralized and correlated, making reports and real-time diagnostic data accessible through a Web interface. Configuration options range from a cost effective All-in-One configuration to broad carrier-scale deployments.
VoIP Lifecycle Test Suite
Fluke Networks offers a comprehensive approach designed to enable the management of the complete VoIP lifecycle, from pre-deployment qualification of a network through deployment, ongoing monitoring and management, troubleshooting, and planning for future growth. The solution includes:
NetTool VoIP, a portable tool for testing communications between the IP phone and other elements of the VoIP network, and for troubleshooting problems at the networks edge.
OptiView Protocol Expert Plus, a protocol analysis and monitoring solution that can isolate and resolve problems such as network degradation and slow response times.
OptiView Link Analyzer, a hardware analyzer that gives full visibility into network traffic. It reports on network traffic and provides real-time packet capture and analysis, QoS metrics, and alerts.
OptiView Integrated Network Analyzer, a portable analyzer that captures and analyzes voice traffic, using advanced algorithms to determine the voice quality.
ReporterAnalyzer, a NetFlow-based monitoring and analysis solution that provides an enterprise-wide view into which applications are using bandwidth, who is using them, and when.
Wireless VQT Solutions
GL Communications Inc, recently announced a drive testing enhancement to their Wireless Voice Quality Testing (VQT) Solutions. Users of this new product can automatically analyze voice quality of their wireless network as they drive through their geographic region of interest. Features include: a complete portable solution, automated call control for most mobile phone models, support for Bluetooth, and mapping software with voice quality results and mean opinion scores stamped with GPS time and location coordinates.
This tool is conveniently packaged, as it plugs right into the cigarette lighter, and connects to a PC Notebook using USB connections. Calls are automatically and continuously placed and voice quality statistics are gathered and mapped. Other quality measures include round trip delay (RTD) measurements, noise and signal levels, jitter, and clipping occurrences.
Along with automated wireless network testing, GLs VQT Solutions can also support standard PSTN, VoIP, and T1 E1 networks.
PROGNOSIS IP Telephony Manager
PROGNOSIS IP Telephony Manager is a solution designed to provide network readiness assessment, testing, and assurance, performance monitoring of VoIP servers/applications and the network (including gateways, phones, routers, switches), ongoing capacity planning and service level monitoring, be easy to install and monitor through comprehensive dashboard views of the infrastructure and applications, along with immediate alerting.
The PROGNOSIS solution offers the following:
Real-time business views: Immediate access to call detail records such as calls in progress, delay-to-dial-tone rates, and incoming and outgoing calls by gateway bearer channel.
Capacity planning: Differentiate incoming and outgoing calls and understand loading by route pattern, route group, route list, and gateway. As an example, a major U.S. financial institution was able to decommission a third of its gateway capacity, significantly reducing the institutions telephony costs.
Route pattern availability: Easily navigate to gateways (configured for the route pattern) to view the bearer channel status and endpoints. Changes to route patterns, route lists, route groups, and gateways are checked automatically, and alerts generated if down or degraded.
NetHawk EAST VoIP Testing
NetHawk EAST, Environment for Automated System Test, is a VoIP capable testing platform designed to simulate and performance test an extensive list of VoIP technologies in advanced communication networks. The most popular technologies supported include SIP, RTP/RTCP, Megaco (H.248) and MGCP. NetHawk EAST also supports security testing with protocols such as TLS, IPsec, and COPS with DQoS.
NetHawk EAST can be deployed to evaluate SIP presence applications and media quality performance of equipment and networks. PoC (Push to Talk over Cellular) is an example of a presence application that can be deployed in the IP Multimedia Subsystem (IMS). Voice quality methods include PESQ, PAMS and PSQM for live and post monitoring analysis. For video testing, NetHawk EAST supports H.263 and H.264 decoding.
IxVoice is a comprehensive hardware and software test framework that provides unified VoIP and PSTN test solutions for the telecom/network equipment manufacturer, carrier and enterprise markets. With its cost-effective and scalable test libraries it addresses all major VoIP protocols: SIP, SCCP (Skinny), H.323, MGCP, H.248 (MEGACO) as well as TDM and analog telephony services. Functional, load and interoperability issues are easily determined using a unique drag and drop architecture for instant creation of test scenarios with pre-defined visual blocks. IxVoice automates the testing of networks and devices using a multi-interface, multi-technology approach while measuring and analyzing quality of voice and quality of fax.
IxChariot is a test tool for emulating real-world applications to predict device and system performance under realistic load conditions. Comprised of IxChariot Console, Performance Endpoints, and IxProfile, the IxChariot product family offers thorough performance assessment and device testing by emulating hundreds of protocols across thousands of network endpoints. IxChariot provides the ability to confidently assess the expected performance characteristics of any application running on wired and wireless networks.
Service Level Test Automation Solution
From the flexible PowerProbe 6000 capable of housing multiple ISDN, TDM, IP and analog voice interface modules, to the PowerProbe 500 with for analog voice/fax and IP-data/VoIP/video testing, to the PowerProbe 2108 trunk testing platform, Minacom offers a variety of hardware to satisfy your particular testing and interface requirements.
The PowerProbe 6000 ISDN/TDM/IP probe houses three analog test modules, and two IP/VoIP test modules for true converged network testing. A Linux-based, 19 rack-mountable network computer, this carrier-grade probe is highly scaleable, remotely upgradeable, and capable of running the full library of Minacom test agents.
Using Minacoms Portable PowerProbe 6000 with a standalone verison of DirectQuality server on a laptop allows you to evaluate and monitor voice, fax, and modem services over IP. You can accelerate service deployment and troubleshoot problems in the field as they occur. With separate interfaces for fax, modem, analog voice, and IP services, you can simultaneously qualify multiple services replicating the actual customer experience.
The PowerProbe 500 service level test probe is a 1U, 19 rack-mountable Linux-based network computer, housing a two-wire analog interface and two IP-interface modules. The PowerProbe 500 is custom-designed for dedicated or converged VoIP, analog voice, fax, video and data over IP service quality and connectivity testing.
SwitchMonitor is designed to monitor the performance and utilization of all network interfaces (not just a couple of links) so you will know that your entire LAN and WAN are operating at full capacity. This provides complete blanket coverage of your network, giving you a complete picture of your networks health and performance characteristics.
Error statistics are collected for all interfaces and are evaluated for severity and criticality to give you ultimate awareness of your networks weak points so you can strengthen them. This is crucial for VoIP implementations where low incidence of errors and high throughput are required to insure that quality of service does not suffer.
Additional features include the ability to track broadcasts traffic across your network so you can easily know which devices are transmitting the most broadcasts as a percent of traffic. Other capabilities of SwitchMonitor include network inventory information collection and download, equipment uptime tracking, and support contract information tracking and expiration notification.
VoIP Security Solution
NetIQ Corp. recently launched the NetIQ VoIP Security Solution to address organizations increasing needs to assure the security of their VoIP environments. The NetIQ VoIP Security Solution is designed to enable those using Cisco IP Telephony to improve security by reducing exposure time and protecting against loss of confidential data, and is based on NetIQs experience with managing more than 400,000 IP phones.
The NetIQ VoIP Security Solution aggregates and correlates security event information collected from the various elements of the solution. It enables organizations to both monitor the performance and availability of their VoIP environments and to detect VoIP security threats on a real-time basis. The NetIQ VoIP Security Solution also correlates security events and logs them for audit purposes and for performing analysis and forensics.
The NetIQ VoIP Security Solution comprises:
AppManager for Cisco IP Telephony (Security option) Monitors a VoIP environment in realtime to detect security events and configuration changes
AppManager Call Data Analysis Analyzes call detail records to identify abuse patterns and provides complete reports based on the records
Security Manager for IP Telephony Applies correlation rules to security events to identify threats and logs security event information for auditing and forensic purposes.
Sniffer VoIP Intelligence
As more and more companies switch from traditional telephony to VoIP, QoS is of paramount importance in deploying and managing this mission-critical communications technology. VoIP QoS is affected by a number of network design and operations factors, including packet loss and delays, available bandwidth, WAN protocols and the presence of echo.
Sniffer VoIP Intelligence is a business solution that delivers expert network analysis, troubleshooting, and monitoring capabilities to this increasingly important element of the enterprise communications infrastructure. Sniffer VoIP Intelligence leverages and extends the underlying functionality of the Sniffer platform solutions, InfiniStream and Sniffer Distributed, or Sniffer Portable, to ensure QoS on packet voice networks, enhancing the quality of converged networks at every level and optimizing the management of voice, video and data over a single network. With real-time expert analysis and decoding capabilities, you can determine if converged networks are delivering the toll-quality voice services their users demand.
Network Instruments, LLC, recently announced the release of Observer 11. This version includes many robust capabilities for network professionals to optimize network availability, efficiency, and performance. Observer 11 encompasses enterprise-strength Voice over IP analysis, a unique time-based interface for examining up to eight TB of data, and the ability to pinpoint transaction delay through up to 10 conversation hops. Observer 11 is also a multi-topology, distributed analyzer written as a native 64-bit application, while also including a version for 32-bit operating systems.
Observer 11 includes significant VoIP advancements. The VoIP Expert now offers aggregate statistics for overall VoIP traffic, call summary, and quality scoring, as well as over 20 detailed per-call metrics including call status, current jitter, call setup, duration, teardown, MOS/R-factor, and QoS prioritization. As with all Observer features, the VoIP Expert is based on the Network Instruments Distributed Network Analysis (NI-DNA) architecture, meaning VoIP analysis is available across multiple topologies (LAN, WAN, gigabit, 802.11a/b/g), throughout the Observer product line, and for local and remote segments.
The OPERA Voice/Audio Quality Analyzer represents the latest developments to objectively evaluate and assure the quality of compressed voice and wideband audio signals, based on modeling the human ear. With OPERA engineers can achieve a comprehensive analysis of the end-to-end quality of todays and next generation networks, such as VoIP, VoDSL, VoATM, ISDN, GSM, POTS, from the caller to the callee.
For the test engineer, one of the issues is how to apply a computer model of the human ear to a test circuit, or to a live network environment. For this reason, OPTICOM developed OPERA as an open, easy to configure hardware platform that can process perceptual models, which are implemented just by software. The basis of the OPERA system is a portable lunch box type of PC host system. Four slots can take various kinds of interfaces. Alternatively, OPERA is now also available in a rack-mountable form factor. For R&D and lab applications, a cost effective software-only version with reduced functionality is offered, as well.
PSI (Psytechnics Speech IP Monitor) monitors the IP bearer of live customer calls in a VoIP network. It produces a highly accurate quality score based on the ITU Mean Opinion Score (MOS) scale P.800.1, which is representative of customers perceptions of quality. PSI also provides a complete range of diagnostic information to help quickly identify and rectify problems and improve efficiency.
PSI can be integrated into network management equipment, test equipment, VoIP devices, network infrastructure, and handsets. It can be implemented anywhere within the VoIP network to monitor traffic and provide real-time quality scores.
In addition to monitoring and improving overall speech quality levels, PSI provides a meaningful metric for use in Service Level Agreements (SLAs) Analysis of VoIP performance is essential to ensure the consistent level of service required by service providers.
Monitoring and Management Solutions
Qovia is a provider of IP telephony monitoring and management solutions to ensure VoIP call quality. Qovia monitors live calls in real-time and alerts IT operators before call quality is affected, enhancing reliability and end user experience. Qovia can help track VoIP assets on the network, optimize VoIP networks before calls are made and troubleshoot call quality problems, increasing user satisfaction, increased reliability for VoIP network traffic, and lower operational and maintenance costs.
Qovias flagship IP telephony monitoring and management system monitors and manages VoIP call quality. Qovia sensors are placed on the VoIP network to collect call quality information about calls as they occur. Information is sent to the Qovia Service Manager software center for analysis and management across the entire voice network.
Qovia IP telephony monitoring and management solutions support the leading VoIP software and hardware environments:
Qovia for Cisco
Qovia E911 for Nortel
Qovia for NEC
RADCOMs quality management system addresses the many challenges of massive deployment of VoIP technologies and services and the monitoring of voice quality. Service providers, ILECs and cable/MSOs are facing a period of mass adoption and usage of these technologies, with little means of monitoring the services provided.
RADCOMs Omni-Q VoIP testing, VoIP analysis, and VoIP monitoring solution gives service providers, ILECs and cable/MSOs complete visibility into the VoIP service running over the network, enabling early stage fault detection, pre-emptive maintenance and optimization, and drill-down troubleshooting that leads to quick and easy fault resolution.
For the VoIP domain, the Omni-Q offers a set of non-intrusive, live traffic probes coupled with active/intrusive ones, covering next-generation VoIP technologies such as SIP, RTP and H323, as well as legacy voice technologies such as POTS, ISDN and SS7.
Scientific Net IP Communications
VoIP Test Tools
Scientific NetIP Communications, was established to bring to large enterprises, corporations, carriers and service providers the foremost innovative technologies such as VoIP, MPLS, Wireless, IP Applications, Conferencing, and Instant Messaging.
Scientific Devices has for the past 20 years provided test solutions to carriers, service providers, manufacturers, and corporations. Many leading manufacturers have called upon Scientific NetIP Communications to represent them as their manufacturers representative.
With the growing demand in the industry for migrating most applications and services over IP networking, Scientific NetIP Communications has taken a position to bring only those products, services, and technologies that provide customers with a clear return on investment, improve the service assurance, guarantee quality of service, effectively managing remote resources, improving application performance, reduce communications costs, and improve the customers competitive edge.
Shunra Software Ltd., recently announced version 4.0 of its Shunra Virtual Enterprise (Shunra VE) solution. Shunra VE simulates any production network environment in a pre-production setting. This latest version of Shunra VE is focused on delivering detailed service level compliance analysis that enables users to make informed go/no-go application rollout decisions, assess and validate alternative solutions or technologies, and determine which modifications are needed to improve performance and ensure service level compliance before deployment.
New capabilities delivered by Shunra VE 4.0 include the ability to automatically profile and predict performance compliance with service level objectives (SLOs) before deploying new or modified applications or infrastructure into the production environment. Shunra VE 4.0 also includes powerful reporting and analysis, advanced network simulation capabilities, and enhanced end-user automation.
Abacus 5000 ICG3
Spirents Abacus 5000 is a cost-effective, flexible, and scalable IP telephony test system, with integrated analog, TDM, and Ethernet interfaces for comprehensive testing of converged IP Telephony network elements.
The Abacus 5000 ICG3 subsystem simulates VoIP calling functionality. The ICG3 subsystem provides one port with dual media Gigabit Ethernet and 10/100/1000Base-T Ethernet for generating and terminating IP Telephony signaling and media traffic.
When performing call generation, the ICG3 subsystem simulates multiple IP telephones and/or gateways generating the call signaling and delivering the signaling and/or media traffic to a system under test.
The ICG3 subsystem executes endpoint registration requests, which allows measuring the capacity of servers. ICG3 subsystems IP Telephony capabilities, combined with the PCG3s and TCG3s TDM, XCG3s and ECG3s analog features, provide a truly integrated VoIP/TDM/Analog platform to test converged IP Telephony Networks.
VXTracker is based on a completely Web-based Enterprise Java Application server. Once engaged, the VX integrates with a companys PBXs, keyset systems, routers, and VoIP gateways to monitor performance and provide clean understandable reports.
VXTracker is made from Sun Microsystems Enterprise Java and is designed from the ground up using a three-tiered architecture. This design makes VXTracker a stable, scalable, and high-performance system. VX runs inside a J2EE compliant server utilizing EJB transactional integrity to guarantee proper handling of massive amounts of data. VXTracker utilizes an SQL server in order to store and report millions of records without the need for expensive hardware or licenses. VX runs on MySQL or MS SQL Server.
VXTracker connects to a PBX via an RS232 serial port or TCP/IP port. SMDR data is sent from the PBX into the VXTracker where the raw data is transformed into call records, which are then immediately available for reporting. For remote sites you can use the Ethernet buffers, which will immediately publish the data to the host PC. During WAN downtime the device stores the data and forwards to the host application when the network is back online.
Tektronix, Inc., recently unveiled new capabilities for the successful WVR7100 and WVR6100 Rasterizers, including Eye pattern display, jitter measurements, and cable length measurement for High Definition-Serial Digital Interface (HD-SDI) and Standard Definition-Serial Digital Interface (SD-SDI) signals, to meet the needs of broadcast, production and post-production applications. Also included is support for monitoring ancillary data that conforms to the Association of Radio and Broadcast (ARIB) standards used in Japan.
The transition to digital broadcast technologies has created new business challenges and requirements for video and audio monitoring, increasing the need for tools that quickly verify the quality of digital signals. With composite, SD and HD video and/or analog, digital, Dolby Digital and Dolby E audio, the WVR7100 and WVR6100 provide cost effective and easy-to-use monitoring that ensure production and broadcast of quality content that adheres to legal broadcast specifications. The addition of new Eye pattern and jitter measurements on the WVR7100 and WVR6100 provide quick insight to the presence and severity of a problem at the physical layer, improving fault finding and error recovery.
Specifically designed for integration into VoIP endpoints such as media gateways, IP phones, traditional TDM gateways and hybrid IP/TDM systems, VQmon/EP monitors voice calls and produces call quality estimates that can be reported as MOS scores and R factors through the media path using RTCP XR (RFC 3611), end of call signaling or SNMP. VQmon/EP is small (630kbytes) and highly efficient (<500 instructions per second)
VQmon/EP detects packet loss and jitter buffer discard events, extracts key information from DSP software and produces call quality scores and diagnostic data. VQmon/EP has been integrated with products from Audiocodes, Global IP Sound, and Texas Instruments. Other leading DSP software vendors work closely with Telchemy to ensure that integration of VQmon with CODEC and Jitter Buffer software is seamless.
VQmon generates listening and conversational quality MOS scores and R factors and a wide range of diagnostic data, making these available through an API as raw metrics, RTCP XR and SIP QoS Report payloads. VQmon is based on the ITU E Model with many extensions to improve accuracy under time varying network conditions, wideband codecs, orthogonal impairments and signal related parameters.
WinEyeQ version 1.5.0
WinEyeQ version 1.5.0 brings a unique new picture of the relationship of Voice and Video over IP traffic to other data components of the network in true, Triple-Play fashion. WinEyeQ provides support for popular VoIP and data protocols (HTTP, SMTP, POP3, FTP, RTSP, SNMP, 802.1Q VLan), affording a clear, concise, and intuitive portrait of all of the components your network.
WinEyeQ now incorporates Data Scopes as a core component of the representation of your network. Data Scopes are graphical representations of logical groups of components, allowing the user to drill-down on any category revealing ever-finer levels of details. Each Data Scope can be represented as either a Bar or a Pie chart and includes histograms for every metric, allowing you to examine the recent history of any and all activities on the network. These data scopes enhance the user experience by providing a natural interface to network analysis techniques. WinEyeQ provides both monitoring and analysis in a seamless, intuitive fashion. This duality makes WinEyeQ the technology of choice for your complete lifecycle verification requirements.
The modem-equipped 860 DSPi cable analyzer provides all of the capabilities needed for certifying analog and digital installations. A special test sequence verifies VoIP performance and calculates MOS. A unique function displays packets damaged by transient bursts capable of affecting VoIP service. AutoTests make installation certification fast, sure and automatic.
The 860 DSPi performs all of the critical transmission and signal quality tests needed to install and maintain analog, high-speed data and VoIP services. It performs these tests quickly and efficiently with none of the pauses and boot-up delays of other field analyzers. The 860 DSPi puts in a full workday, with a battery life up to three times as long as other field analyzers
The standard 860 DSPi is ready to measure latency, jitter, packet loss, and other VoIP parameters in seconds. Analyze VoIP performance from the subscriber to the CMTA and test the connection from end -to-end. The 860 DSPi displays separate test results for To and From paths and even calculates an MOS score for each path
NetAlly Lifecycle Manager
Viola Networks offers a fully integrated and centrally managed solution for the management of VoIP networks called NetAlly Lifecycle Manager. The solution enables complete management of VoIP across a converged voice and data network in full support of each stage of the VoIP lifecycle readiness assessment, service deployment, operational support and capacity planning. NetAllys extensive and integrated capabilities include:
Active testing that enables pre-deployment assessment and ensures VoIP network-readiness through verification of QoS configurations; identification of maximum call capacity and planning; simulation of calls with exact reproduction through the Real-Time Protocol (RTP) and setting benchmarks for quality and performance.
Passive monitoring that verifies end-user call quality by monitoring the Mean Opinion Score (MOS) and polls SNMP variables to collect health and performance data from network devices.
Testing with patent-pending lightweight active algorithms that reduce the load on the network down to less than 10 percent for a three minute VoIP call.
Easy, centralized management from anywhere through remote control of intelligent, distributed agents with a customizable dashboard, enhanced reporting and capacity planning to optimize provisioning of bandwidth and resources.
WildPackets Omni is a distributed network analysis platform for optimizing network services and maximizing uptime on enterprise networks. Omni gives network engineers real-time visibility into every part of the network including Gigabit, 10/100, 802.11 wireless, VoIP, and WAN links to remote offices. Using Omnis centralized console, distributed engines, and expert analysis, engineers can rapidly troubleshoot faults and fix problems, restoring essential services and maximizing network uptime.
Omni enables network engineers to:
Accelerate troubleshooting and maximize network uptime across the entire enterprise, including remote offices
Gain real-time visibility into any part of the network from a central location
Respond more rapidly and effectively to end user requests
Eliminate travel to other buildings and campuses
Ensure that mission-critical applications get the network bandwidth and availability they need
Increase network performance while decreasing time and expenses for analysis and troubleshooting
Integrate network troubleshooting with other NOC applications and procedures
Adapt global troubleshooting capabilities to new processes and applications as the network evolves
CT Labs is a full service testing and product analysis firm specializing in testing services to converged communications product manufacturers and next-generation network service providers. A primary goal of CT Labs is to deliver top-quality services that fit within the often tight timelines given to us by their clients.
CT Labs is truly a service-based business, whose team pledges to do everything possible to meet and exceed the needs and scheduling deadlines of their clients. CT Labs stays on the cutting edge of convergence technology through its project involvement with the very latest products. Through this work, the CT Labs team is constantly expanding and updating its areas of expertise.
Miercom is a privately held network consultancy, specializing in networking and communications-related product testing and analysis. The firms highly-skilled engineers, many with over 20 years of experience in the networking industry, have developed methodologies for testing products as diverse as SAN switches, SIP phones and IDS systems. The results of VoIP, VPN testing and other technology research articles continue to appear in many prestigious industry journals.
In 1995 Miercom launched the NetWORKS As Advertised program, in which leading technology companies submit their networking-related products for a comprehensive, independent assessment. Only products that meet stringent test requirements, including product usability and performance capability, receive this recognition.
The company further offers on- and off-site diagnostic consultation, equipment recommendation and selection, cabling/fiber planning and design, installation of high-end equipment, and assistance with customized strategic network planning and direction. IT
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