Metaphor Solutions and XO Interactive Deliver Automated Phone-Based Customer Service Solutions

Metaphor Solutions and XO Interactive (XOI) (news -alert) announced a reseller agreement for XOI to deploy Metaphor’s speech IVR applications on XOI’s hosting infrastructure. This partnership enables XOI to offer all 30 packaged and configurable Metaphor applications and use its life-cycle management tools as part of XOI’s Teleservices offering. Metaphor’s Support Entitlement application has now been deployed at the leading provider of network security and availability software with four more ordered in the current customer agreement. This joint customer now enjoys deflection of calls from human agents to the automated speech IVR system while providing callers with a high quality of service without putting them on hold. “Metaphor provides us with a competitive advantage in offering cost-effective speech IVR solutions that can be quickly implemented for our customers. Packaged applications designed for and targeted at specific vertical industries is clearly what our customers require today,” said Andrea Jadwin, Director of Sales and Business Development at XO’s Interactive Division. “With its highly reliable, flexible and scalable hosting infrastructure, XO Interactive provides a world class platform for our packaged speech applications. And as one of the largest providers of hosted speech IVR solutions, XOI will be contributing to our mission of enabling adoption of speech applications at every enterprise,” said Michael Kuperstein, CEO of Metaphor Solutions, Inc.

8x8, Brightpoint Ink Deal

VoIP and videophone service provider 8x8 Inc. announced an agreement with Wireless Fulfillment Services LLC, a subsidiary of mobile phone distributor Brightpoint Inc. (news -alert) in order to distribute Packet8 Internet phone services and related equipment into its Value Added Reseller (VAR) and System Integrator (SI) channel. Through its online Wireless Enterprise Solutions (WES) program, Brightpoint’s nationwide network of VARs and System Integrators will serve as subagents for Packet8 residential and business service plans. Brightpoint will warehouse, activate, provision, program and fulfill Packet8 devices, eliminating the need for subagents to maintain device inventory while at the same time allowing them to receive commissions for the activation of Packet8 VoIP-based services. Additionally, the WES Web site will provide Brightpoint’s customers with access to its catalog of wireless accessories, mobile applications and content to provide customers with a bundled wireless solution. “By focusing on the wireless and mobility needs of the enterprise market, Brightpoint is creating a valuable opportunity for its dealers to augment their business with targeted solutions like Packet8 Virtual Office,” said 8x8’s vice president of sales & marketing Huw Rees. “We are very pleased to partner with Brightpoint in this effort to introduce small businesses to a phone solution that will dramatically cut their telecommunications costs while enhancing productivity and growth.”

Two Leading Infrastructure VARs To Resell Visual Networks’ New Select Bandwidth Manager for Citrix Presentation Server

Visual Networks, Inc., a leading provider of network and application performance management solutions, today announced that it has entered into reseller agreements with two leading Citrix resellers. Convergence Technology Consulting and IntelliSuite Technologies, Inc. will resell Visual Networks’ new software solution, Select Bandwidth Manager for Citrix Presentation Server, which provides in-depth, cost-effective visibility at local and remote sites into published applications under Citrix’s standard ICA protocol. “Our Citrix clients want to know what information is traversing the network, particularly given that enterprises are increasingly relying on the transfer of data from one location to another,” said Larry Letow, COO of Convergence Technology Consulting. “Select Bandwidth Manager for Citrix Presentation Server is an important tool in our arsenal, helping our clients successfully deliver data from one site to another. Its price point and robust functionality make this the best offering for us and our clients.” Select Bandwidth Manager for Citrix Presentation Server enables IT managers to identify bandwidth utilization on a per-user, per-application basis while also having end-to-end visibility into all Citrix-published applications and non-Citrix applications traversing the network — at both local and remote sites. “We are very excited to have two experienced, highly regarded Citrix resellers in Convergence and IntelliSuite helping us take our new Select Bandwidth Manager for Citrix Presentation Server to market,” said Mark Skurla, executive vice president of worldwide sales at Visual Networks.

Tech Data U.S. to Distribute Latest Communications Solutions from Mitel

Tech Data Corporation announced it has established a distribution agreement with Mitel, a leading provider of IP communications solutions. Tech Data, through its Telephony Specialized Business Unit (SBU), is the only U.S. broadline distributor to offer Mitel hardware, software and accessories. “Mitel is a significant addition to Tech Data’s Telephony SBU product offering, which already includes a broad array of converged voice, video and data communications solutions from industry-leading manufacturers,” said John Ruel, Tech Data’s director, Networking Product Marketing. “The Mitel agreement provides our customers even greater flexibility when developing state-of-the-art communications solutions that offer end users compelling cost savings, improved operational efficiencies and enhanced workforce productivity compared to legacy analog systems.” Tech Data will distribute Mitel’s broad range of IP communications platforms that support from 10 to as many as 65,000 users in a single network configuration; applications and services gateways that provide connectivity to Microsoft’s Live Communications Server; and numerous desktop devices. The Telephony SBU also will offer resellers innovative Mitel solutions for call centers, mobility, speech-enabled unified communications, messaging, video conferencing and wireless communications for deployment in a wide array of vertical markets including hospitality, healthcare, retail and government.

Aspect Software Announces New Version of Aspect EnsemblePro 6.0

Aspect Software, the world’s largest company solely focused on the contact center, announced the general availability of Aspect EnsemblePro 6.0. This latest release from Aspect Software’s Unified Product Line offers substantial new features and functionality, including hosted services capabilities, enterprise quality monitoring, increased security functionality and expanded enterprise administration. The new product capabilities allow companies to more easily achieve their customer service, sales and telemarketing and collections business objectives. Aspect EnsemblePro is a complete contact center solution that unites inbound, outbound, and blended multi-channel contact, while delivering robust queuing, routing, reporting and agent empowerment capabilities. The product also provides, for the first time, application service provider (ASP) capabilities. The enhanced architecture enables service providers to host multiple clients segmented as individual tenants to prevent them from accessing or exhausting each other’s resources while leveraging a common platform. “The newest version of Aspect EnsemblePro not only enters Aspect Software into the hosted services solution arena for the first time, it also provides enterprise users with an even greater level of flexibility and security in solving their diverse contact center challenges,” said Steve Herlocher, vice president of product management at Aspect Software. “Aspect EnsemblePro bridges the gap between inbound, outbound, self-service, and quality monitoring functionality, enabling organizations to easily provide blended contact center services. In addition to offering three times the scalability of previous versions, it offers the most comprehensive features and capabilities available today from any unified product.”

KANA Response Selected for Integration into Gem Contact Center Services

KANA Software has been selected to partner with Gem in its contact center services. Gem, a Northern-Ireland based provider of multi-channel, multi-lingual contact center services to global organizations, is implementing KANA Response from KANA Software, a provider of Service Resolution Management (SRM) solutions. KANA Response is an email response management application that will assist gem in handling its ever increasing volumes of email, up to 500,000 per day from 22 countries. Since the implementation of KANA’s technology, response capabilities have increased by 40 percent. Gavin McGoldrick, Technology Director at gem noted that the company bombarded KANA Response with emails during the testing phase and it quickly became apparent that the system could easily manage any seasonal volume spikes, such as the Christmas period. With the level of performance and extensive response automation, McGoldrick felt that KANA Response delivered the wow factor that gem was looking for. Through KANA Response’s extensive automation functionality, agents are free to handle more complicated calls into the contact center instead of referring these calls directly to clients. According to Marchai Bruchey, SVP of Marketing and Alliances at KANA, the KANA Response enables gem to tailor its operations to meet the best practices of each client it supports.

TI helps Net Telephony Providers Manage Devices

Texas Instruments has developed software that aims to help Internet telephony operators better manage devices sitting in their customers’ homes and offices. The company unveiled Piqua, a software that sits in voice over Internet Protocol (VoIP) gateways, such as those sold by Linksys or D-Link, and VoIP phones. The software can detect and automatically make adjustments to ensure better calling quality. For example, if there is an echo on the call, the software can detect it and automatically fix the problem. Problems can now be fixed even before customers notice there is an issue, TI said. The Piqua software also can help technicians in the VoIP provider’s call center diagnose problems more efficiently and effectively by giving them more visibility into the device, the company said. Troubleshooting problems with a technician on the phone can turn into marathon ordeals where the problem is still unresolved at the end of the day. And all too often, customers blame the VoIP devices for their troubles, which they often send back to the service provider. But most times, the device is fine — it just wasn’t configured properly, said Lindsay Schroth, a senior analyst at The Yankee Group. ”In an IP service, like VoIP the intelligence is in the phone and not at the central office, like it is in the traditional phone network,” said William Simmelink, general manager of the packet voice and video business unit for TI. “So it’s important for technicians in a call center to be able to control the devices sitting at the customer site.”

Mercom Releases Version 2.0 of Mercom Interaction Quality

Mercom Systems, Inc. announced the release of version 2.0 of Mercom Interaction Quality (MIQ). Mercom’s MIQ software is a powerful browser-based call evaluation and quality monitoring system, which advances the correlation of quality and productivity metrics in the contact center. New features in MIQ 2.0 include: • Quality Key Performance Indicator Report — shows correlations between quality scores for behaviors and an outside metric the user would like to impact (e.g., revenue, customer complaints, etc.). This directly translates into improved coaching and training as well as enhanced return on investment. • Performance Dashboard — allows users to get a quick graphical overview of quality score trends and key quality indicators in their center. Organizations can customize their performance dashboard so they can easily view information pertinent to them. • Error Analysis Reporting — allows users to view and analyze specific areas in which groups or individuals are making errors during calls • Enhanced Calibration Function and Reporting — makes comparing multiple evaluators’ scoring to a mean or standard faster and easier. Standard deviations can be tracked against goals in easy-to-read graphical views and drill-down functionality enables root cause discovery when deviation is present. “The enhancements we’ve made to MIQ version 2.0 demonstrate our continued commitment to providing the best tools for our customers,” said Avi Margolin, president and CEO of Mercom. “MIQ has always been a strong, popular product for us, and by adding these new intuitive features for performance management, we feel it will bring more value to the marketplace and that our customers will be very pleased with the results.”

Verizon Completes $8.5 Billion Merger with MCI

Verizon Communications and MCI (news -alert) announced they have finally closed their $8.5 billion merger, creating a regional Bell operating company with a national footprint and approximately 250,000 employees serving customers in 150 countries. As part of the integration of MCI, Verizon said it has created a business unit encompassing business and government customers and related functions of the former MCI. Michael Capellas, former president and CEO of MCI, also announced that he is leaving the business, now that the merger has been completed. “This milestone for Verizon creates a new competitive force with the power of the global MCI network and the reach of Verizon’s broadband and wireless networks in the U.S.,” said Verizon Chairman and CEO Ivan Seidenberg. “The combination of our world-class wireless and broadband access networks with the leading global IP (Internet protocol) backbone will allow us to deliver the highest quality end-to-end experience for our customers.” As a result of the merger, Verizon will now operate three network-based businesses: the newly created Verizon Business, Verizon Wireless, and Verizon’s landline segment, which is deploying wireline broadband and video networks. Verizon Business will also include part of Domestic Telecom, including the former Verizon Enterprise Solutions Group. Verizon Business will target medium and large businesses and government customers. New products and services are expected later this month.

Linksys Announces SIP-Based IP PBX, Desktop Phones, and Gateway for Internet Telephony Service Providers

Linksys, a Division of Cisco Systems, Inc., announced a new line of SIP-based telephony products for Internet Telephony Service Providers (ITSPs) targeting large residential, SOHO, and very small business customers. The new line of IP communication solutions includes an IP PBX/Key system, a wide range of IP desktop phones, and an analog gateway for connection to the PSTN. Used together with an ITSP voice service, they provide a complete IP telephony system for up to 16 users. “With the new Linksys SIP-based IP communication offerings, ITSPs can offer residential and small businesses a voice service with many of the features found in large business voice IP networks, such as multi-line service, music on hold, auto attendant, and more at a more affordable price,” said Jan Fandrianto, vice president of voice engineering at Linksys. “The new IP PBX and IP phones bundled with a service provider offering will make the deployment of voice networks easy to install and simple to use at a price small businesses can afford.” This new solution will complement the recently announced Linksys One solution. Linksys One is an ideal solution for small business with 5 100 users needing a complete communications solution that addresses voice, video, and applications. The LVS series was developed to address the residential and very small business of 1-4 users that may grow to 16.

Pactolus Intros InService High Availability Software to Prevent IP Service Session Loss

Pactolus Communications Software Corporation introduced its InService multi-site high availability architecture, an innovative framework to provide its SIPware carrier service applications with unprecedented resiliency and the fastest achievable recovery speeds. The first InService integration is for Pactolus’ widely-deployed SIPware Audio Conferencing, making it the industry’s first IP service application to help network operators and service providers ensure uninterrupted subscriber service delivery during both occasional IP network component failures and extraordinary disruption events,. “Momentary IP service outages can be particularly apparent in hosted multi-user service sessions. Even a fraction of a second’s interruption can force all audio conference participants to terminate and restart a call session, which is why many established carriers have balked at offering IP-based conferencing service, despite their obvious economic, flexibility, and innovation advantages,” notes analyst Deb Mielke of Treilage. “Pactolus is the first IP services vendor that has addressed this need, which opens important doors for IP service delivery for both new and established service providers.” “The ability to encapsulate a SIP-based services architectures with ‘Tier 1 Telco-Grade’ reliability and call state protection hasn’t been available, which has slowed migration to IP services for some Tier 1 service providers,” noted Chris Brimhall of XO Communications. “Pactolus has been uniquely innovative in identifying and responding to this issue, first with CallComplete for service sustainability, and now with the session resiliency that only a multi-site availability architecture can offer. The ability to support calls across multiple POPs simultaneously — and the intelligence to automate call state preservation during a disruption — are definitely capabilities that help redefine and ‘market-harden’ IP services.”

Concurrent Rolls Out MediaHawk 4500 Server for VOD Applications

A new high-density server ideal for video-on-demand (VOD) applications is now available from Concurrent. (news - alert) The company has announced its MediaHawk 4500 server, enabling video service providers to store more than 20,000 hours of on-demand content and enabling VOD ingest rates of more than 2,500 hours per day. The new server offers improved stream density of up to 2,400 streams per 2RU enclosure, enabling small and large deployments with minimal power consumption and footprint. The MediaHawk offers a scalable RAM cache approach and a commercial hardware design, enabling service providers to employ a scalable, pay-as-you-grow deployment model. “Since introducing video-on-demand to the US market in 1998, Concurrent’s expanding portfolio of MediaHawk products have successfully met the growing needs and ambitions of video service providers the world over. The MediaHawk 4500 takes VOD server capabilities to the next level, delivering significantly improved performance with the utmost in platform reliability, while meeting the stringent economic demands of our customers,” said Gary Trimm, CEO of Concurrent. “Concurrent was the first VOD server company to offer independent scalability of ingest, streaming and storage,” said Bob Chism, Concurrent’s CTO. “The MediaHawk 4500 offers a fourth dimension of flexibility, enabling RAM cache to be scaled as needed. The result is an ideal combination of high performance and economic attractiveness.” Chism added, “The MediaHawk 4500 platform is fully compatible with existing MediaHawk deployments, allowing Concurrent’s existing customers to take advantage of this breakthrough technology without the need for forklift upgrades.”

Vertical Introduces InstantOffice 7.0

Vertical Communications announced Vertical InstantOffice 7.0, the new release of its award-winning integrated communications platform for retailers and other large, distributed enterprises. InstantOffice 7.0 takes advantage of the Session Initiation Protocol (SIP) standard, which allows customers to create a true channel to manage voice across their entire networks. “Vertical InstantOffice was designed as a unified platform, to make it simple for our customers to streamline the need for multiple devices,” said Bill Tauscher, Vertical’s Chairman and CEO. “In InstantOffice 7.0, we have made it even easier for our customers to use the latest voice, data and application services technologies. In leveraging SIP, we are our customers with a critical advantage for better serving their customers and thus creating the opportunity for more loyal and profitable customer relationships.” Designed for retailers with hundreds or thousands of stores, InstantOffice 7.0 is a integrated communications platform that consolidates voice, data networking, and voice applications, thus allowing customers to track, manage and optimize voice communications in a flexible, timely way to support their business goals. Because these organizations are continually looking for more costeffective ways to provide better customer service, InstantOffice 7.0 now leverages SIP to provide the greater productivity benefits and lower costs delivered through Voice over IP (VoIP). Importantly however, InstantOffice 7.0 does not force customers to commit to a pure IP environment. The open, standards-based platform offers investment protection for existing resources and allows customers to leverage new technologies as they emerge and as their business needs dictate. As part of the company’s open IP approach, InstantOffice 7.0 allows customers to choose the best mix of phones for their businesses, including digital, analog, wireless and IP phones.

IMS/SIP-Based Services Delivery Made Easier Through Pactolus and Convedia

Pactolus Communications Software Corporation and Convedia Corporation (news - alert) are teaming up to accelerate and simplify deployment of media-rich services and features for applications based on the Pactolus RapidFLEX Service Creation Environment and its SIPware Carrier Services Suite. Convedia’s IP multimedia processing capabilities will be leveraged through the partnership. The partnership initiative will both simplify media-rich feature deployment, as well as delivery of services across fixed and mobile networks. The IMS cross-network service architecture promotes adoption of advanced calling features, unified calling databases and customizations. These enhancements promote subscriber loyalty and extend the functionality of carriers’ services. The integration of the two network services will enhance carrier support for integration of rich media applications like conferencing and collaboration, as well as entertainment applications. “Convedia and Pactolus consistently rank among the most respected, innovative and broadlydeployed names in VoIP-enabled telecommunications,” said Dave Horton, president, CTO and founder of Pactolus. “Our joint IMS development further enriches our ongoing relationship, and simplifies service provider IMS migration and expansion.” ”Pactolus and Convedia were among the first companies to whole heartedly embrace SIP and the access agnostic, vertically layered approach to enhanced services that 3GPP’s IP Multimedia System so well articulates,” said Grant Henderson, co-founder and executive vice president of Convedia. “By embracing best-in-class products from companies like Pactolus and Convedia, fixed and mobile service providers are better able to realize the true promise of IMS.”

Intel Bows Developer Tools Designed for Dual-Core Apples

One day after legendary Steven Jobs convinced Intel’s President and CEO Paul Otellini to don its hallmark white suit and unveil the new Apple Computer (news - alert) models based on x86 processors, Intel bowed new software development tools and resources specifically for dual-core processors. As part of the Intel Core Duo processor tools and resources, the company launched free, beta versions of the Intel Fortran Compiler, Intel C++ Compiler, Intel Math Kernel Library and Intel Integrated Performance Primitives for developers to try out as well as other resources to assist with software optimization, dual-core threading and migration information. And officials with its software products division hint that is only the tip of the iceberg. “We look forward to developer feedback prior to introducing our products in the coming months,” said William Savage, general manager of Intel’s Software Products Division. “Our highly optimized compilers and libraries yield significant performance advantages for applications and take advantage of the opportunities made available through multi-core and multi-threaded environments.” These development tools are integrated into Apple’s Xcode development environment and offer an alternative to existing tools and compilers. The Intel Fortran compiler enables the scientific and technical community to bring the fastest versions of their applications to Mac OS X and Intel-based Mac platforms using industry-standard math libraries and language. The Intel C++ Compiler provides the ability for Xcode users to apply targeted optimizations to performance-sensitive areas of their applications, allowing them to get the most out of the Intel Core Duo processor.

Phihong’s Redundant Power Source Provides up to 180 PoE Ports for VoIP, IT Applications

Phihong USA,a global leader in Power-over-Ethernet solutions, has developed a redundant power source with up to 180 PoE ports to meet the needs of telecom applications, such as VoIP systems. The RPS accepts three 500-watt power supplies for powering VoIP phones, DC UPS, and lighting systems with single UPS. A typical VoIP phone draws less than 8 watts. With 1500 watts available, 180 PoE phones can be supported from this redundant source. “Redundant power sources are essential to VoIP systems, particularly for businesses where downtime due to power failure can significantly compromise operations,” said Keith Hopwood, vice president of marketing for Phihong USA. “If VoIP power is provided by PoE devices, an RPS is necessary to keep power generating during an outage.” Each output in the redundant power source is individually protected against overloads causing overheating of wiring from excessive current, thus fully protecting the system against overload, over-temperature and over-voltage. It features diagnostic capabilities and LEDs next to each connector. The RPS was designed to allow easy connection of a battery plant for full DC UPS capability for VoIP phone systems or network switches. A single battery plant can provide a UPS function for over 180 IP phones or a complete rack of network switches.

Picolight Addresses Needs of Market With New 1310nm VCSEL

Picolight, Inc., a designer and manufacturer of optical transceivers and components, is the first to ship 1310 nanometer (nm) vertical cavity surface-emitting laser (VCSEL) transceivers in a 4 gigabit per second (Gbps) triple-rate Small Form Factor Pluggable configuration to target one of the fastest growing segments of the data center market. The new transceivers offer extended reach and lower power consumption for short-to-medium distance applications. VCSEL technology emits light vertically through the surface of water as opposed to Fabry-Perot and differential feedback lasers that emit light through the edges. VCSEL devices require very little electrical current to produce optical energy output of 850nm and above and are easy to use to transmit light into an optical fiber as they emit narrow, circular beams. The 1310nm VCSELs from Picolight deliver lower power consumption, lower electromagnetic interference (EMI) and lower heat generation to produce increased performance and reliability over single-mode fiber according to Vidya Sharma, vice president of marketing for Picolight. Sharma’s position is that next-generation high-density form factors will strongly favor the exclusive use of VCSEL technology due to its low heat and low EMI-generation characteristics compared to that of edge-emitting lasers. Highlights of the 1310nm VCSEL include the use of 850nm transceiver architecture; single transceiver architecture addressing multiple specifications at 4km and 10km, reaching up to 40km; an un-cooled 1310nm oxide-confined, high-speed VCSEL coupled to an LC optical connector; extended temperature and voltage range options; internal AC coupling on both transmit and receive data signals; all-metal housing for increased EMI shielding.

Arista Launches its New Compact, Low-power Industrial 6.4-inch LCD Panel Computer

Arista Corporation, a leading industrial PC solutions provider, introduced its new industrial 6.4-inch LCD Panel Computer, the ARP-2606AP Designed to meet the demands for mission critical, high-performance, and reliable operation, the unit is ideal for factory automation, facility monitoring, machine automation, and environmental monitoring. The new ARP-2606AP is equipped with 6.4” LCD and 3.5” embedded board, which incorporates a VIA Eden 667MHz processor. System memory for the unit comprises one 144-pin SO-DIMM socket with up to 512MB of memory capacity. The Local Area Network (LAN) uses a Realtek 8139 C PCI PnP Base-T Ethernet controller. Video support features a built-in VGA controller with up to 32MB of shared memory for display. The system comes with three RS-232 ports, a four-Wire resistive touch screen and an optional PCMCIA for wireless applications. In addition, the ARP-2606AP has an on-board CompactFlash Type-1 socket, an optional 2.5-inch hard disk drive (HDD) and one 16-bit PC/104 extension connector. “The new compact ARP-2606AP is ideal for compact environments,” said David Marich, sales manager for Arista Corporation. “The unit’s rugged design and its panel-mount capability ensure easy installation for a variety of industrial applications. All of Arista’s panel computers are backed by our optional next day replacement warranty, further ensuring our commitment to quality and excellence.”

Octasic Expands its Voice Processing Product Line with New Devices

Octasic Semiconductor is enhancing its OCT6100 family with the introduction of lower power, lower cost devices, enabling higher density and the highest voice quality for customers designing down to four channels. The OCT6100L builds upon the proven success of the current OCT6100 devices, offering unprecedented Voice Quality Enhancement features through the use of 130nm technology. As echo cancellation moves from a standalone box to an internal embedded feature of gateway and switching equipment, designers are faced with increasing demands for lower power to meet their density targets, while improving quality. Octasic has responded to this demand with the OCT6100L. The OCT6100L proves Octasic’s commitment to compress product development time for their customers and provide cost efficient, power efficient, performance driven silicon, while ensuring the longevity of the product for the enterprise and carrier equipment market space. “With over 15 million ports deployed, the OCT6100 has been a great success. We are pleased with the acceptance of this architecture. Now, with the OCT6100L, Octasic can respond to the most stringent customer requirements for density and cost,” explains Doug Morrissey, Octasic CTO. By retaining all the features and functionality of the existing OCT6100 devices the OCT6100L leverages the extensive field experience of the existing devices which have been qualified into several leading carrier networks.

IPTV and HDTV Video Content Creation Sped Up with Tarari Encoder Accelerator

Creating Windows Media and SMPTE VC-1 video content for consumer electronics devices is now quicker and easier, thanks to a new hardware video encoder accelerator from Tarari and its partners. The Tarari Encoder Accelerator for Windows Media speeds up the creation of video content for devices that play HD video over the Internet, through VOD systems, via IPTV systems, on set-top boxes, on HDTVs and on next-generation DVDs. The new product features expanded support for SMPTE VC-1 encoding, in addition to Microsoft WMV-9. That Tarari accelerator reduces the time to encode both standard-definition and high-definition video content. It is integrated with Microsoft’s encoding software stack, available via Windows Media Format SDK and the GUI encoder application. The Encoder Accelerator directly offloads functions within the Microsoft encoder, offering up to a 10x boost in encoding speeds. The Microsoft codec can automatically detect if Tarari hardware is present and subsequently offloads complex encoding tasks, leaving existing workflows unchanged. The plug-and-play Tarari board may be installed in a server or workstation to enable the additional encoding performance. “Tarari’s new Encoder Accelerator makes a significant contribution toward enabling quick output of high-quality video content for network distribution through the new media channels now emerging in business, education, and entertainment,” said Randy Smerik, Tarari CEO. “Tarari’s partnership with Microsoft has produced an acceleration processor that is capable of seamlessly working with practically any application that can encode Windows Media and SMPTE VC-1 video.”

Phihong’s One-Port Midspan Supports High-Speed Wireless Access Points

Phihong USA has developed a one-port midspan for highpower devices that are capable of powering 802.11N access points. Typical applications for the 30W midspan include wireless/WiMax network access points, security cameras and IP telephones with streaming video. “Along with the new standard, customers are looking for equipment that is interoperable across multiple vendor platforms,” said Keith Hopwood, vice president of marketing for Phihong USA. “Our 30W one-port midspan provides Power-over-Ethernet technology that does just that.” Phihong is a member of the new IEEE 802.3at task force, which was formed to increase the power levels distributed via Ethernet to at least 45W. When implemented as a standard, the PoEPlus initiative will more than double the wattage available to powered devices. The 30W midspan features diagnostic LEDs and is fully compliant with the IEEE 802.3af standard in detection, disconnect and voltage control. The device is also Gigabit-compatible.

Enabling Automatic Parallelization of Application Programs

Cisco Systems announced it has developed a set of new products that will make it easier and less expensive for municipalities to deploy free WiFi Internet access over large, outdoor areas. The company claims that programs, without modifying them. by harnessing the power of a new technology known as “wireless mesh networking,” it can set up cities and towns with free public WiFi — something that many city officials now view as essential in order to boost economic development. The new technology was born out of Cisco’s acquisition of tion is almost always completely accurate. Airespace, Inc., a provider of centralized wireless local area network equipment (including mesh architecture), earlier this year. By combining Airespace’s technology with Cisco wireless products, the company developed its “wireless mesh” offering, which takes conventional WiFi “hot spots” and links them together, much the same way routers link the access point. In essence, mesh technology extends WiFi service from “spot” coverage to “blanket” coverage, making it possible for users to be continuously connected while traveling through a large area. Mesh access points can be installed on street practical period of time. lights, power poles, or other public infrastructure in a matter of months One advantage of mesh technology is that it makes centralized management much simpler than with previous WiFi technology on a field-programmable gate array (FPGA). technology. For example, if one access point fails, the network can continue to operate just as it did before, only lost of speed is actually accelerated.

Intel Launches Viiv Entertainment PC

Chip-maker Intel Corp. launched its Viiv entertainment PC platform and announced a slew of deals to provide content for the new systems. Viiv PC owners will be able to watch video that’s stored in Google Inc.’s video service, high-definition highlights from NBC’s coverage of the Winter Olympics and classic TV shows from America Online. In all, Intel has signed dozens of content deals. Viiv computers will be capable of replacing the array of standalone boxes that surround the television — such as a digital video recorder, DVD player or cable box. Intel also says it’s working to ensure a Viiv PC works seamlessly with other equipment. By focusing on more than chips, Intel hopes its technical and marketing muscle will help make entertainment PCs easier to use — and more appealing. With our new platforms, we’re not only boosting wireless computing, but also advancing digital entertainment a few steps closer to effortless,” Intel CEO Paul Otellini said. Some of the options include support for a technology that allows Viiv users to quickly turn their systems on and off after the initial boot. Machines also will ship with at least 5.1 surround sound and support for high-definition video. TV capabilities are optional.

Brix Offers Improved IPTV Service Assurance and Monitoring

Brix Networks has expanded its IPTV service assurance and monitoring capabilities to offer the required visibility into channel change times and network performance related to multicast infrastructure. The company’s solutions can now assure quality of bundled services, while providing network operators with service lifecycle support. The solutions provide extensive visibility into delivery infrastructure from super head ends to central offices to subscribers, helping to ensure the success of IPTV services. The Brix System of integrated hardware and software products proactively monitors VoIP and IP video quality. The solutions also enable providers to measure user response times and monitor network performance and service degradation related to running dynamic IP video services over IP. Providers may also test simulated channels by emulating settop boxes for zap and VOD function delays using the solutions. “Service quality matters, and consumers have a higher expectation of quality when it comes to IPTV,” said Robert Travis, director of product marketing at Brix Networks. “Working closely with our customers, we know that measuring channel change times and monitoring IP infrastructure performance is imperative to the success of rolling out quality IPTV services. These factors, combined with the growing demand for bundled services, have made service assurance a requirement to the successful deployment of revenue-generating services.”

TRENDnet’s New Mobile Solution Combines Bluetooth and 802.11g

TRENDnet, a best-in-class networking manufacturer of wired and wireless networking solutions announced its latest mobile innovation, the TBW-103UB 802.11g WiFi & Bluetooth Combo USB Adapter. The device integrates WiFi and Bluetooth access into one USB 2.0 adapter to enable the seamless transfer of information between Bluetooth enabled devices, such as cell phones, and computers. By integrating the two wireless access tools into one unit, a bridge between the Bluetooth-equipped cell phone and the PC has been created. While Bluetooth has become a common feature in newer cell phones, it is not always included as a standard feature in PC’s. By connecting the TBW-103UB to a regular laptop USB port, mobile users can now sync contact files, share photos, listen to music or make VoIP calls via their Bluetooth-equipped cell phone or headset. Simultaneously, users can surf the Internet via their WiFi connection. “This unit is the latest product in a line of solutions recently launched intended to improve the productivity and lifestyle of laptop users,” stated Heath Gregory, Marketing Director for TRENDnet. “By combining these two standards into one solution we are helping users make the most of their mobile devices and enhance their ability to distribute their content and stay connected anywhere.”

Internet TV and Satellite Radio Come Together With Wi-Fi TV InterSat

Wi-Fi TV Inc., which streamed the first full-length movie on the Internet in 1995 and, in 2005, became a pioneer in Internet TV, announced it is offering global TV stations, radio stations, and content producers a new distribution platform. The Wi-Fi TV distribution platform is called InterSat because it utilizes both satellite and Internet delivery technologies in tandem to offer a global delivery platform compatible and accessible to the widest possible number of TV stations, radio stations and content producers. Wi-Fi TV InterSat, like satellite radio, offers stations the opportunity to reach a large geographic audience and to do so without the licensing associated with traditional TV and radio. Wi-Fi TV InterSat delivery also offers a number of interactive features not possible with traditional TV and radio. The Wi-Fi TV InterSat system was successfully implemented by Wi-Fi TV in a Beta format for two live events: New Year’s Eve from Times Square and the Tarver-Jones 3 pre-fight weigh in. In each case, live coverage from cameras on the scene were uploaded to a satellite, retrieved from the satellite at the location of Wi-Fi TV Internet servers, encoded on the fly, and streamed by Wi-Fi TV both live and archived to a global audience. In the case of the New Year’s Eve Webcast, a live Wi-Fi TV chat feature included comments from viewers.

Broadcom Delivers Revolutionary Solution for Video IP Phones

Broadcom Corporation, a global leader in wired and wireless broadband communications semiconductors, today announced the world’s first WiFi video phone chipset for wireless handsets and desktop video IP phones. The chipset and its associated software leverage several market-leading Broadcom(R) technologies including VoIP, mobile multimedia and wireless LAN (WLAN). The revolutionary video IP phone chipset further expands Broadcom’s leadership position in the IP phone chip market. Video phones that are based on Broadcom’s new WiFi video phone chipset will change voice communications into a completely new form that is practical and fun. Imagine business travelers saying good night to their children “face-to-face” by simply dialing home, or teenagers being able to compare outfits before they meet up for an evening out together, or families being able to celebrate special occasions together despite the distances that separate them. “The age of video telephony is finally here. Our ability to integrate several of Broadcom’s leading technologies will enable our customers to bring to market cost-effective, low-power video phones with world-class video quality,” said Paul Shore, Director of Marketing of Broadcom’s VoIP phone products. “Video phones on the market today lack the price point and video quality needed for mass market adoption — our new chipset and software will radically improve both of these key market drivers, establishing Broadcom as the market leader in video IP phones.”

U.K. Cities To Get Blanket WiFi Coverage

The United Kingdom has unveiled plans for citywide WiFi networks that will give residents in nine cities high-speed wireless Internet access from laptops, PDAs, and mobile phones. The first phase of the project, due to be completed by March, will see citywide WiFi hot spots rolled out in Birmingham, Cambridge, Edinburgh, Leeds, Liverpool, Manchester, Nottingham and Oxford, along with the London boroughs of Kensington and Chelsea, Camden and Islington. The networks are being built by European wireless provider The Cloud. They will be open to any Internet service provider that wants to offer services. Blanket wireless coverage will be provided in the cities through WiFi equipment fitted on lampposts and street signs. People who want to use the wireless network will pay one of the ISPs for access, and revenues will be split between The Cloud, the local council and the ISPs. WiFi coverage for more cities is expected to be announced later this year. George Polk, CEO of The Cloud, said the aim is to provide wireless coverage across all U.K. cities and major centers of population. “Providing ubiquitous wireless broadband access over a network that is available to millions of WiFi devices...will have a major impact on the way people communicate, work and play in city centers,” Polk said in a statement.

Tele Atlas and Skyhook Wireless Announce Agreement

Tele Atlas, a leading global geographic content provider, and Skyhook Wireless, provider of the industry’s first WiFi Positioning System (WPS), announced today an agreement aimed at delivering next generation location-aware solutions. The companies will work together to provide application developers with a single source for location-aware geographic content. The agreement will not only expand the addressable market for location-aware applications to any WiFi enabled laptop, PDA or mobile phone, but will also enhance the location coverage available in environments where traditional location technologies fail. “The ability to easily pinpoint a specific location on a map — whether indoors or out — is imperative for the fast developing location-based services (LBS) applications market,” said Michael Shean, co-founder and vice president of business development, Skyhook Wireless. “Tele Atlas has a superior process for ensuring its maps and other geographic content are highly accurate and completely up-to-date, as well as a long heritage of serving the very complex emergency services market. They are also the ideal partner, with the best database, for the emerging location application market.” “Skyhook’s WPS technology breaks traditional barriers in metropolitan markets, and will deliver a significant advancement to our partners,” said Mike Gerling, chief operating officer, Tele Atlas, the Americas. “The combination of Skyhook’s WPS solutions and Tele Atlas street navigation and geocoding data means mobile consumers of all stripes and emergency services personnel can, without interruption, locate any place, product, or person.”

Navizon Transforms Cell Phones and WiFi Devices into A GPS Device

Mexens Technology announced Navizon, the world’s first software-based, peerproduced, wireless positioning network. Navizon is a software system that can be loaded onto a mobile device to provide its users with geographical positioning information plus many more advanced customized features. Navizon’s innovation arises from the fact that its map is both created, and constantly enhanced, by the users themselves. Navizon’s software is currently compatible with Windows mobile devices and cell phones using the Symbian operating system, and is free for non-commercial use. Navizon members are allowed access to a massive storage and retrieval repository containing highly accurate positioning data which is contributed from members around the world. “The soft launch of Navizon has been successfully tested, and now we are very excited to grow the community even more,” said Houri. “Navizon utilizes WiFi and cellular signals to help you find your way around most cities and to provide you with pertinent information about your location. What makes Navizon unique is that the network will continue to expand as the community expands. This shared information is at the very center of what makes Navizon a powerful tool.”

Vonage Now Provides E911 in Over 1700 Calling Centers

New Jersey-based VoIP service provider Vonage Holdings Corp. now has E911 service in over 1,700 PSAPs across the country. Vonage claims that, in just one week, the company equipped an additional 112 calling centers in more than 50 new counties with 911 calling capabilities. “Vonage has been continuing to quickly roll out Enhanced 911 service to new counties across the U.S.,” said Jeffrey A. Citron, Vonage’s chairman and CEO. “Our goal is to get every single Vonage customer help when they need it, and we will continue working with the FCC, regulators, Congress and public safety until there is equal access to E911 for Vonage customers.” The VoIP provider currently offers E911 access in the U.S. by sending the call, along with the customer’s address and phone number, to the proper local emergency call center based on the caller’s street address. The caller’s info is then displayed on the dispatcher’s screen whenever they dial the digits 9-1-1 from a Vonage phone. In the event local authorities cannot display the Vonage customer’s phone number or address, Vonage offers basic 911.

ICOA Powers WiFi at Stop & Shop Locations

ICOA, Inc., a national provider of wireless broadband Internet networks and managed services in high-traffic public locations, announced that it is powering Stop & Shop’s WiFi initiative. Stop & Shop tested the technology in their corporate headquarters waiting room this summer. Based on the success of that trial, the company, which has more than 360 stores throughout New England, New York, and New Jersey, has begun rolling out free amenity wireless Internet access for its customers. The first store to offer this service is Stop & Shop’s South Bay Plaza outlet in Dorchester, Massachusetts. “We are pleased to have been selected by a service oriented company such as Stop & Shop,” said ICOA’s CEO Rick Schiffmann. “Providing Internet access in their cafe seating area is another service that helps their customers accomplish more during their busy day.” “The ability to stop and surf, and maybe have a meal or a cup of coffee is yet another convenience for our customers,” adds Stop & Shop’s Mike Drumm. “We look forward to working closely with ICOA during the rollout to optimize our overall customer experience.”

Ekahau Deploys WiFi-Based Location Technology for Emergency Management

In the event of an earthquake or train derailment that results in mass casualties, hospitals are confronted with simultaneously treating hundreds of patients with varying degrees of injuries. To meet the challenge of managing patients, checking the availability of doctors and locating open beds and medical equipment in wide-scale emergency, Nagoya Ekisaikai Hospital in Japan has deployed and successfully tested the Real Time Location System (RTLS) from Ekahau Inc. The Ekahau RTLS is a mission-critical location tracking solution that easily integrates with WiFi networks that are already in use in many large public and private facilities, such as Nagoya Ekisaikai Hospital. “The ability to incorporate a location tracking solution on top of existing WiFi networks provides hospitals and other facilities with a cost effective way to prepare for natural or man-made disasters and mitigate the chaos associated with these events. The Ekahau RTLS gives hospital staff the realtime information they need to ensure quality care and ultimately save lives,” said Jarmo Ikonen, Ekahau’s director of sales, EMEA/APAC. “Hospitals represent just one target segment for such a location-aware emergency management system. Enterprises and public agencies, such as those tasked with homeland security, also can benefit from this technology.” During a recent disaster simulation drill, Nagoya Ekisaikai Hospital put WiFi tags on each hospital staff member and triage patient. Once the exercise was complete, Nagoya Ekisaikai Hospital reported that with the T201 WiFi tags and the Ekahau Positioning Engine, it was able to locate patients and staff with an accuracy of three to five meters.

BellSouth and 8x8 Team Up To Deliver Residential VoIP

BellSouth Corp. announced today that it will be partnering with 8x8 Inc. (news - alert) in launching a new residential VoIP service. The new service, BellSouth Digital Phone Service, will use 8x8’s Packet8 service. Martin Chandler, vice president of product management for BellSouth, said in a company press release that the new service “is another example of our commitment to providing customers with the greatest choice when it comes to their communications and entertainment services.” The BellSouth Digital Phone Service is based on 8x8’s internally developed technology. This includes a suite of VoIP service components including a call switching platform, feature servers, customer portals and consumer premise equipment. “BellSouth’s selection of 8x8 as its VoIP partner is a tremendous credit to the technology and service expertise which has made 8x8 one of the industry’s leading VoIP service providers,” said 8x8 chairman and CEO Bryan Martin. “Our extensive residential, business and video technology portfolio was a key component to our winning BellSouth’s business. We look forward to working closely with BellSouth to offer their customers an enhanced, robust Internet phone service alternative.”

Dash911 to Launch E911 Service for VoIP Providers

“Dash911’s architecture offers coverage to any telephone number, and with our SOAP API interface, we can quickly implement a VoIP provider within ten days,” said Michael Giagnocavo, Dash911’s CTO. Dash911 is “telephone number provider agnostic,” therefore providing E911 services to any USA telephone number, at any USA address.“Our relationship with our backbone provider allows us to provide an incredibly rich service, because they are the largest and most experienced company in the emergency calling services field,” noted sales engineer Denise Ferrara. This news follows Dash911’s request in October for a 21-day extension for all of its customers from the FCC. The FCC had required that all VoIP providers deploy an E911 solution for subscribers by November 28, 2005. Dash911 requested an extension for its customers “due to the unprecedented damage caused by hurricanes Katrina, Rita and Wilma which has caused administrative hardships and infrastructure disruptions for many smaller VoIP companies.”

Verizon Web Calling powers Windows live messenger's PC-to-phone calls.

The more than 250 million users of Windows live messenger can make high quality PC-to-phone calls to any landline and mobile phone in the world through Verizon Web Calling. I mentioned in my last post that the company behind microsoft call termination for the windows live messenger is Verizon. This they are doing through their VoIP service named Verizon Web Calling. To be able to make low cost phone calls to any telephone number in more than 220 countries from your windows live messenger you have to sign up for Verizon Web Calling and purchase pre-paid calling credit which are in $5, $10 or $25 blocks. Calls to USA, Canada and a host of western european countries cost less than 2 cents per minute and calls to other destinations are at low VBC rates. At the moment those that sign up for the service will get up to one hour of free calling."Verizon uses the best and most reliable technologies to provide customers with a wide variety of calling options, including wireless, wireline, voice over IP and other high quality voice services, said Eileen Cassidy, vice president – voice and bundle solutions for Verizon. "We’re excited about using the strength, reach and reliability of our global network to create new options for people who want to use their broadband connections to easily and affordably make calls while online. Verizon Web Calling is a great match for our customers using our popular DSL and advanced FiOS high-speed Internet services." More info at http://get.live.com/messenger/overview About Verizon ------------- Verizon Communications Inc. (NYSE:VZ), a Dow 30 company, is a leader in delivering broadband and other wireline and wireless communication innovations to mass market, business, government and wholesale customers. Verizon Wireless operates America’s most reliable wireless network, serving 53 million customers nationwide. Verizon Business operates one of the most expansive wholly-owned global IP networks. Verizon Telecom is deploying the nation’s most advanced fiber-optic network to deliver the benefits of converged communications, information and entertainment services to customers. Based in New York, Verizon has a diverse workforce of more than 250,000 and generates annual consolidated operating revenues of approximately $90 billion. For more information, visit www.verizon.com.

Apple to join the VoIP race with iChat 4.0

Sources close to Apple Computers are revealing that Apple will be entering the voip market by including voip internet dialing as one of the features of iChat 4.0 which Apple is expected to bundle with the operating system overhaul, nick-named Leopard. This will pitch iChat 4.0 which is a video conferencing and messaging application against similar voip solutions from Skype, Google and Microsoft. These move is not unexpected due to the increasing popularity of VoIP Technology and to say the the future is VoIP will actually be an understatement. Most companies in the telecommunications and internet business are now jostling to take a bite from voip market since Skype, a small software start-up in 2003 has grown to become a billion dollar business with a user base of more than 100 million.

Jajah lntroduces free Global phone-to-phone Calls

Jajah has gone a step futher with their award winning web activated telephony by introducing free global phone-to-phone calls between registered Jajah users and a host of other interesting features. Actually i call this latest release "The new improved Jajah". If you live in any of the following countries: USA, Canada, China, Singapore, Hong Kong, Taiwan,Argentinia, Australia, Austria, Belgium, CzechRepublic, Denmark, Finland, France, Germany, Greece, Hungary, Ireland, Israel, Italy, Japan, Korea South, Luxembourg, Malaysia, Mexico, Netherlands, New Zealand, Norway, Poland, Portugal, Russia, San Marino, Spain, Sweden, Switzerland, Thailand, UK, Venezuela, you can now call other registered Jajah users in these countries for free on their landline phone. Even calls to mobile phones of Jajah users in USA, Canada , China, Singapore , Hong Kong and Taiwan are also free. The calls are unlimited but users are expected to adhere to a fair use policy in order that everybody can enjoy the service. Calls to landline and mobile phones in other countries are the Jajah's competitive rates. Jajah really popularizes web activated telephony which enables one to make voip calls without the need to download any software, get headset, have broadband internet or sit in front of your PC, you just use whatever internet connection you have to connect your regular phone to the destination number and the call then becomes phone-to-phone afterwards. Jajah itself has become so popular among voip users because of its ease of use and low rates. Other new features include Jajah outlook plugin and JAJAH Mac OSX Address Book Plugin, i will give details in future posts after testing the plugins. Also Jajah now has a business account which enables one to manager multiple users from one account and Jajah API which makes it possible to embedded Jajah with your websites and applications. More details are available at Jajah website.

MITEL's Voip Solution for Share Traders

MITEL, a leading ip telephony hardware maker now offers new solutions for financial services organisation. These range of solution is a combination of resiliency with mobility and easy development.paetec comunications is widely known solution provider based in new york with more than 1500 core biz customers serves as Mitel's solution portfolio marketer in U.S.This offers intends to provide hedge funds and boutique firms with technology tools that will increase their efficiency and productivity.Mitel 3300 ip comunications platform provides inherent resilient phone connections with advanced technologies.

Microsoft Launches Windows Live Messenger

Microsoft has launched the public version of Windows live messenger. Before now only the invite only beta version was available for a select few but now anybody can download the application and use. This is beleive to be microsoft responce to the growing popularity of skype and another furray into the voip market. With windows live messenger you can make text, audio and video chat with other users with exeptionally good audio and video quality. Users can also setup a sharing folder with their friends and can drag and drop photos and other stuffs into this folder. The sharing folder can be seen even when one of you is offline. Soon users will also be able to talk to Yahoo messenger with voice. This interoperation with yahoo will futher increase the size of the worlds lagest IM network. This will also not be good news for skype which does not interoperate with any IM or VoIP client because of its proprietary architecture. Another feature is the ability to make pc-to-phone calls with really good audio quality to any regular phone in the world. Microsoft uses Verizon for termination and not IDT/Net2phone as was the case in NetMeeting. At the moment Windows live messenger does not support any SIP device. Time will tell if yahoo messenger with voice and now windows live messenger will cut into skype popularity in the voip terrain, skype by far still remain the primary voip client for lots of people. I personally found the fact that windows live messenger does not support SIP based devices and SIP-based termination provider dissapointing. I believe SIP based voip which allows easy interoperation is the best way to go for the industry. Interested? download it Here

More free minutes for Skype USA users

Skype is not stopping at free skypeout calls to USA and Canada for Skype users in the USA but they are giving them more incentives to continue to use Skype. The latest is the offer of 60 free Skypeout minutes to call about 30 countries. This offer will last till June 30. "With free calling within the United States and Canada until the end of the year, and one hour of free international calling until the end of the month, we are making it incredibly convenient for our U.S. Skype users to keep in touch with friends, family and business associates," said Henry Gomez, general manager, Skype North America. Part of the press release from the skype north America reads:As of today, all existing U.S.-based users who are new to Skypeout and opt-in to receive Skype emails will receive an email when they log onto Skype. The email contains a link that users click to redeem 60minutes of free international calling using Skypeout. Users can then start making free international calls to 30 countries immediately. Skype users in the USA can now call the following countries for free: Argentina, Australia, Austria, Belgium, Canada, Chile, China, Denmark, Estonia, France, Germany, Greece, Ireland, Italy, Korea, Mexico, Netherlands, New Zealand, Norway, Poland, Portugal, Russia, Singapore, Spain, Sweden, Switzerland, Taiwan, United States, United Kingdom. Related ======= GET SKYPEOUT CREDITS GET SKYPE ACCESSORIES GET SKYPEIN NUMBER GET SKYPE VOICEMAIL Voip-buzz stole this article

VoIP/Broadband Internet Accelerator

The D-Link VoIP/Broadband Internet Acceleratordetects and prioritizes bandwidth-sensitive packets for faster real-time processing Ideal for VoIP, online gaming, and video streaming Improves VoIP call quality and minimizes interruptions Dynamically configures itself by automatically detecting Internet upload speed Plug-N-Play use Includes AC adapter. This voip/internet accelerator can be obtained Here.

6 Port Usb 2.0 Hub With Voip Keypad

The 6 Port Usb 2.0 Hub With Voip Keypad , complete with Headset This product allows users to communicate with other users that have VoIP Telephony or Instant messenger programs that support Voice Communication, the built in keypad has volume/mute and call control, including the ability to dial, connect/disconnect calls and to access features on VoIP telephony programs such as skype. Once installed it is shown as a separate USB Audio Device, this allows independent audio connection, and enables the user to have a independant uninterrupted conversation, while listening to music or playing a game etc. The unit includes a manual switch which is used to select between the two attached audio devices (I.e. select between a audio headset or microphone and speakers) This unit also offers a 6 Port USB Hub that is compatible with the Hi Speed Revision of USB 2.0, The ports have been located on the VoIP-010 to maximise productivity by allowing easy access to the users USB Devices. Features; -Compatible with USB 2.0 Specification -Supports a Data Transfer Rate of up to 480 Mbps -Provides 6x USB 2.0 Ports -Backwards compatible with USB 1.1 -Compact desktop design -Portable for travel -Plug and play USB Connectivity -Selectable dual audio inputs/outputs - allows switching between headset and desktop microphone with speakers -Installs and Registers as a second independent USB sound device on system -Dial pad is compatible with most VoIP type applications -Power Supply included Pack Includes; -VoIP Keypad with 6 Port Hub -Driver CD -Power Supply -USB Cable -Quick Start Guide -3.5 mm Mono Headset with Mic System Requirements; -PC compatible with USB 2.0 operation -CD-ROM Drive -Windows 98SE/ME/2000/XP Please Note: These products require an existing internet connection for operation, the keypad can only be used with VoIP telephony programs, which may require subscription and/or additional cost. For more information about and to order click Here

iCall giving away free incoming numbers

iCall internet phone is given away free incoming numbers though using a slighly different approach from most internet phone providers. Instead of given a particular phone number to a user as voip-in number they assigned each user an extension number when they signed up to iCall and they have local access numbers in most cities in the US. To call an iCall user on his computer from your phone you just need to call a access number closest to you and enter his/her iCall extension number when prompted, his icall internet phone will ring and you talk to him no matter where he is in the world only for the cost of a local call. For iCall users all they have to do is to give their 4 digit extension numbers to their contacts and they can be reached on their PC anywhere in world free. Allready iCall has local access numbers in 175 area codes in the US from Alabama to Wisconsin. With iCall you can call PC to PC free and unlimited and calls to regular phones in US and Canada are also free. Visit iCall website for more information and download.

Ubiquity Launches Ubiquity Developer Studio to Speed Deployment of SIP Applications

Ubiquity Software,developer of standards-based SIP (Session Initiation Protocol) application servers, announced the launch of Ubiquity Developer Studio (UDS), a development and debugging platform for its SIP Application Server (SIP A/S). UDS provides a simplified set of tools that enables SIP developers to build highly modular applications and services that run on top of the Ubiquity SIP A/S. UDS is designed to work in conjunction with the Ubiquity Appcelerator product set, an optional add-on to SIP A/S. Appcelerator provides a unique SOA including a Service Oriented Object Framework (SOOF) with reusable code components. And by using Eclipse, developers get an enterprise or "IT" perspective on real-time telecommunications applications software development, which makes it easier to develop converged applications that work with carrier services or within the enterprise. Applications created with UDS can be built tested and debugged in Ubiquity Developer Edition, so you can test and debug right on your desktop.

Dialexia and Grandstream in SIP-eroperability Testing

VoIP solutions provider Dialexia Communications and Grandstream Networks, (news - alert) which provides IP telephone hardware, have successfully completed comprehensive interoperability testing between Dialexia's IP-PBX Dial-Office and Grandstream's GXP2000 SIP IP Phone. The partnership is set to allow future service providers and enterprise customers easy VoIP deployments. With the need for connecting distant branch locations, while maintaining the functionality of existing PBXs, customers are looking for an edge, which often is found in ease of implementation - and cost, of course. Dialexia's flagship product, Dial-Office, is one of the lowest cost VoIP solutions available to enterprises. It is designed to easily integrate with legacy TDM and IP networks, thus offering customers the dramatic savings that often drives them to a VoIP solution. Grandstream Networks' GXP-2000 is a next-generation enterprise IP telephone based on open SIP standards. Built upon unique and proprietary technologies, it features superb audio quality, rich functionalities, and excellent manageability at affordable prices. The Grandstream GXP2000 IP-Phone combined with Dialexia's Dial-Office brings a rich feature set and extraordinary value to both consumer and corporate markets. With adherence to the SIP standards the products are well suited for enterprise users of IP PBX systems, remote office users, and many other cost saving and productive uses.

RADVISION Releases New SIP Toolkit With

RADVISION, a provider of multimedia conferencing and communications platforms, has released a Java version of its Session Initiation Protocol (SIP) toolkit for real-time communication applications, such as VoIP and IM. The new toolkit significantly shortens development cycles, thereby enabling even faster time-to-market for a variety of new and innovative services.Java programming language, which is rapidly gaining popularity among developers, can be used to write software on one platform and run it on another,create programs that run within Web browsers, develop server-side applications, and write applications for cell phones and other consumer communication devices. Java SIP is a fundamental building block for Web-based endpoint-oriented communication applications. The Java SIP Toolkit can be used to develop click-to-talk applications, as well as a variety of applications for web call centers, and soft phones for laptops and PDAs. It also can be used as a major building block for innovative and time-critical advanced telecom services on the server-side. Similar to all of RADVISION's SIP toolkits, the Java SIP toolkit addresses carrier-side applications for softswitches and application servers. The Java SIP Toolkit supports operator-grade loading of busy hour call attempts (BHCAs) and has enhanced carrier grade features, including advanced SIP signaling functionality. The small footprint allows efficient operation in SIP terminals and IP handsets, as well as network-side implementations.

UPC Broadband Signs One Millionth VoIP Customer

UPC Broadband, the European broadband division of Liberty Global, Inc., whose networks pass more than 11 million homes and businesses in Western Europe is using solutions from both Nortel and ARRIS, as well as several others, in growing its European network. In fact, over the past 15 months, UPC's telephony deployment has doubled and it has now signed its one millionth VoIP customer. UPC is also one of the first cable operators to deploy SIP (Session Initiation Protocol) -based VoIP services, laying the foundation for new, feature-rich voice and multimedia services, like presence, instant messaging, video and mobility, and taking the first step towards the converged services. "As consumer and business demand for multimedia services grows, operators want scalable, reliable and open VoIP solutions that deliver services in a timely, reliable and cost-efficient manner without abandoning existing network investment," said Steve Pusey, executive vice president, Nortel and president Nortel Eurasia. UPC provides high quality VoIP service that is attractively priced to offer a real alternative for consumers to their existing telephone service. Its deployment of VoIP-based equipment - like softswitches and gateways from Nortel and customer premise equipment from ARRIS - allows UPC to provide enhanced triple play services to its customers. There are plans to develop a VoIP/SIP-based converged product portfolio over a single IP network to both fixed and wireless networks. The goal is to introduce integrated fixed/mobile services in Switzerland and other key markets based on these SIP softswitches.

Genesys Readies Enterprises for IMS World with Release of 7.2

With its release of the newest version of its flagship contact center software, Genesys Telecommunications laboratories is betting that enterprises will eventually fix their IP communications switching infrastructure on the Session Initiation Protocol (SIP) and incorporate elements of the IP Multimedia Subsystems (IMS) architecture. The centerpieces of the Genesys 7.2 release are the Genesys Customer Interaction Management (CIM) platform, a framework that adapts to any media channel, and the Genesys SIP Server, which acts as a software-based automatic call distributor (ACD) to direct calls to up to 30,000 IP agents through multimodal customer contact channel options that include voice, e-mail or chat. This new standards-based approach to call control and signaling means Genesys 7.2 can freely communicate with media gateways and softswitches without having to conform to a myriad of different variations on the Media Gateway Control Protocol (MGCP). In addition, the company has incorporated the Real-time Transport Protocol (RTP) to manage the media elements within the network, as outlined by the IMS specifications. Meanwhile, the Customer Interaction Management (CIM) platform features an Open Media software component that tightly integrates IP into key elements of the suite, such as the Genesys Voice Platform to support voice selfservice, video and voice-enabled applications. As such, Genesys 7.2 is able to incorporate business process routing (BPR) capabilities by applying intelligent real-time routing, tracking, reporting and management capabilities to work items and interactions over any channel inside the contact center and across the enterprise.

DiVitas Networks Announces Interoperability with Trapeze Networks' WLAN

DiVitas Networks, a provider of seamless, unified mobility solutions for enterprise networks, announced that it has successfully tested the interoperability of its platform with Trapeze Networks, (news - alert) the provider of the wireless LAN (WLAN) Mobility System, MP-372 access point, and new MX-200 and MX-216 series of switches to enable enterprise campus mobility - and the industry's first demonstrable Fixed Mobile Convergence for WiFi and Cellular. DiVitas and Trapeze successfully tested a VoIP-over-WiFi application, handing off VoIP calls from the WLAN to the cellular network and back for seamless roaming over the best available network. Leveraging cellular networks, Internet/Wi-Fi technologies, and dual-mode handsets, DiVitas delivers mobile voice and text on a single platform, both within the enterprise and outside it. The DiVitas solution also serves as an application platform for enhanced communications capabilities at extremely attractive cost points. Above all, the DiVitas solution is controlled by the enterprise. Administrators can now centrally manage mobile communications independent of the network that delivers them.

VoiceXML Gains Acceptance as

Microsoft announced that Speech Server 2007 - the latest version of the company's speech and telephony platform - will provide full support for VoiceXML. VoiceXML is an open standards-based language used for developing speech applications, managed by the World Wide Web Consortium (W3C). Microsoft said the addition of VoiceXML compatibility (Speech Server continues to support Speech Application Language Tags, or SALT) will "enable customers to choose the development standard that will work best in their environment." According to Clegg Ivey, Vice President of Operations at Voxeo, Microsoft's adoption of VoiceXML is a good indication that the language has matured into the industry standard. "When Microsoft signs on to a standard, you know you're in good shape," he said.Because of the VoiceXML standard, companies can deploy speech products knowing that when they upgrade or switch vendors down the road, they won't have to re-write all their applications. VoiceXML can be used to create any type of interactive speech application, including voice-enabling of call centers, interactive tracking applications, and outbound notification.

The New PacketCable: SIP ... and More

The new specs announced this week by CableLabs Inc. for its PacketCable system bring the cable industry closer to adopting SIP, but still refrain from specifying SIP as the only option for cable operators deploying VoIP services. The current signaling system used by cable operators, called NCS, does not allow the portability that SIP supports. Unlike SIP, customers cannot plug their IP phones or terminal adaptors into the Internet anywhere and get service. Therefore, the services closely resemble the PSTN, with a few extras, such as Web interfaces to configure features. The changes to PacketCable signal that the leading operators in the cable industry will continue their relatively conservative approach to VoIP in the short term, but the specs also open the door for future services, such as WiFi VoIP and combinations of voice and other media, that take advantage of the portability of SIP and its support for multimedia. Also included are select aspects of IMS, chosen for their relevance to the cable industry. Cable operators now have a mechanism to do both voice and video telephony. They can combine services across platforms, for example, sending caller ID to a set-top to display on a television. There are also ways to integrate fixed and mobile services. "It gives operators ways to integrate across all those services," Miller says. The specs developed by CableLabs, the technology development center for the North American cable industry, also deal with some of the knotty deployment issues of SIP VoIP, namely security and network address translation (NAT) firewall traversal. Other new features address QoS and device provisioning.

Cisco Expands Unified IP Phone Portfolio

Cisco Systems has introduced a number of new or enhanced IP telephone models, including the 7911G (Global) and 7941G-GE and 7961G-GE (Global and Gigabit Ethernet) models. All support IEEE power and advanced SIP capabilities. Recently, Cisco also announced Cisco Unified CallManager 5.0, a new software version of the company's IP PBX that runs on the Linux operating system and adds native support for SIP (SIP line and trunk support) in addition to Cisco's existing protocol, Skinny Client Control Protocol or SCCP. With native support for SIP Cisco customers and channel partners will have a choice of endpoint devices from various vendors for a customized solution. Further flexibility is available to Cisco customers since they can choose devices based on SIP or Cisco's SCCP Cisco's IP Phone 7911G is the company's newest basic IP phone with the same hardware design of Cisco's earlier IP Phone 7912G, including one line and four soft keys for access to popular telephony features and functions. Designed for business employees with low to medium telephone usage such as those in retail, school or manufacturing environments, the 7911G adds enhanced memory, IEEE 802.3af Power over Ethernet and advanced security. XML support enables text-based applications (including double-byte and unicode characters) to be viewable on the monochrome display. The 7911G works with Cisco's CallManager 3.3 software (or higher) and CallManager Express 4.0.

App Server Vendors Prep for Imminent Combination of IT, Telecom

Application server vendors are accelerating their rollout plans to prepare for the imminent blending of information technology with the telecom world, brought on by the influx of IP Multimedia Subsystem (IMS)-based architecture. Oracle has outlined its roadmap to deliver a standards-based Service Delivery Platform (SDP) for the telecommunications industry to transition from current silo-based network investments into a service-oriented architecture (SOA) to help reduce time and costs to deploy next generation communication Internet Protocol (IP) voice and data services. Oracle SDP plans to extend its Oracle Fusion Middleware platform with technologies obtained through its recent acquisition of HotSip, adding elements like Session Initiation Protocol (SIP), Presence Server, Proxy Registrar, and Location components. Oracle's announcement underscores its desire to buy its way into the lucrative telecom market, which is witnessing an economic upswing as standards like IMS and Linux enable the convergence of delivery platforms. "IT-standards-based service delivery platforms offer compelling value to operators as the basis for developing interactive, media-rich, next generation data services," said Philip Marshall, director, Wireless/Mobile Technologies, Yankee Group. "Service delivery platforms that enable immediate ROI through out-of-the-box services and integration with [operational and business support systems] through standard interfaces are likely to be particularly compelling. Vendors who are able to provide a broad portfolio of products that are stable, mature and carrier-grade will have a definite edge over the competition."

Avaya Compliant Compass Guides Expand Networks

Expand Networks, a leading provider of application acceleration solutions over the WAN, announced that its Expand Compass platform is now “Avaya compliant” — meaning that it is compliant with key Internet protocol (IP) telephony solutions from Avaya. The Expand Compass application acceleration platform integrates multiple technologies and maps them to the corresponding business mandates. A key initiative is Expand’s voice and data convergence capability, along with its QoS feature, which can detect and prioritize IP telephony traffic to ensure it receives the quality of transport required for superior call performance — a feature whose effectiveness can be maximized by matching it with Expand’s WAN compression capabilities. WAN compression increases the available bandwidth for all applications. Avaya-compliancy ensure’s Expand’s customers that their Avaya VoIP traffic will have the dedicated bandwidth needed for high-quality calls.

NTELOS Readies for IPTV Service with Skystream

Telecom carriers and service providers are constantly looking to improve their service offerings to retain and attract customers. Today, a large part of that is offering converged video and data services. But to do this, they must have a cost-effective migration plan for moving their networks to an IP-based architecture and shortening their time to market. That means they must find a suitable technology partner. NTELOS Inc., a Waynesboro, VA-based communications provider, has found its partner — it has selected SkyStream as its video head end partner for a new IPTV service over Fiber-to-thePremise (FTTP) for its Virginia customer base. Skystream has expertise in video and data content delivery infrastructure and will help NTELOS add new capabilities to its network to reliably deliver additional services and content. NTELOS has built a “super video head end” center, where it has deployed the SkyStream Mediaplex-20 video head end. NTELOS is distributing the MPEG-2 feed over a 10-Gigabit Metro Ethernet ring to several regional offices and will deploy the SkyStream iPlex edge video head end system for additional aggregation and distribution. NTELOS will be able to offer local channels, advertising and other programming from these regional centers.

Emergent Network Solutions Announces ENTICE 2.4

Emergent Network Solutions, Inc. announced ENTICE 2.4, a significant new release of its VoIP call control platform. Additions and upgrades in Version 2.4 include support for SOAP based Web services, a new traffic monitoring and reporting module, and dynamic least-cost routing. As part of Emergent’s continued progression towards IMS compliance, ENTICE has been enhanced with advanced mobility features for roaming SIP endpoints and new security features enabled by the addition of p-header support. ENTICE (Emergent Network Telecommunications Infrastructure Control Environment) is the core operating system that drives the Emergent VoIP product line. Suitable for either existing or greenfield environments, ENTICE provides complete control services and a software foundation to layer new services. ENTICE products include the ENTICE Softswitch, ENTICE Session Controller, and various gateways products and service offerings. ENTICE 2.4’s improvements provide greater flexibility and value to Emergent’s entire product line with particular advantages to the E-REV solution, a turnkey VoIP solution for ISPs, whole sale carriers and CLECs.

AT&T Inks Voice and Data Contract with Advanced Healthcare

AT&T Inc. announced that Advanced Healthcare, a physicianled, multi-specialty clinic group in southeastern Wisconsin awarded the telecom giant with a new optical and data-networking services contract. The new, five-year contract calls for AT&T to expand Advanced Healthcare’s bandwidth through a deployment of a fully-redundant Synchronous Optical Network (SONET) ring to six locations throughout the state. With the additional bandwidth, Advanced Healthcare will be able to consolidate large amounts of voice, data, and video traffic from its various locations to a single, fail-safe platform. The new services will also support an X-ray Picture Archiving and Communication System (PACS), which can store, display and transmit digital medical images across Advanced Healthcare’s network. Additionally, because of its increased capabilities, Advanced Healthcare will have the capacity to set up a disaster recovery site that mirrors the current data center. AT&T is currently the primary provider of voice and data services to Advanced Healthcare

Openwave Joins MobileIGNITE

MobileIGNITE, the leading industry association fostering collaboration to accelerate fixed-mobile convergence (FMC), today announced that Openwave, a leading independent provider of open software products and services, is joining MobileIGNITE. Openwave has agreed to the group's tenets of working on open standards-based interoperability and leveraging SIP (Session Initiation Protocol) with a commitment to IMS (IP Multimedia Subsystem), the industry's leading blueprint for convergence. MobileIGNITE's Interoperability group is working to define common use cases, interoperability best practices, and test cases from the baseline open standards on which FMC and other IMS-related applications are based. "As the line fades between mobile and VoIP networks, carriers will be able to give consumers adaptive messaging services that are independent of network, device, or protocol used," said Richard Wong, senior vice president, products and solutions, Openwave. "We look forward to becoming active in the MobileIGNITE association to ensure industry interoperability and accelerate development and deployment of converged services."

New Global Telecom Launches IPower MSC Service

Telephony solutions provider New Global Telecom (NGT) announced the launch of its new IPower session border control leasing service. The service gives carriers exclusive access to license and support for a NexTone Multiprococol Session Controller (MSC). The IPower service will be sold under NGT’s General Telecom brand, complementing the company’s established IPartition hosted softswitch services. IPower provides CDR reporting, enhanced call rating, CDR-based alarming and automated least cost routing capabilities,among other features. Additionally, service providers using IPower rental services will have access to stand-alone MSC (collocated at NGT’s facilities), bandwidth to handle higher traffic volumes, maintenance support, and 24/7 monitoring.

Telsima Launches Complete WiMAX Solution

Cellular carriers and WiFi solution providers have had a strong run in recent times. But both consumers and business customers want more — they want the range of cellular AND the speed of broadband. Which is precisely what WiMAX is out to deliver. Santa Clara, California-headquartered Telsima Corporation has launched StarMAX, its carrier-grade WiMAX solution for WBA. The product family comprises a complete suite of WiMAX solutions, including base stations, subscriber stations, GUI-based management system, and a provisioning and mobility manager. Telsima sees WiMAX as the ideal last mile solution for broadband connectivity and will leverage the benefits of standardization, innovative design, and large market size to market its latest product set to a group of operators that requires standards-based, easily and quickly deployable, broadband service for their enterprise and/or residential customers.

Metalink Delivers HDTV via WiFi

Metalink, a provider of high performance wireless and wireline broadband communication silicon solutions, announced that it has concluded a series of live demonstrations of its WLANPlus chipset, a technology based on the emerging 802.11n standard, with unique capabilities for wireless multimedia distribution. For the first time, advanced Wi-Fi technology was used to demonstrate the delivery of multiple simultaneous high-definition television (HDTV) streams anywhere in the home, with full quality of service (QoS) guarantees. Metalink’s WLANPlus family consists of the company’s MtW8170 baseband device and the MtW8150 radio frequency integrated circuit (RFIC). WLANPlus has been developed in line with draft specifications for the future 802.11n high-speed Wi-Fi standard, which were unanimously approved by the 802.11n Task Group (TGn) in January 2006. Metalink continues to play a pivotal role in the standards finalization process, and is at the forefront of efforts to optimize 802.11n technology for demanding wireless video entertainment applications in such consumer electronics products as digital video recorders (DVRs), set-top boxes (STBs), high-definition televisions (HDTVs) and media adapters.

Meru Networks Now Compatible with Blackberry WLAN Device

Meru Networks, which provides wireless VoIP infrastructure worldwide, has announced a comprehensive solution for on-premise wireless voice and data applications with support for the BlackBerry WLAN solution from Research in Motion (RIM). Meru has certified interoperability of the Research in Motion’s BlackBerry 7270 handheld with Meru’s WLAN System. BlackBerry’s WLAN solution includes the BlackBerry 7270 handheld, which connects to a BlackBerry Enterprise Server to enable secure wireless access to email, instant messaging (IM), organizer, Internet, and other corporate applications. It also allows wireless VoIP calls through enterprise telephony systems, such as IP-PBXs, via the industry-standard SIP. Enterprise customers can now use the BlackBerry WLAN solution to access a reliable, scalable enterprise WiFi network for high-performance, on-premise wireless voice and data applications. Unlike other WLAN systems, where APs (access points) work independently and, therefore, need to be individually configured to deliver adjacent, non-overlapping coverage, Meru’s WLAN System uses a coordinated, cellular architecture in which all APs work in tandem. Thus, this solution provides a seamless blanketing with transparent scalability and zero handoff times between APs. It also affords QoS assurance to each WLAN user — now including users with the BlackBerry 7270 device. Meru’s bi-directional QoS technology has is designed to identify SIP-based traffic and to provide it priority over other traffic, to assure dial-tone quality wireless VoIP.

Netrake and Kineto Wireless Now UMA-Interoperable

Netrake and Kineto Wireless (news - alert) announced that they completed all interoperability testing procedures to support currently installed UMA systems. According to the companies, Netrake’s nCite Security Gateway and Kineto’s IP-based UMA Network Controller (IP-UNC) are currently deployed in commercial trials for UMA-based mobile/WiFi convergence. Netrake’s nCite Security Gateway bundles voice and multimedia in encrypted IP tunnels for wireless usage over the public Internet. The system uses IPSec encryption and establishes multiple media and signaling security associations without adding network latency. The company’s Secure Edge Architecture offers Denial of Service (DoS) attack prevention, firewalls, intrusion detection services, IMSI filtering, blacklisting, impersonation detection, spoof prevention, dynamic virtual routing, and Network Address Translation (NAT). Kineto’s mobile/WiFi convergence system offers the Kineto IP-UNC, a carrier-class network component that integrates with a service provider’s existing voice and packet switching systems. The technology uses standard interfaces to extend higher-performance mobile services over IP access networks and WLANs. The company’s IP-UNC manages and secures subscriber’s access to mobile, voice, data, and IMS services from WLAN locations and facilitates automatic, roaming and handover between licensed radio access networks and WLANs.

WifiTastic Converts Wireless Networks Into Public WiFi Hotspots

WifiTastic, LLC. introduced its public BETA program, which enables broadband subscribers to create a revenue-generating hotspot providing high-speed, wireless Internet access to users of WiFi-ready laptops, PCs, Macs and PDAs. setting up a commercial hotspot, owners can earn money by charging people in the vicinity of their wireless router to connect for a fixed hourly, daily or monthly fee. While the company provides a suggested pricing model, WifiTastic owners can determine their own rates. Users can connect to the router, pay by credit card, and then access the Internet. WifiTastic handles the billing and returns 60% of the proceeds directly to the hotspot owner. Currently, the service is compatible with the Linksys (news - alert) WRT54G router. Users can simply purchase a pre-configured router or update their existing Linksys router by downloading and installing a free firmware update from WifiTastic.

EV-DO VoIP to be available soon

Nortel and Sierra Wireless (news - alert) have achieved a significant technology milestone needed to power wireless services such as push-to-talk, push-to-video, VoIP and interactive gaming at much faster speeds than what is currently available. The two companies achieved the industry’s first over the air test calls using EV-DO Revision A wireless technology and pre-commercial wireless data devices. The successful tests confirm the commercial viability of EV-DO Revision A for powering next-generation broadband wireless services. Operators will be able to prioritize different users and applications, enabling tiered services and multiple pricing options. Operators also can make additional security enhancements by using authentication mechanisms that can identify users and give specific access rights based on the user’s profile. EV-DO Revision A technology will offer significant performance improvements to both the uplink and the downlink speeds compared to current EV-DO (Release 0) networks. EV-DO Revision A enables peak data download rates of up to 3.1 Mbps and upload rates up to 1.8 Mbps.

XO Interactive Deploys Pactolus RapidFLEX Media Server

Pactolus Communications, developer of SIP-based IP voice services, announced that its RapidFLEX Media Server with VoiceXML interface is being deployed by XO Interactive. XO Interactive, a subsidiary of XO Communications, (news - alert) provides hosted interactive voice response (IVR) services. The company will use Pactolus’ media server to support seamless integration of advanced technologies and rich customer content, ensuring service reliability both on an ongoing basis and during peak periods of demand. Pactolus’ SIPware Calling Card application also previously was deployed by XO Interactive, on behalf of a major telecommunications service provider that distributes cards in retail outlets throughout North America.

VoIP Monitoring System

Empirix Inc., which helps organizations adopt complex communications solutions with confidence, announced the latest release of its carrier-class VoIP monitoring solution, Hammer XMS 1.5. Hammer XMS now features a simplified user interface specifically designed for first-level support representatives, enabling rapid problem identification and more efficient escalation; and enhanced customer reporting that carriers can use to give their service provider customers web-based access to reports on their own service metrics. Hammer XMS 1.5 also adds support for H.248, SIP-T, SIGTRAN and ETSI ISUP protocols — requirements of the world’s top service providers for VoIP service monitoring.

Nortel Tunes in to IPTV

Only days after reaching an agreement on insurance payments in two class-action lawsuits, Nortel is leveraging its SIP-based services to develop new IPTV functionality that promises to change the way people use their TVs for both entertainment and communications. At its IPTV lab in Ottawa, Nortel is demonstrating the delivery of IPTV services over its IMS infrastructure, the first step for enabling telcos to create personalized IPTV services across wireline and wireless networks that helps to provide long-term differentiation from competitive offerings. Demonstrated services include mobile-to-TV picture sharing, on-screen instant messaging and presence, and click-to-call functionality. Nortel is also enhancing its integrated voice and text communications capabilities built using a new jointly developed Application Programming Interface (API) that enables the integration of real-time IPTV services with Minerva’s iTVManager software. Nortel is using this jointly developed interface as a first step to bring together Minerva’s IPTV middleware platform with Nortel’s IMS solution to make content available to users anywhere, anytime, on any device.

Alcatel Intros 7710 Service Router

French technology provider Alcatel (news alert) introduced its 7710 Service Router, the company’s compact, optimized service routing platform with a throughput of 12 Gb/s. The device’s compact form factor, variety of lower-speed interfaces, and purpose-built high availability, would seem like a great option for mobile operators and service providers with smaller points of presence (POPs) and emerging markets. The 7710 SR incorporates all features of the Alcatel 7750 Service Router portfolio, including the high availability and Quality of Service (QoS). The addition of the 7710 SR will enable IP network providers to support voice, video and data services in an alwayson, highly resilient manner.

Ditech Integrates New Voice Quality Management Solution

Ditech Communications Corporation, (news - alert) provider of voice processing and border service solutions, announced it will integrate Voice Quality Management (VQM) into its Packet Voice Processor. The addition will provide enhanced border services for providers. VQM has the capability of monitoring calls on an individual, real-time basis as well as performing basic codec transcoding and voice quality enhancement functions at the same time. VQM uses SIP-based signaling and monitors voice quality for packet delays, loss and jitters — even speech faults like background noise, hybrid and acoustic echo. With VQM service providers can further improve their call quality and better perfect their networks.

Veraz’s New ControlSwitch Eases Migration to IMS

Service providers migrating from traditional voice networks to IP multimedia subsystem (IMS) networks now have some help from VoIP solutions provider Veraz Networks. (news - alert) The company announced the launch of its updated ControlSwitch product. ControlSwitch allows providers to make the switch to IMS without sacrificing the infrastructure investments they’ve already made. It uses Service Control Multi-Protocol (SCMP) capabilities to mix and match legacy services (such as tollfree) with new multimedia services (such as messaging). ControlSwitch is also programmable, so providers can adapt it to meet their unique network needs.

Aculab Releases Upgraded Prosody S

Aculab released the latest version of its host media processing software, Prosody S. The anticipated upgrade to Aculab’s popular HMP software was specifically designed to minimize the problems that are normally associated with IP telephony, offering developers greater flexibility with a new range of functions. The upgrade of Prosody S is able to run up 360 half duplex channels, 180 full duplex channels, and 300 conference parties in any sub-group combination with all the necessary speech processing functions on the same server as the host application. Mike Matthews, Head of Product Marketing for Aculab, explained that conferencing requirements vary greatly, ranging from just a few people to broadcast style conferences, like a CEO’s end of year bonus announcement. During larger conferences, it becomes important to identify the speaker — which is Prosody S conferencing active speaker detection feature. “The flexibility of the conferencing functions for the application developer sets Prosody S apart from other products,” said Matthews.

CLECs Deploy Tekelec’s VoIP and Broadband-Enabled Services

Tekelec announced that two competitive local exchange carriers (CLECs), Burlington Telecom and Crocker Telecommunications have deployed the Tekelec 7000 Class 5 Packet Switch and Tekelec 6000 VoIP Application Server to deliver voice, Internet, and video services. The Tekelec 7000 is a softswitch that integrates with IP and time division multiplexing (TDM) network interfaces, enabling a smoother network migration to VoIP. The Tekelec 6000 VoIP Application Server integrates with Tekelec’s switching platforms to enable delivery of voice, data, video and wireless services. The new agreement enabled Burlington Telecom, the new fiber-optic public network operated by the city of Burlington, Vt., to deploy Tekelec’s switching platforms to enable the launch of triple-play broadband Internet, cable TV and voice services early this year for the city’s residents, businesses, and government agencies. Greenfield, Mass.-based Crocker Telecommunications delivers voice and data services, including broadband Internet, ISDN, co-location and Web-based hosting to business and residential customers in western Massachusetts.

CallTower IP Call Center Suite Integrates Cisco Solution

CallTower has introduced the new CallTower IP Call Center Suite, a hosted end-to-end voice and data solution that includes the Cisco Unified Contact Center Enterprise Edition. The integration of Cisco’s solution into its product suite will enable CallTower to provide sophisticated call center capabilities for growth enterprise businesses at the flip of a switch. The new solution will allow the mid-sized and growing businesses to seamlessly add call center operations and capabilities to their portfolio. They can now leverage their existing phones and infrastructure across multiple offices, without having to purchase new hardware. The CallTower IP Call Center Suite enables the representatives and sales agents to be based anywhere, segment customers, monitor resource availability and deliver each contact to the most appropriate resource. Further it allows a single agent to support multiple interactions simultaneously regardless of the communications channel the customer has chosen. Companies now have the choice to expand or contract their call center operations based on seasonal requirements. Administrator and user capabilities can also be deployed across a single site or multiple locations, including remote service and support personnel.

TeleTech Boosts on Demand Solution

TeleTech Holdings, Inc., a provider of customer management and business process outsourcing (BPO) services, announced that it has added new functionality to three key suites of its TeleTech OnDemand hosted contact center offering, which was designed to improve the customer experience and reduce the demand for live agents. The new features support easy-to-access information that responds quickly and accurately to customers’ needs, in addition to real-time measurement and coaching to enhance agent performance. Efficient and user-friendly interactive voice response (IVR) and Web services empower customers by automating routine interactions. These features are now part of the TeleTech OnDemand suites for customer experience management, workforce optimization and multichannel interaction routing. By employing TeleTech’s hosted contact center solution, clients can realize more than 50 percent in cost savings over building, integrating and maintaining their own environments, according to the company. Furthermore, the TeleTech OnDemand services can scale easily and quickly to accommodate evolving client needs, delivering business agility and flexibility, which is critical in today’s contact center marketplace. Enhancements have been incorporated into the customer experience management, workforce optimization and multichannel interaction routing suites.

Covad behind Bandwidth.com and CompUSA’s VoIP Offer

Covad Communications Group Inc. has been chosen to be a preferred DSL and T-1 provider to Bandwidth.com for its VoIP system to be sold through an agreement with office supplies retail chain CompUSA Inc. Bandwidth.com will launch its VoIP system targeting small businesses through CompUSA’s retail locations, commercial sales and Web site. The service is a fully-hosted IP telephony system that bundles Bandwidth.com’s platform and feature set and will be available over Covad T-1 and SDSL voice-optimized access (VOA). Covad’s VOA enables small business customers to conduct multiple phone calls and access large data files concurrently over the same connection. The company controls the flow of voice traffic over its network and prioritizes telephone conversations over data traffic.

Aspect Software Enhances the Patient Experience

Contact center products supplier Aspect Software is making life a little easier for patients calling into Evanston Northwestern Healthcare. Located in Chicago, the healthcare facility wanted to put itself ahead of the city’s health care competition by offering better patient care and services, both of which were made possible with the implementation of 190 Aspect EnsemblePro seats in its contact center. “The patient experience doesn’t begin when they step into the hospital or doctor’s office, it begins the minute they pick up the phone and we believe that technology, like Aspect EnsemblePro, is helping us achieve our strategic business objective — making the patient experience as valuable and as pleasant as possible,” said Fran Horner, senior director of patient access for Evanston Northwestern Healthcare in a press release. Horner also noted that the facility’s goal is to ensure that patients are receiving the same care on the phone as they would if they were waiting in the hospital. The contact center supports 23 clinical departments and physician locations where it can route calls and will be bringing another 17 online by fall 2006.

mPhase Signs Distribution Agreement with LEA for IPTV Solution

mPhase Technologies, (news - alert) which developed with Bell Labs a system to provide video over fixed broadband networks combining IP addressing with arrier-grade management features, has now partnered with Paris-based LEA S.A.S. LEA, which provides components and subsystems to telecommunications equipment and service providers, will distribute the mPhase TV+ System, taking on marketing and sales support for mPhase’s flagship system to its traditional accounts mPhase TV+ delivers broadband data and high quality video over telephone networks’ both copper DSL and fiber loops allowing telephone companies to compete vigorously in broadband markets. The mPhase System is comprehensive, supporting industry leading hardware products to ensure end-to-end quality for the consumer and profitability for telephone companies The mPhase TV + system is built around an innovative cluster architecture and offers key advantages to phone companies, including: • Improved maintenance environment,with links to industry-standard operations support systems, including element management of IP traffic • Compatibility with industry standard services creation environments • Transport media independence over copper or fiber facilities • A high degree of reliability to protect against network failover • Unprecedented scalability, capable of growing to hundreds of thousands of users

VoiceLog Launches First Hosted/ASP Call Recording Solution

VoiceLog LLC has announced the formal launch of its VirtualLogger division. VirtualLogger is a call recording and quality monitoring solutions provider, using the hosted/application services provider model to deliver its services. In addition, VirtualLogger offers other agent optimization technologies, such as speech analytics and real-time dashboards on a similar “hosted” basis. Call recording and quality monitoring are well established disciplines in US call centers, with more than 70% reporting some kind of recording in place. Almost all of this is done via licensed software systems, which can involve costs per workstation of over $1,000 initially and annual maintenance charges of $200, plus internal costs for system administration and data storage. VirtualLogger brings the value of hosted solutions to the call recording and monitoring space, making it easier for both small businesses and enterprises to record, and speeding up the implementation process. It eliminates upfront capital costs and greatly reduces the cost of system administration and storage, all for one small monthly fee. In addition, VirtualLogger reduces both technological and business risk since clients can inexpensively upgrade to completely new technologies (e.g., traditional telephone to VoIP) and can reduce or eliminate the cost to match changes in business needs.

IBM and 3Com to Deliver IP Telephony Suite for SMBs

Technology giants IBM and 3Com Corporation announced their plans to offer an IP telephony suite for small and medium enterprises. Specifically, both companies will introduce the 3Com VCX suite of IP telephony solutions on IBM’s “allin-one” System i business computing solution. The new agreement between the companies will enable small to mid-sized businesses (SMBs) to integrate IP telephony capabilities with the System i, which already comes loaded with hardware, database software, storage and security. IBM and 3Com want to deliver an IP telephony suite that can run in a single Linux on POWER partition on the System i. This will allow enterprises to run business and telephony applications simultaneously, managed by the System I’s management tools. The new suite is a good option for enterprises with 100 to 2,000 users. 3Com delivers a native, standards-based IP telephony and application system using SIP. The suite offers telephony features and applications, including unified messaging, large scale audio conferencing, video conferencing, presence, contact centers and enterprise mobility.

Vodavi and Covad Team to Provide VoIP Phones for the Very Small Market

Vodavi Technology Inc. and Covad Communications (news - alert) have announced the two companies will provide VoIP endpoints designed for the Very Small Business (VSB) and Small Business (SB). The arrangement calls for Vodavi to provide the new phone models to Covad, a provider of integrated voice and data communications. The new 6800 series handsets will be manufactured by LG-Nortel Co. Ltd., Vodavi’s strategic partner and largest stakeholder. The phones will employ both SIP and MGCP standards, and will operate seamlessly with the Covad vPBX, its PBX designed for companies with fewer than 250 employees. What’s more, despite being targeted for the small business, these phones will provide the same kinds of enhanced features as units designed for enterprise users, like “find me/follow me,” call logs, and contact lists. Prior to Covad’s introduction of the 6800 Series phones, small businesses lacked a variety of cost-effective alternatives. Today, VSBs and SBs can implement the productivity advantages of VoIP with minimal up front costs, making total telecom savings even greater. The phones will fit in with Covad’s all-in-one VoIP solution for smaller businesses.

BCE Elix to Distribute Mercom Call Center Call Products

Mercom Systems, Inc., a privately held company based in Lyndhurst, New Jersey and BCE Elix, a Bell Canada company focused exclusively on contact center solutions, announced that the two companies have entered into a distribution agreement whereby BCE Elix will add Mercom Audiolog Call Recording Servers and Mercom Interaction Quality (MIQ) Agent Evaluation software to its solutions portfolio. This new partnership will make Mercom’s suite of open architecture recording solutions and quality monitoring tools available to BCE Elix’s mid-market customers. “We are pleased to be working with Mercom,” said Daniel Rouleau, General Manager, Pre-Sales and Solutions Portfolio, at BCE Elix. “We are particularly pleased that Mercom has developed solutions that integrate with our portfolio, which is carefully selected to bring the very best in the market to our customers.” “The integration with the solutions that BCE Elix is currently selling is complete, and Mercom is delighted that our products are now part of the BCE Elix portfolio,” said David Zibelman, Vice President Business Development for Mercom. “The timing for this partnership is ideal since both organizations acknowledge the need to offer reliable TDM and VoIP recording solutions that provide call centers with cost-effective management tools.”

GL Communications 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, GL’s VQT Solutions can also support standard PSTN, VoIP and T1 E1 networks.

Fluke Networks 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 network’s 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.

Integrated Research 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-dialtone 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 institution’s 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 technolo gies 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.

Minacom 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 Minacom’s 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 — eplicating 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.

Ixia IxVoice/IxChariot

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.

Ameritec Corporation Fortissimo NLG-IP

The Fortissimo NLG-IP is a SIP network load generator that simulates high volumes of SIP sub scribers 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.

NetIQ 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 NetIQ’s 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 VoIPsecurity 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.

Net Latency SwitchMonitor

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 network’s 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 network’s 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.

Agilent Technologies 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 Analyzer’s 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 PreVoIP Networks; and • Test voice quality for VoIP end users — VQT emulates an IP phone.

Network General 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.

Snom IP PBX solution

With the snom VoIP Box IP PBX solution there’s no more need for expensive hardware components, such as gateways or servers — snom is breaking new ground with the VoIP Box.The snom VoIP Box is a 9x8x4cm small complete IP PBX solution that’s easy to attach to your LAN. It doesn’t contain any moveable parts, like hard disks or ventilators, so it’s durable and extremely low maintenance. It’s also completely silent.The snom VoIP Box can be completely configured through Web interfaces, by the customer or one of our partners. The configured software sits fully on the VoIP Box’s built-in Compact Flash (CF). Then simply connect the snom VoIP Box to the local LAN, using an Ethernet cable, plug in the Compact Flash, and switch on the power. After less than two minutes lyour whole telephone system is up and running — with all the performance features of a telecom munications private branch exchange which, in addition, can be extended over Web interfaces.

Arca Technologies emutel Harmony

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 Harmony’s 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 interControllers. 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 cus tomer 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.

Network Instruments Observer 11

Network Instruments, LLC, (news alert) 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.

Psytechnics PSI

PSI (Psytechnics Speech IP Monitor) (news - alert) 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.

Qovia Monitoring and Management Solutions

Qovia, 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. Qovia’s 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

Opticom OPERA

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 today’s 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 avail able 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.

Spirent Communications Abacus 5000 ICG3

Spirent’s 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 subsystem’s IP Telephony capabilities, combined with the PCG3’s and TCG3’s TDM, XCG3’s and ECG3’s analog features, provide a truly integrated VoIP/TDM/Analog platform to test converged IP Telephony Networks.

RADCOM Omni-Q

RADCOM’s 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. RADCOM’s 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 carri ers, service providers, manufacturers, and corporations. Many leading manufacturers have called upon Scientific NetIP Communications to represent them as their manufacturer’s 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 customer’s competitive edge.

Shunra Shunra VE

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.

Trilithic 860 DSPi

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.

Touchstone Technologies 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

SyncVoice VXTracker

VXTracker is based on a completely Web-based Enterprise Java Application server. Once engaged, the VX integrates with a company’s PBXs, keyset systems, routers, and VoIP gateways to monitor performance and provide clean understandable reports. VXTracker is made from Sun Microsystem’s 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 WVR7100/WVR6100

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.

Telchemy VQmon/EP

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 (6-30kbytes) 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 exten sions to improve accuracy under time varying network conditions, wideband codecs, orthogonal impairments and signal related parameters.

Miercom

Miercom is a privately held network consultancy, specializing in networking and communications-related product testing and analysis. The firm’s 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, equip ment recommendation and selection, cabling/fiber planning and design, installation of high-end equipment, and assistance with customized strategic network planning and direction.

CT Labs

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.

Viola Networks 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. NetAlly’s extensive and integrated capabilities include: • Active testing that enables predeployment 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 m onitoring 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 minuteVoIP 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

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 Omni’s centralized con sole, 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 remoteoffices • 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 vailability 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

Telecom New Zealand International Selects Veraz’s VoIP Solution for U.S. and U.K. Deployments

Veraz Networks, Inc., a leading global provider of VoIP softswitch solutions, announced that Telecom New Zealand International selected Veraz’s VoIP softswitch to be the preferred solution for TNZI’s telecom switches. Initially, TNZI will deploy Veraz’s ControlSwitch softswitch, service delivery platform and I-Gate 4000 family of media gateways in key U.S. and U.K. cities. This initial deployment will be a phased rollout that will be completed in early 2006 and will include installations in Los Angeles, New York, Miami, and London. “The strength of Veraz’s offering is its open and flexible architecture and, in particular, the flexibility of their system’s voice traffic routing that can adapt to TNZI’s business agreements now and in the future,” said Anthony Briscoe, General Manager of Telecom New Zealand International. “Veraz’s open architecture paves the way for TNZI to build new business models using best of breed application servers where appropriate, and fits with TNZI’s strategy of building its business on voice application wholesale rather than transactional minutes based revenue.” “TNZI wanted a new switching and operational model in order to scale its business and meet future demands,” said Doug Sabella, CEO of Veraz Networks. “This is a great win for Veraz in a major market and demonstrates the strength and flexibility of our solutions as well as our ability to deliver what is required in the market of today and tomorrow.”

Vonage Selects Sonus Networks Architecture to Support Expansion

Vonage is supporting its continuously growing subscriber base through a newly announced agreement with Sonus Networks. (news - alert) The Sonus architecture, including the GSX9000 Open Services Switch, PSX Call Routing Server, SGX Signaling Gateway and Sonus Insight Management system, is being rolled out in Los Angeles and New York. Vonage plans to expand their network to support additional markets in the United States, and eventually broaden to international locations. The Sonus solution enables Vonage to easily add new services while reliably increasing their subscriber base. Vonage has already deployed more than one million lines throughout its network. The Sonus IMS-based Voice over Broadband suite of solutions, improves operators’ abilities to easily make changes to their service offerings, and has already been deployed in broadband VoIP networks throughout the world. “Voice over broadband is a technology that’s redefining how service providers and consumers alike think about communications,” said Hassan Ahmed, chairman and CEO, Sonus Networks. “From day one, Vonage has been one of the most successful service providers in the industry. By selecting Sonus Networks, Vonage will have the ability to build out a premium network infrastructure to support the rapidly increasing demand for its services.” “Vonage has been a VoIP pioneer in North America,” said Michael Tribolet, executive vice president of operations for Vonage Network. “Throughout Vonage’s history we have remained committed to providing high-quality telephony service. By selecting Sonus, another pioneer in the VoBB market, we are in a position to continue to do this, while rapidly and successfully scaling the size of our network.”

Penn State Selects Qovia for VoIP Management

The education market — college campuses, in particular — have fully adopted IP telephony systems for their powerful, money-saving features that offer an alternative to traditional phone systems. The size of the systems, however, poses a challenge for providers to maintain both call quality and system security. According to David Woodall, Qovia CEO, the institutions in the education market are among the most advanced in their acceptance of VoIP. However, because they are on the cutting edge, they are also the first to feel the full effects of the limitations of the systems. Deep insight into their systems is needed to ensure that voice quality remains high, while the systems remain secure. Penn State has invested more than five years in research into VoIP as it started the process in 1998. In 2002, Cisco systems were deployed in earnest. Home now to approximately 6,000 VoIP phones, the University Park campus continues to increase its VoIP integration. Security for Penn State has been a priority from the onset. The university chose to lock down handsets to specific ports instead of allowing the portability of the handset, negating one of the key features and benefits of VoIP. Only predefined IP telephones are attached to the network, according to Phil Coolick, network services manager in the Information Technology Services department at Penn State. The MAC address of the telephone is locked to the switch port, protecting the core equipment from problems such as viruses by eliminating the possibility of a rogue device, such as a laptop, being used in a VoIP port. The bottom line is that Penn State will realize not only benefits from the use of the VoIP, it will also realize significant cost savings; that, in and of itself, is a major selling point for a public university.

Rangers and Stars Pick XO for Data and Voice

Telecom service provider XO Communications Inc. announced it scored a three-year agreement to provide voice and data services to the Texas Rangers and Dallas Stars. The agreement calls for XO to deploy communications systems for the Rangers at the Ameriquest Field in Arlington, Texas and for the Stars at the Frisco headquarters and at three Dr Pepper StarCenters located across the Metroplex. With the communications system in place, XO will be able to integrate all four locations. The deployment will increase the Rangers’ voice capacity by 25 percent at Ameriquest Field, while doubling available data bandwidth. Stars headquarters will be experience a similar increase in data bandwidth as well. The agreement also asks for XO to install its XOptions Flex VoIP system at the Dr Pepper StarCenters. Because Flex works with the existing communications infrastructure, there will not be a need to invest in new VoIP equipment to benefit from Flex’s dynamic bandwidth allocation across a shared voice and data connection, producing a minimum of three times the currently available bandwidth at those locations. “Businesses throughout the Metroplex want four things from their communications provider, product, performance, predictability and price,” said James McLendon, XO Communications’ general manager for the Dallas market. “We look forward to delivering all four to the Texas Rangers and Dallas Stars for the next three years, and beyond.”

Avaya and Polycom Team to Offer Complete IP Video Telephony Solution

Placing a phone call from your office in New York to a colleague in San Diego is one thing. Doing the same via IP is another. But being able to see your colleague by simply placing a phone call takes business communication and productivity to an entirely new level and helps to speed up decision-making processes. Avaya and Polycom have teamed to offer a complete business communications solution that includes desktop video conferencing, group video conferencing, and multipoint video conferencing. Avaya IP Video Telephony Solution integrates Avaya Communication Manager software for IP telephony with desktop video conferencing, group video conferencing, and multipoint network solutions from Polycom. “With single number dialing for video and voice, and call control functions that make video easier to use and content easier to share, the Avaya - Polycom solution makes video an integral part of a communications applications suite,” said Micky Tsui, Avaya vice president and general manager, Communications Systems Division. The Avaya/Polycom collaborative solution provides a single, easy-to-use, easy-to-deploy voice/video infrastructure. Enterprises will benefit from simplified integration of video across locations, lowers cost of ownership, and scalability — the solution is based on established industry standards for communications, including SIP H.323, H.320 and 802.11, and is able to accommodate more than 750,000 IP video stations on a single network. Naturally, network administrators also will benefit from having a single, unified communications environment to manage. “Avaya is a longstanding Polycom strategic partner in video telephony and this is a critical milestone in our joint development of IP-based, next generation, video telephony conferencing and collaboration solutions,” said Ed Ellett, senior vice president and general manager of Video Communications at Polycom. “Now the best video experience is available to our joint customers through Avaya’s familiar softphone interface. Video collaboration is now as simple as making a phone call.”

Tekelec, Convergin Enable MiRS’ IMS Future

Tekelec, a developer of telecommunications products for next-generation fixed, mobile, and packet networks, and Convergin, (news - alert) a provider of core fixed mobile convergence (FMC) solutions, have provided Israeli wireless provider MiRS with the Tekelec next-generation gateway mobile switching center (GMSC) server solution, the companies announced today. The solution will provide MiRS with next-generation capabilities, including FMC and hosted session initiation protocol (SIP) services, and will facilitate its transition to the Internet protocol multimedia subsystem (IMS) architecture. A major part of the solution is Convergin’s Accolade Wireless Convergence Server (WCS), which is designed to serve in the core of mobile and fixed networks. Tekelec’s IMS architecture drives switching for both MiRS’ entire mobile device line and SIP-based clients, and it supports the execution of common network applications. The solution also includes Tekelec’s next-generation gateway mobile switching center (GMSC) server solution and its 9000 Distributed Switching Solution (DSS). Tekelec will package the Accolade WCS as a product in the Tekelec wireless convergence gateway (WCG) family of products. “By enabling the convergence of SIP and cellular, MiRS is well positioned to capture additional lucrative markets as well as to strengthen its position in its existing business and enterprise markets,” said Dr. Ayal Itkovitz, CEO of Convergin.

PMC Telecom Prepares for Continued VoIP Market Expansion

PMC Telecom, already one of the leading suppliers of telephony equipment in the UK, is preparing for continued expansion of VoIP use in Britain, with new product ranges to cope with demand. VoIP — Voice over IP — is fast becoming a revolution in telephone use. With just a home computer and a phone that will connect to it, people can call anywhere in the world for less cost than using fixed landlines or mobile devices. According to Paul Conway, the Managing Director of PMC Telecom, “2005 has shown that VoIP isn’t going to be a passing phase. People realize that VoIP’s promise of low-cost internet phone calls isn’t a gimmick, it’s a reality. It’s a real consumer market, and one that’s expanding rapidly.” And VoIP isn’t simply just for the home user. There are real advantages for business with VoIP as well. “Businesses are waking up to the idea that they can cut down their telephony costs simply and easily,” said Paul. “Heavy users, such as call centers, are moving to VoIP use.” Whichever VoIP network supplier becomes dominant in the UK, PMC Telecom has moved itself into a prime position to supply all aspects of the domestic and business VoIP product market.

Primal Solutions and TMNG Partner to Tackle the Challenges of IP Transaction Management

Primal Solutions, the leading provider of IP transaction platforms, and The Management Network Group (TMNG), (news - alert) a leading provider of management consulting services to the global communications industry, announced a new strategic partnership. This relationship combines Primal Solutions’ IP Correlytics platform with TMNG’s Digital Services Assurance program to deliver a revenue assurance solution for the cable industry as it moves to sustain market leadership in an aggressively competitive marketplace. In this new environment, cable operators and other advanced communications and new media companies are offering new IP-based services and content. Increasingly, delivery of these new services necessitates complex relationships and delivery models that create revenue assurance challenges. The TMNG and Primal Solutions partnership addresses these challenges by offering an end-to-end program designed to maximize revenues, by linking all stages of product planning and deployment, to ensure successful launch, operations, and profitability. “We are very excited about this relationship because TMNG and Primal Solutions bring together the technology, people, processes and professional services required to address these complex and growing business challenges,” said Bob Richardson, executive vice president of operations at Primal Solutions, Inc. “After listening to our MSO clients, TMNG chose Primal Solutions because we believe its IP Correlytics platform is best in its class, and unique in its abilities to enable our clients to maintain the integrity of their revenue streams as they launch new voice, video, high speed data and content services,” said Bill Opet, principal and vice president of TMNG’s cable practice.

Whaleback Systems Releases SMB1500

Whaleback Systems has released the SMB1500, a groundbreaking business phone solution that operates over a broadband connection instead of a traditional phone line to maximize cost savings and deliver cutting-edge capabilities. Whaleback Systems reduces companies’ telecommunications expenses and eliminates fluctuating monthly costs with an affordable, all-inclusive, flat rate per station service package. The package includes unlimited calling throughout North America, an IP multi-line executive handset, voice mail, video calling, desktop messaging, remote access, caller ID, speed dial, call forwarding, direct inward dialing, local number portability, auto-attendant, first-class sound quality, e-911 coverage and U.S.-based technical support. In addition, Whaleback Systems provides a free server to support the network and free upgrades to prevent obsolescence. “Whaleback Systems developed a service package to ensure our cutting-edge technology is immediately affordable for businesses of all sizes,” shared Ken Stess, Whaleback Systems’ vice president of business development. “We integrated advanced technologies to engineer a system with easy to use features that improve staff productivity and streamline internal and external communications,” said Wray West, Whaleback Systems’ co-founder and vice president of engineering

ShoreTel Marries Converged Conferencing Solution with IP Telephony Solution

California-based ShoreTel is one of the fastest growing providers of IP telephony solutions in the United States. Not coincidentally, it also happens to be the industry leader in customer satisfaction. ShoreTel was the pioneering force first introduced Converged Conferencing to the IP telephony world back in 2003. Now, ShoreTel has introduced Converged Conferencing 5.6, the latest in its line of innovative conferencing and collaboration platforms, designed to work in conjunction with the ShoreTel 6 IP telephony system. Together, the two will provide integrated audio and Web conferencing, enterprise instant messaging, and document sharing. The solution, hosted locally, rather than as an Internet-based service, helps ShoreTel’s customers reduce costs, while providing a stable telephony infrastructure and improved employee productivity. “ShoreTel Converged Conferencing gives enterprises the freedom to abandon expensive conferencing services while enhancing collaboration and productivity across the enterprise,” said Steve Timmerman, vice president of marketing for ShoreTel. “By extending ShoreTel’s IP telephony system to include Converged Conferencing, we are addressing customers’ key business needs and delivering a compelling return on their investment.” With Converged Conferencing, ShoreTel provides a next-generation conferencing platform combining voice and data in a single, unified medium. Now, because ShoreTel Converged Conferencing is fully integrated with the ShoreTel’s phone system, it removes the cost barriers traditionally associated with in-house collaboration solutions. It also ends the reliance on and costs associated with external conference service providers.

PowerDsine Announces Next-Generation Power over Ethernet Midspan Series

PowerDsine announced the launch of its next generation 6500 PoE Midspan series. The new IEEE 802.3af-compliant 6500 PoE Midspan series is reportedly the first PoE midspan to offer advanced secured Network Management System and the only PoE Midspan to ship with a lifetime warranty. The PowerDsine 6500 PoE Midspans will be available in 6-, 12-, 24- and 48-port versions. The 6500 Midspans are designed for enterprises, which are deploying IP phones, wireless LAN access points or IP surveillance cameras. Midspans allow networks the added functionality of PoE without the need to replace existing Ethernet switches. This optimizes PoE port count, improves the ROI of networks and saves costs associated with these deployments. “The 6500 PoE Midspan Family provides enterprises a new level of security with this advanced remote management solution,” said Igal Rotem, PowerDsine’s CEO. “Moreover, we have such confidence in the 6500 Midspans’ reliability that we are offering customers a lifetime warranty.”

Zultys systems IP PBX Systems

The three Zultys systems — MX250, MX1200, and MX30— are fully integrated IP PBX systems based on open standards. Each system can autonomously provide full PBX functionality enhanced with realtime collaboration applications. Customers can cost-effectively deploy an enterprise class VoIP platform for 5 to 10,000 users. The MX30 is a powerful system that enables multimedia communications for small offices. By integrating the functions of many devices into a compactbox, the MX30 simplifies the VoIP network. It is a comprehensive solution that is easy to install, use, and maintain. The MX250 makes VoIP affordable for SMBs or branch offices. It scales from 5 to 250 users without additional hardware, combining the functionality of a PBX, voice mail server, voice gateway, and Internet gateway. Built with the same technology and the same productivity tools as the MX1200, it provides smaller sites with premium features at attractive prices. The MX1200 is a truly converged product that makes VoIP viable for the enterprise. This single 2U box brings together the functionality of a PBX, voice mail server, switch router, and Internet gateway. With a host of productivity applications, such as IM, presence, and PC voicemail management, the MX1200 can streamline communications for facilities with 25 to 1200 users.The system is 100% based on open standards, embedding technologies such as SIP, TAPI, VoiceXML, and Linux.

RTX Intros Hybrid VoIP Cordless Phone

Wireless product developer RTX America has introduced its PORTALphone, a hybrid, Web-enabled cordless phone that allows consumers to view customized Web-based content on a color screen. The new phone bundles VoIP-based calling with standard telephony and personalized info delivery in a household cordless telephone. The phone is optimized for Casabi’s platform to offer rich Web-based content services that have traditionally been tethered to a PC. Customers can now access news, local traffic and weather reports, stock portfolios, entertainment, personal address books, buddy lists and instant messages (IM). The RTX PORTALphone allows customers to make and receive calls anywhere in and around the house. The included PORTALphone base station simultaneously connects to a traditional telephone socket and a broadband Internet connection. The device supports SIP-based VoIP telephony as well as standard landline (POTS) telephony. “We believe that the PORTALphone gives carriers a unique opportunity to differentiate their VoIP services and add value beyond just voice,” said Curtis Schmidek, vice president of marketing at RTX America. “And customers get the best of both worlds. They can have personalized rich Web content and VoIP capabilities on a telephone that seamlessly integrates with their traditional telephone line.”

Apex Voice Communications Adds Multi-User Interface to Omnivox AES

APEX Voice Communications, a leading supplier of multi-service platforms for enhanced services and real-time billing solutions to telecommunications carriers, service providers, developers, and enterprises worldwide, announced that it has added a Multi-User Interface to its OmniVox AES multi-service platform for enhanced services solutions. Key features include enhanced account administration, new object control functions, and the greater use of permissions for various account actions. The new OmniView Operation, Administration, Management and Provisioning (OAM&P) module of OmniVox AES gives systems administrators the ability to create Domains, Groups and Users. For example, Domains are used to partition systems when services are being “sub-leased” to third-parties. Each Domain in return has its own domain administrator that can create User and Group accounts and assign them various permissions and object access capabilities. Groups are used to define a specific set of permissions that can be applied to Users. Also, password storage and verification now uses 160-bit encryption for increased security. “We have had extremely positive feedback from our customers who have started to implement the new multi-user interface of OmniVox AES,” said Elhum Vahdat, executive vice president of APEX Voice Communications. “The new multi-user interface, along with the added security features, continues to keep OmniVox AES at the forefront of multi-service platforms.”

Siemens Enters Router Arena with Acquisition of Harbour Networks Products

Siemens, the world’s third largest mobile telecom network equipment supplier, has reportedly acquired core assets of Harbour Networks Co., a broadband IP network equipment provider in China, for $110 million in cash. According to a news report, Siemens officially signed the purchase agreement with Harbour Networks and, as part of the deal, Siemens will acquire the Beijing-based company’s technologies and patents for high-end broadband products, as well as routers. Some industry experts have said the relationship between the two companies may become strained if Siemens starts to sell its own router products. The deal marks Siemens’ entry into the router arena. The company sells carrier Ethernet switches, but has relied on a reseller deal with Juniper Networks Inc.

Whaleback Systems IP PBX Solution

Whaleback Systems has leveraged cutting-edge technologies and the latest industry standards to engineer a software-driven PBX that transports voice signals over a broadband connection instead of a traditional phone line. The software-driven solution seamlessly integrates with your IT network to simplify system management, streamline communications, save money, and provide the advanced features your company needs, including desktop messaging, video calling and our exclusive Road Warrior Functionality. Best of all, Whaleback’s all-inclusive service package makes the most advanced technology immediately affordable for small and mid-sized businesses. The Whaleback SMB 1500 solution was built from the ground up specifically for broadband. It also is 100% premise-based and software-driven to simplify system management and make advanced features easy to use. There’s no equipment to purchase and upgrades are free so you do not have to worry about obsolescence. Whaleback’s exclusive Road Warrior Functionality feature connects customers with employees, regardless of their location, and gives mobile employees full access to the system, which streamlines communications and improves productivity.

Vodavi TeleniumIP system with Generation III software

Available for businesses seeking cost-effective mobile applications for remotely dispersed workforces ranging in size from 10 to 250 station users, the TeleniumIP system with Generation III software is based on distributed architecture that’s completely LAN-based with Web-based system administration and maintenance. TeleniumIP offers a full suite of user applications that improve communica tions and deliver mobility required to network multiple offices, home offices and remote buildings together for cohesive communications solution. Enhanced capabilities include the IP7000 Series of Desktop Telephones and Soft Phone IP Endpoints that improve mobility and deliver voice and video capability on a laptop PC or PDA. Enhanced calling features, integrated ACD and enhanced system administration features are included. A wireless IP (WiFi) handset that operates via wireless access points will be available in 2006.

Vertical's InstantOffice IP PBX Solution

InstantOffice delivers the features most in demand in an IP PBX, including voicemail, conferencing, auto-attendant, and integrated fax. Equally important, it accommodates the budget and staffing constraints of smaller organizations, allowing you to dra matically enhance the quality and efficiency of voice communication while keeping a tight lid on costs. InstantOffice allows your organization to consolidate voice and data communications on a unified platform that is easy to use, easy to manage and economical. The system provides VoIP capabilities, integration of voicemail, e-mail, and fax, and integrated networking — all delivered using a single, cost-effective T1 line. The InstantOffice PBX provides a comprehensive set of features for handling calls promptly, efficiently and flexibly. Easy-to-use functions such as call transfer, call waiting, off-site call forwarding and paging through speakerphones and overhead systems help ensure that calls reach the intended recipient, even if that individual is away from their desk or out of the office. InstantOffice supports a wide variety of analog and digital endpoint phones, giving you the flexibility to deploy the best and most cost-effective device for the purpose at hand, all using a single infrastructure. With an extensive set of customizable features, the system can be tailored to meet the unique and fast changing needs of your organization.

Vonexus Enterprise Interaction Center

Traditional PBX phone equipment was never made for VoIP and open standards like SIP. Enterprise Interaction Center, however, is a preintegrated IP PBX application suite and phone system designed for VoIP and SIP out of the box. More than that, EIC gives your enterprise everything it needs for “new-age” business communications. With EIC, customers get an integrated SIP-based media server that offloads the processing of call recording and other media operations, allowing enterprises to leverage SIP’s network scalability. They also get a bundled software suite for IP technologies that favor software over hardware, and that incorporates a ost-effective disaster recovery solution. Vonexus developed the EIC IP communications and phone system software exclusively for businesses using Microsoft’s product families. But EIC is truly designed for any company looking to lower costs and gain long-term investment protection by deploying VoIP and Microsoft-based standardsusing the SIP-architected EIC IP communications solution. And whether your enterprise is centralized in one location or looking to consolidate distributed branch offices, regional offices and remote and mobile workers on a single IP communications platform, the EIC solution is powerful enough to accommodate your entire organization.

TalkSwitch Phone System

TalkSwitch is a newbreed, all-in-one phone system designed specifically for businesses with 1 to 32 phone users per location. Its big-company features and intelligent capabilities work right out of the box and are a snap to customize. It integrates standard analog telephones, fax machines, cellular and offsite phones with the traditional telephone network and the Internet for VoIP. TalkSwitch delivers a world of connections, easy configurability and unprecedented prices. TalkSwitch comes with features that many other companies charge extra for and can be up and running in an hour. There also is no need for specialized telephones; TalkSwitch is compatible with standard analog telephones, so it works with existing phone sets — corded or cordless. TalkSwitch is a SIP-based hybrid phone system that can be connected to both the traditional telephone network and the Internet. With broadband access, and the optional VoIP module, businesses can place branch-to-branch calls over the Internet and access SIP based service provider networks. Built with an upgradeable, modular architecture, expansion with entails snapping in an expansion board or connecting additional units over the LAN. TalkSwitch does the rest.

Toshiba Strata CIX family review

The Strata CIX family includes modular, scalable, and networkable IP systems that provide sophisticated business communication features for businesses of all sizes. These systems deliver on the promise of IP telephony by providing all the features and benefits of our traditional business communications systems on a converged IP platform. Because Strata CIX systems support all types of telephones, they provide the configuration flexibility to build the communications system you need, in addition to the investment protection from re-using devices from other Strata systems. That’s why the Strata CIX is much more than just an IP system. It is a unified communications environment that supports all types of client devices. Toshiba’s Strata Media Application Server supports voice processing and all value-added applications integrated within one platform that connects to the Strata CIX via Ethernet.Applications include: • Auto Attendant • Voice Mail • Automated Speech Recognition (ASR) • Text-to-Speech • Unified Messaging • Interactive Voice Response (IVR) • Automatic Call Distribution (ACD) • ACD eporting • Toshiba-approved 3rd party CTI applications • Info Manager Web-based telephone applications • FeatureFlex adaptability tools • eManager browser-based system administration.

Tadiran’s Coral IPenabled servers

Tadiran’s Coral IPenabled servers allow you to add functionality and capacity as your business needs change and grow, helping to minimize your total infrastructure cost over time. Designed to provide unparalleled scalability, Coral’s voice and IP enabled features are completely compatible from the smallest system to the largest. Whatever the size of your business, there is a Coral system to deliver the bottom-line benefits you expect from advanced networking: lower networking costs, improved customer service, and increased employee productivity. • Coral6000 — High traffic duplicate processors for large businesses needing 6,000 ports and critical component redundancy; • Coral 5000 — Duplicate processors for medium to large businesses needing 5,000 ports and critical component redundancy; • Coral 400 — Full-featured communications in a mid-sized platform supporting up to 384 ports; • Coral 200 — Powerful, state-of the-art communications for the small business needing up to 200 ports; • Coral FlexSets — Digital telephones to complement your Coral system; • Coral QNet — Build a converged network with the QSIG global standard incorporated in your Coral system; • Coral FlexAttendant — Make every customer’s call a smooth, professional experience by delivering information directly to telephone attendant PCs.

Siemen's HiPath Series

HiPath is a modular portfolio of multifunctional communication and security solutions. It makes it possible to set up a modern real-time communications landscape and ensures that employees are always accessible and that corporate resources can be accessed from any location. HiPath solutions are based on open interfaces and compliance with common standards, so companies need only invest in those components they truly require. Existing systems can be migrated, new applications installed, and existing telephones adapted to current requirements through software updates. HiPath convergence solutions integrate the wireline and wireless worlds offering comprehensive options combined with an attractive return on investment and low Total Cost of Ownership HiPath applications integrate seamlessly into existing business applications and deliver the necessary presence and status information. HiPath offers secure access to all resources, accelerating communications and decision-making processes. The basis for a reliable and resilient communications environment, HiPath Real Time IP Systems transmits voice, data and video worldwide and over any type of network. This means service is never interrupted and productivity is enhanced. This goes for very small business networks, as well as huge global networks with 100,000 or more users: • HiPath 1100 and HiPath 1220 for small enterprises • HiPath 3000 for small to medium enterprises • HiPath 3000 and HiPath 5000 for medium to large enterprises • HiPath 4000 for medium to very large enterprises • HiPath 8000 for very large global enterprises

Switchvox IP PBX

Switchvox is everything that you don’t expect from a PBX. It’s truly affordable, easy to set up, simple to configure, and a breeze to maintain. It has features that let your business run more effectively and with fewer hassles. And it does all of this for a fraction of the cost of the PBX dinosaurs of the past. Switchvox is VoIP enabled, can use basic telephone equipment, and almost any sized business will find it’s a per fect fit. The Switchvox PBX interoperates seamlessly with traditional stan dards-based telephony systems and VoIP systems, so you can call anyone in the world regardless of their phone, PBX, or phone company. Setting up your Switchvox couldn’t be any easier when you use VoIP because no additional hardware is needed. You can set up your PBX to use VoIP in a matter of minutes. Your Switchvox PBX supports SIP, the technology choice for real-time communication session control throughout the Internet, corporate networks, and within next generation wireless networks.

SIP Foundry sipXpbx

The sipXpbx solution is an open source enterprise SIP PBX, complete with voice mail, auto-attendant, and a host of features. It is suited for large and small enterprise use, as it supports fully redundant call control (HA) as well as a SOAP-based Web services interface. sipXpbx offers plug and play Web-based management, including integrated management and configuration of attached phones and gateways. sipXpbx is a modular server based solution that runs on standard Linux. sipXpbx does not require any additional hardware as it interoperates with any SIP compliant gateway, phone or application. sipXpbx is a native SIP communications solution strictly following and implementing all the relevant SIP IETF standards. sipX is an IP PBX, but it’s much more than that — it is an ECS (Enterprise Communications Server). It handles voice and video, but its vision goes far beyond that. Real-time communications should be like email — a global interoperable system that allows for the exchange of real-time information of any kind (voice, video, IM, collaboration). A system based on presence throughout. A system based on SIP URI addresses that over time will replace PSTN phone numbers.

Sphericall IP PBX Solution

Sphericall IP PBX is a business communications application that runs on industry-standard servers across your existing data network with out expensive proprietary hardware. It is an open system solution that supports standards-based business telephony devices, gateways, and other communications endpoints, giving you the greatest flexibility and choice to create a customized enterprise communications solution to fit your unique needs. Through its unique distributed software architecture, Sphericall IP PBX scales to 30,000 ports and achieves 99.999% reliability with no single point of failure. Rather than simply clustering servers to achieve redundancy, Sphericall IP PBX is built upon a distributed software architecture that enables you to achieve 99.999% reliability in every location. Sphericall IP PBX offers Sphericall Web Services, providing a rich set of communications services for integration with other enterprise-class business applications. With Sphericall Web Services, the IP PBX becomes an enterprise softswitch that is utilized by various business applications to integrate communications into your business processes. The result is an advanced, proven enterprise communications platform that seamlessly integrates with your business. Now you can stop thinking about your phone system and start leveraging integrated communications to create value for your business. Sphericall IP PBX is the world’s most flexible, scalable, reliable and cost effective IP PBX.

Samsung’s OfficeServe 7200 communication system

Samsung’s OfficeServe 7200 communication system is a completely converged platform supporting both voice and data communication with powerful IP-based wired and wireless flexibility. The Samsung OfficeServe 7200 will support traditional voice communication, VoIP IP-based data, and wireless solutions through a wireless LAN Businesses can deploy OfficeServe 7200 to build sophisticated telephony applications, secure data-communications infrastructure, and policy-driven networks. OfficeServ 7200 with its comprehensive range of features and functionality, offers an effective, affordable solution for any organization. Whether you are a small office, a main office, or a branch of a larger organization with a need to take advantage of cutting-edge solutions, the OfficeServ 7200 can be at the heart of your com munications network. The OfficeServ 7200 includes a firewall, intrusion detection/prevention, policy manager and packet shaper. By deploying both a firewall and an intrusion detection/prevention system (IDS) the OfficeServ7200 is particularly strong on network security. The IDS not only monitors network traffic but reports attempted security breaches to the network manager by way of an alarm. The included technology ensures that voice will always be automatically prioritized. The OfficeServ 7200 also provides all of the telephony features and rich functionality necessary for mobility solutions and home working.

ShoreTel IP PBX Review

The ShoreTel system is a completely integrated IP phone system that scales seamlessly from one to 10,000 users including PBX, voicemail, and automated attendant functions. The ShoreTel system is built from the ground up to be easy to use and manage full-featured IP PBX system. Its distributed architecture is ideal for multi-site companies that span multiple locations because their phone system appears and behaves as one, unified system. ShoreTel 6 is the sixth generation of its distributed IP PBX voice solution, incorporating ShoreTel’s management and integration capabilities and integrated software distribution, media encryption, on-net dialing, and increased support of international operations. With native SIP support,ShoreTel 6 provides increased flexibility supporting connectivity to third party devices such as WiFi phones, conference room phones, and gateways. Office Anywhere extends the power of the ShoreTel system to remote workers without relying on the internet for voice quality. Remote users have all the power and productivity of all their Personal Call Manager delivered over their internet connection yet have the confidence of toll quality voice since the phone call uses the PSTN. ShoreTel 6 allows you to mix and match multi-vendor equipment with SIP support per the IETF’s RFC 3261.

Fusion Telecommunications launches efonica (R) VoIP Service

Fusion Telecommunications International, Inc. , a provider of advanced VoIP services worldwide has launched internet phone service called efonica. Users of this service will be able to call one another for free no matter where they are in the world using the efonica softphone, ip phone or their regular phones. All subscribers to the efonica internet phone service will be immediatelly after sign up assigned a eNumber with which other users can reach them on the efonica softphone (which can be downloaded for free on the efonica website) or their prefered hardware (ip phone, ATA or other SIP enabled device). eNumbers are telephone numbers starting with 10 the internet area code. With efonica it is possible to use your present telephone number as your eNumber so that your contacts do not need to remember a new number to reach you on efonica. For security the numbers are verified before they can be used as efonica eNumber. The efonica service is in three categories namely efonica free, efonica plus and efonica pro. With efonica free users can call other members for unlimited time free and also enjoy phone features such as Call hold, 3-way calling and Do not disturb. Efonica plus gives users all the benefits of efonica free and other interesting features such as efo out, web callgate, efo in, efonifax, efonicash and voicemail. Efo out enables users of efonica plus to call any regular phone in the world at the highly competitive efonica rates. Web callgate enable users to use the web to establish phone calls between two landline or mobile numbers in the world ( web activated telephony). Only efo out and web callgate are available now other features, the company says will be available soon. I have used this service and voice quality is really good thesame thing applies to efo out and web callgate. Efonica pro is business communications products, and services that can be customized to a companies communication needs enabling them to communicate more effectively and efficiently. My first impression about efonica is that its a more complete service and by the time all the features a fully available it will cater for the communication needs of individual users and businesses. More information about efonica can be obtained at www.efonica.com.

Voipbuster lowers rates again!

The calling rates for voipbuster users has gone down again. Finarea, the company behind voipbuster is proving to be the company with unbeatable low rates in the voip ecosystem by costantly lowering the rates for all their voip clients. Users of voipbuster can now call Albania, Iran, Malawi and Reunion for just 3 cents per minute. All calls to Aruba, Bahrein, Benin mobile, Botswana, El salvador, Guatemala, Jordan, Kenya, Kuwait, Mali, Mayotte, Namibie, Niger, NL antilles, Togo and Uganda now cost just 4 cents per minute.

Linksys Network Optimizer for Gaming & VoIP

The Linksys Network Optimizer for Gaming & VoIP from Linksys streamlines your home network for next-generation Internet applications. By applying various Quality of Service (QoS) techniques to your network traffic, the Network Optimizer makes sure time-sensitive applications like online gaming and VoIP calls run smoothly. The Network Optimizer sits between your home router and broadband cable or DSL modem, and monitors the data going through. It automatically determines what data is time-sensitive and what isn't, and prioritizes delivery of that data to its destination, ensuring the best-possible performance for the various types of data that go through your network. Traffic prioritization helps make sure that your online games don't lag when someone else on the network is surfing the web or sending email. Voice over IP phone calls sound cleaner. Outgoing streamed video from your Internet Video Camera runs smoother. And the best part is that the Network Optimizer takes care of prioritizing the data automatically, with no user setup of QoS policies required. After a simple installation it gets right to work, enhancing your network's performance and helping you get the most out of your high-speed Internet connection. Click Here for more information and to get this invaluable device.

Vbuzzer messenger 2.0 Beta launched

Softroute Corporation has released the beta version of Vbuzzer messenger 2.0 with new and interesting features. The latest version now includes text messaging, emoticons, the ability to personalize users profile with pictures and change "skin" to suit their mood!. Other built-in features in Vbuzzer 2.0 are call recording which allows a user to record a conversation anytime and for any duration and call forwarding which allows users to forward their phone number to any regular phone in the world. This is an exiting feature because it enables you to enjoy vbuzzer's low rates on regular phones. In addition to this launch Softroute also announced a PC-to-phone special offer of only 1 cent per minute to Australia, Austria,Belgium,Canada,China Mainland,Denmark,France,Germany,Hong Kong, Ireland, Italy, Netherlands, Norway, Singapore, Spain, Sweden, Switzerland, Taiwan, United Kingdom and USA. This speacial offer is only valid till june 30, 2006. Click HERE to upgrate your version of vbuzzer or to download the vbuzzer 2.0.
Related Post ============ Vbuzzer internet phone review

PBXpress Review

If your phone system is based on a legacy PBX using old style telephone lines, PBXpress can offer a range of affordable products that will enhance your corporate telephony infrastructure, while dramatically reducing overall capital and operating costs. PBXpress products are at the forefront of the next generation of VoIP-based PBX replacements that offer tremendous benefits by consolidating your organization’s data and telephony capabilities to a single network. PBXpress uses a reliable VoIP connection if it is available or it will fall back to a PSTN connection. In a PBXpress VoIP network each user is immediately recognized by handset or log on. Minimum investment is required to get started, since the PBXpress VoIP technology is embedded in your existing corporate data network. Subsequent telephony system expansion is as simple as plugging extra handsets into your existing LAN connections. PBXpress is software based and so upgrades can be per formed remotely via the internet. The wealth of enhanced telephony user features that arrive with PBXpress will increase employee productivity while the superb array of built-in call recording and archiving features will facilitate control of telephony traffic.

PBXNSIP Review

The pbxnsip PBX is the perfect product for the small to medium-size enterprise. It offers all popular features of the PBX and maximizes the interoperability with existing vendors of SIP equipment. It works with most ITSP vendors that offer SIP services. pbxnsip customers enjoy: • The look & feel of a traditional PBX, • Connectivity to an ITSP or to a customer premises PSTN gateway, • The ability to run infrastructure solely on IP, • Vendor independence, • Built-in security using the sips, srtp, and sdes standards, • Built-in SBC for solving NAT problems, • Call recording and paging functionality, • Trunking to gateways and service providers, The PBX offers two ways to connect to external devices. First, you may register SIP user agents (UA), for example hard or soft phones, FAX or other SIPcompliant devices like conferences systems. Second, you may connect to the outside world via trunks. The PBX supports several kinds of trunks. You may connect a regular SIP PSTN gateway to the PBX and terminate your calls to the PSTN through this gateway. The gateway does not need to support advanced features like transfer or dialog stage. See the list of PSTN gateways that we were able to test already.

Finarea launches WebCall Direct with free calls to 46 countries !

Introducing direct callback feature to all their voip clients seems not to be enough for Finarea and they have decided to launch a full featured web activated voip telephony nicknamed webcall direct. Its much similar to jajahweb the web activated telephony service by Jajah Inc. No download, headset or installation is required to use webcall direct and its very easy to use. After you register for an account, all you have to do is login, enter your telephone number and the destination telephone number and click call, wedcall direct will immediately dial your number and connect you to your destination number. Calls to landlines in 46 countries are free but limited to 300 minutes per week and a connection fee of 3 cents is charged per call. Calls to other destinations are at the Finarea very low rates as in voipcheap or voipbuster. Web activated voip telephony pioneered mostly by Jajah has now become so popular because of its easy of use and the fact that it does not require broadband internet to work. The only disadvantage of web activated voip telephony is the fact that you are charged for two calls, one to call your own phone and the other to call the destination you want to reach but this is not a problem for users of webcall direct who reside in one of the free countries as it costs nothing to call their number. Also if you live in one of the free countries and you are calling a country that is also free then the call is completely free. I have used webcall direct and the voice quality is really good like that of most voip clients from Finarea. You can try out webcall direct at www.webcalldirect.com.

Back from Holiday in Atlanta

For the past two weeks i have been on holiday in Atlanta, GA. It's a much needed break for me to cool off and re-energize. This is obviously the reason why i have not been updating my blog for some time now. I am now back to work and i have a lot of cathing up to do in order to bring my readers the latest in the world of voip. I am happy to be back blogging and i will soon give you guys all that have happened in the voip ecosystem an the past two weeks and will continue to provide the latest information as usual.

Zoom Telephonics X5V ADSL Gateway Router Overview

The ADSL X5v integrates a full-rate ADSL modem, router, firewall, four-port 10/100 Ethernet switch, USB port, and VoIP phone port into a single cost-effective product.The Model 5565 has a special Local Phone port that lets you use your standard single-line phone for conventional phone calls or free and very low-cost Voice Over IP (VoIP) calls. Because the X5v integrates Internet telephony with a full featured ADSL gateway, setup is easy and system security is ensured. Integration of the voice processing with the firewall and the router ensures reliability of VoIP calls and reduces service provider support requirements. The X5v also comes with the Global Village VoIP phone service that sets up in minutes and lets you simply dial phone numbers and make free calls to devices like the X5v anywhere in the world. And because the X5v uses SIP, the most popular protocol for voice communications over the Internet, you can tap into a rapidly-growing world of free and very low-cost phone services. The X5v's built-in four-port Ethernet switch eliminates the need for a separate Ethernet hub for most home networks and many small offices. The USB port allows direct connection of a fifth computer to the ADSL X5v. By adding external hubs or a wireless access point, the X5v can provide simultaneous Internet access for up to 253 computers.The Local Phone port of the Model 5565 includes an intelligent relay that allows a single phone to place and receive both VoIP calls and calls over the Public Switched Telephone Network (PSTN). PSTN Fail-Over allows the X5v to automatically route calls over the dial-up phone network when power is lost. PSTN support also allows emergency dialing using services like 911 or 112 calling.Setup is easy using any authorized local or remote computer's Web browser and the graphical user interface built into the X5v. Flash memory allows easy firmware updates, including additions to the voice compression codecs. The X5v cost about $80 and its available at Onsale

IOGEAR's USB Speaker Phone

IOGEAR's novel USB VoIP Speaker Phone is a high quality and portable USB device with DSP voice processing technologies. It is designed for Internet telephony applications such as Instant Messenger (e.g. AOL, MSN, and Yahoo) or Voice over IP Softphone (e.g. Skype, ,voipbuster, gizmoproject, Vonage, or Webex). The USB Speaker Phone is the ideal communication solution for any mobile worker or IM chatter to engage in hands-free conversation or conference calls. Features Full duplex speaker phone with 110dB peak volume Use with IM software (AOL, MSN, Yahoo, etc) Use with VoIP software (Skype, ,voipbuster, gizmoproject, Vonage,Webex, etc) Speaker: 2w peak, 40mm,4ohm, 500Hz to 5KHz 8KHz input sampling rate USB Audio 2.5mm headset jack Dimensions: 3.2 x 2.2 x 0.8 inches (8.1 x 5.5 x 2.0 cm) Weight: 1.5 ounces (42 grams) DSP voice Processor 40 mm mylar speaker and Electret Condenser omni-directional microphones.Click Here for more information about IOGEAR's USB VoIP Speaker Phone.

Wengo launches star destinations - Wengophone users can call all 26 star destinations free

The France based Wengo Limited, developer of wengophone has launched 26 free destinations tagged star destinations. Users of wengophone can call the star destinations free of charge but only users with at least 10 euros credit can make unlimited free calls to the star destinations while users without any credit can make unlimited number of free calls but the duration of the calls are limited to 2 minutes. The rates to other destinations are very low too. The star destinations are Belgium, Guadeloupe, India, Martinique, Poland, Vietnam,Argentina, Australia, Austria, Canada, China, France, Germany, Ireland, Israel, Italy, Luxembourg, Moscow, Netherlands, Portugal, Puerto Rico, Singapore, Spain, Sweden, UK, USA. Also a beta version of wengophone 2.0 has also been launched with more improved features for advanced users. Wengophone is an open source SIP based softphone and its features include, free PC-to-PC calls, free and cheap PC-to-PSTN calls, conference calling, IM and video calling. Want to know more about wengophone? check out http://www.wengo.com/

Transfering Contacts from Voipbuster to Voipcheapcom.

Anybody that have used voip clients from Finarea such as voipbuster, voipcheap, voipcheapcom etc must have been forced to change from one application to other more than once for several reason but mainly due to the fact that Finarea keep changing the free destinations and if one of them is the one you call frequently you simply change to another of their applications where that destination is still free. Finarea is always changing the free destinations as a way to make you pay another 10 euros minimum credit on another of their application where that destination is still free. One of the major problem one usually have due to this change is to transfer the contact list to the new application, before now you have to re-enter them one by one to the new application and if you have a lot of contacts this could be very time consuming. The good news is that the latest feature in almost all Finarea applications now has removed that limitation and make it easy to transfer contact list easily from one of Finarea voip application to the other. This new feature is Contact Archiving which makes it possible to download your contact list and safe in an archive file on your computer. This contact list can be restore easily from the archive file and most importantly the archive file from one application can be uploaded to another application thereby tranfering the contact list from one application to the other. For example to transfer your contact list from voipbuster to the new voipcheapcom, On voipbuster just click on tools on the menu bar, select Archive contacts and click on backup contacts to file, enter a name for the file and click save ( see the figure). To load this contact list in voipcheapcom you only need to click on tools, select Archive contacts, click restore contacts from file, select the previously safed file and click open. It so easy and it really saves you a lot of trouble of having to enter the contacts one by one.