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Quintum Intros BX Series Tenor VoIP MultiPath Switch
By Johanne Torres

VoIP technology provider Quintum Technologies announced the introduction of the Tenor BX Series MultiPath Switch. The company described the Tenor BX as a complete VoIP switching system for enterprises with ISDN Basic Rate Interface lines. Quintum explains that with the Tenor BX in place, enterprises will now benefit from a variety of valuable applications such as PBX extension, remote office connectivity, long distance consolidation and call centers. The BX is built on Quintum’s Tenor VoIP MultiPath switching platform.

The Tenor BX can support two, four, or eight BRI S/T lines, and a good VoIP CPE access option for service providers that are deploying services to customers that have PBXs that support ISDN BRI trunks.

Eatontown, NJ-based Quintum said the series increases the breadth of the Tenor product line allowing support of ISDN BRI installations. The company is sure that the addition of BRI provides the opportunity for both enterprises and service providers to deploy the Tenor VoIP access solution to a variety of offices that support analog, T1, E1, PRI and BRI interfaces.

“We have had a lot of demand for the ability to extend Tenor based network to locations that support BRI,” said Chuck Rutledge, vice president of Quintum Technologies. “With the addition of the Tenor BX product, Quintum further expands it ability to support a wide variety of PSTN connections, and it has all the advantages of the Tenor product line, most importantly, the ease of installation, and transparent deployment.”

“It is difficult to find any VoIP solution that supports ISDN BRI and the Tenor BX was the perfect solution for our customers needing VoIP deployment”, said Josef Bressner, president of Communiports AG. “We are able to support VoIP off of an existing ISDN BRI based PBX without any modification to the existing network, it deployed transparently.”
The Tenor BX is currently available starting at $1,600.

3Com Adds To IP Phone Software
By David Sims

3Com Corporation has announced a comprehensive enhancement to its Internet Protocol telephony software designed “to help small and medium-sized enterprises reduce costs, increase employee productivity and strengthen customer interactions,” according to company officials.

Key features in this new, fifth generation release of the 3Com NBX IP Telephony system, the industry’s first IP PBX (private branch exchange), include:

• New automatic call distribution option to allow businesses to easily expand their customer care service for up to 199 agents as part of their own in-house operations.
• Support for silent monitor/whisper/barge-in. When used with the ACD feature, supervisors can listen in on an incoming call, silently coach agents while on an active call and interrupt a call in progress if needed by barging into a call.
• Account code dialing and verification to allow professional service companies to more easily track and bill professional staff time.
• Support for the 3Com 3103 Manager large screen, Gigabit-speed phone with personal directory and call logs and support for the 3Com 3100 Entry, a single-line Power over Ethernet compliant IP phone designed to be a cost-effective alternative to traditional analog telephones.
• Support for two cordless IP phones for retail and small office applications.

With this release, 3Com offers a total of eight IP phones to complement the 3Com NBX system, including a competitively-priced Gigabit phone, the new 3Com 3103 Manager IP phone, which increases user productivity by managing up to 12 simultaneous calls.

The 3Com 3103 Manager Phone offers a long-term investment as the acceptance of “Gigabit to the desktop” becomes more widely accepted by enterprises. Pass-thru ports on 3Com IP phones also allow users to conserve LAN switching ports. For example, users can connect a personal computer to the 3Com IP phone, which is connected to the Ethernet port in the wall.

IP PBXs Up, Overall PBX Market Down

Worldwide PBX/KTS revenue dropped 12 percent to $1.5 billion between 4Q04 and 1Q05, but is two percent higher than a year ago. So says an Infonetics Research report. The number of PBX/KTS lines shipped worldwide decreased to 6.3 million in 1Q05, 15 percent lower than in 4Q04.

According to forecasts, the market will bounce back to $1.7 billion by 1Q06 as the market continues to move towards IP voice technology and benefit from an improving economy.
Pure IP is the only PBX category to beat the budget blues in 1Q05, increasing 10 percent to $223 million, 36 percent higher than a year ago. Revenue is forecast to jump another 24 percent to $277 million in 1Q06. Pure IP and hybrid system revenue together total $1 billion.

TDM system revenue sank 20 percent and hybrid system revenue dropped 13 percent in 1Q05. Most vendors recognize the shift in the market towards IP and have moved their products accordingly. The TDM segment, which still accounted for half the market at the end of 2003, is now less than a third.

Highlights of the report include the following:

• The top IP PBX system vendors for lines shipped are Alcatel, Nortel, Avaya, Cisco, and Mitel.
• 3Com had the highest sequential line shipment growth, up 26 percent in 1Q05.
• Hybrid PBXs account for 57percent of PBX/KTS revenue, TDM 28 percent, and pure IP 15 percent; by 2008, hybrids will increase their dominance to account for 67 percent of the market, pure IP will increase to 23percent, and TDM will decrease to nine percent.

SMC Expands Enterprise Wireless Solutions

SMC Networks recently announced the newest additions to its arsenal of solutions for extending enterprise connectivity with the flexibility of secure, robust wireless. Designed to expand the range, placement options, and throughput of the Enterprise wireless network for cross-campus and multi-story applications, SMC’s newest antenna and amplifier solutions help networks go the distance indoors and out, point-to-point and point-to-multipoint.

Compatibility being paramount for easy set-up and expansion and reliable connection SMC’s wireless solutions for the Enterprise comply with the 802.11g standard to provide speedy, long-range connections for users, whether via 802.11g or 802.11b adapters, or for end-to-end applications. Among the newest products for easy network expansion, the new EliteConnect 2.4GHz 500mW Power over Ethernet Amplifier (SMCAMP-500G) increases the signal strength and operating distance of any in SMC’s suite of 802.11b/g Enterprise Access Points, Bridges and High-Gain Antennas. And, Power over Ethernet support via SMCAMP-INJ means that placement restrictions due to lack of electrical power are lifted, as it takes its power from the CAT5 network cable.

AVST Launches CallXpress 7.7

Applied Voice & Speech Technologies, Inc. (AVST), has announced a new version of its flagship unified communications (UC) solution, CallXpress. Boasting new features and applications, CallXpress 7.7 is the ideal solution for companies looking for alternative options to outdated voice mail systems. The new version is enhanced with easy-to-use telephone user interface (TUI) options, increased unified messaging flexibility, and simplified administrative capabilities.
“AVST continues to drive the development of applications, features and enhancements with the mobile workforce in mind. Delivering on our commitment to enable communications anywhere/anytime, our solutions provide remote access to unified communications and collaborative applications through laptops, cell phones, PDAs and BlackBerrys,” said AVST’s Vice President of Product Management, Tom Minifie.
“IP technology is broadening the definition of the mobile workforce,” said Jay Lassman, research director, Gartner. “Innovative mobility enhancements to system architecture and user interface are necessary to simplify anytime, anywhere access to business communications infrastructure.”


Newport Unveils Carrier-Class SBC
By Johanne Torres

VoIP Session Border Controller developer Newport Networks announced release 2.0 of its 1460 session border controller software with enhanced redundancy and regulatory compliance in the U.S. market. Newport described the product as the industry’s only carrier-grade session border control system. The company also announced it expanded the organization with new offices in Frisco, Texas. “Service providers now require the same predictability, reliability and profitability as the PSTN for voice and multimedia over IP,” says Terry Matthews, Chairman of Newport Networks. “It’s time for vendors to step up and help providers succeed with new packet-based services. The US market is absolutely ready for a session border controller of this caliber. Newport is here to raise the bar.”

The company explained that where competing solutions address the challenges of completing calls behind corporate firewalls and Network Address Translation (NAT) devices, the 1460 adds architectural and cost-performance advantages. According to Newport, these bring the predictable quality and reliability expected of the PSTN to VoIP for the first time. The company says that as IP-based telephony and multimedia gain traction, the 1460 satisfies the stringent requirements of Tier 1 and Tier 2 service providers, re-defining “carrier-class” session border control.

The 1460 architecture offers the following features:

• Redundant Management Cards, ensuring manageability under card and network failure scenarios;
• Redundant power feeds supplying segregated power zones, ensuring that any power failure does not impact the ability to offer service;
• Automatic fail-over among cards;
• Link Aggregation (802.3-2002), allowing traffic to continue to flow even in the event of line failure.
• Support for up to over 120,000 concurrent calls
• Up to 1 million registered subscribers per node;
• Architectural flexibility allowing the product to be configured optimally to accommodate processor intensive applications such as SIP re-registration filtering;
• Architected for bandwidth-intensive multimedia applications such as video.
• Each call is policed to ensure that the bandwidth used matches the authorized bandwidth, thus preventing service theft and ensuring call-quality for other users,
• Session admission control: Allows service providers to manage QoS for each customer and therefore ensure that Service Level Agreements are met.

Newport explained that the chassis-based 1460 was designed for an extended life cycle in Tier 1 and Tier 2 carrier networks. The system can easily address new applications through software upgrades and the addition of new interface and processing cards. The 1460 allows advanced features, such as CALEA, to be implemented.

Release 2.0 of Newport Networks’ 1460 session border controller will ship in the North American and international markets through third quarter 2005.

VoIP Inc. Buys Caerus For $30 Million
By Johanne Torres

Its official: VoIP, Inc., and Orlando, FL-based Caerus, Inc., have merged. The companies announced that VoIP, Inc. has acquired Caerus, immediately initiating the process of merging operations. As part of the agreement, VoIP, Inc. acquired 100 percent of the business of Caerus, Inc., and its wholly owned subsidiaries Volo Communications, Inc., Caerus Networks, Inc., and Caerus Billing, Inc., in exchange for 16.9 million common shares of VoIP Inc. stock, a transaction valued at $30 million.

The deal will allow VoIP, Inc., to continue operations of Caerus, Inc., and its subsidiaries under their existing names, consolidating all operations under VoIP, Inc. After the merger, Steven Ivester will continue as CEO of VoIP, Inc. as well as all of the subsidiaries, and Shawn Lewis, Caerus founder and CEO will become CTO of VoIP, Inc.

"The synergies of VoIP, Inc. and Caerus are unmistakable and cannot be found anywhere else in the industry. We are creating a combined company with one of the broadest product offerings, deepest distribution channels, largest sales and service support teams, which just doubled in size in the industry not to mention one of the largest proprietary VoIP networks in the U.S. Its a combination that will benefit all of VoIP, Inc. and Caerus, Inc. customers and suppliers, VoIP, Inc.s CEO Steven Ivester said.

Combining Volos current customer traction, technology, and network with VoIP, Inc.s existing portfolio of companies is a huge step for the industry. Now there is a true, complete end-to-end solution for this ever-expanding VoIP industry. Nowhere else can you find a turnkey solution from end-point devices to fulfillment, and support services of such magnitude, said Shawn Lewis.

Were especially pleased that Shawn Lewis will continue to lead technology innovation and development efforts for VoIP, Inc., as CTO of the company, added Ivester. He is a well-respected leader and innovator, having written the patents for the first softswitch and SS7 Media Gateway. One of his many accomplishments was the sale in 1998 of XCOM Technologies, Inc., a CLEC that he co-founded, to Level 3 Communications, Inc. for common stock, options and warrants valued at $154 million.

Volo Communications, Inc. is the licensed facilities-based CLEC (Competitive Local Exchange Carrier) and IXC (Inter Exchange Carrier). Caerus Networks, Inc. is the technology research and development subsidiary, and Caerus Billing, Inc. is the billing and mediation subsidiary. Caerus, Inc. and its three subsidiaries claim to have generated revenues during calendar year 2004 that totaled $14 million, and based on current contracts and purchased orders, revenues are estimated to exceed $38 million for calendar year 2005.

VoX Intros Total VoIP System
By Johanne Torres

Packet communications services provider VoX Communications announced plans to introduce a ‘total VoIP system.’ Using the new system, the company, through a nationwide network, will be terminating traffic with Global Crossing VoIP Outbound Services and using the wireless IP phones and VoIP customer access devices from UTStarcom, Inc.
The company’s total VoIP system includes VoX’s wholly owned packet telephony technology and its nationwide network with Tier One interconnection partners like Global Crossing, who operates a global IP-based network and SIP-based VoIP platform.

VoX explained that it is also deploying UTStarcom’s wide range of VoIP customer premises equipment (CPE), including UTStarcom’s iAN-02EX VoIP analog terminal adapter (ATA) for residential customers and UTStarcom’s iAN-08E series VoIP gateway for SOHO/enterprise customers. The company recently added new CPE, UTStarcom’s F1000 portable WiFi handset.

VoX Communications offers wholesale broadband voice, origination, and termination services for cable, wireless, and wireline operators and VoIP service to the small business and residential marketplace. The company’s services feature Call Hold, Call Waiting, Caller ID, Call Transfer, Hunt Groups, Do Not Disturb, Call Forward, International Call Blocking, Call Return, Repeat Dialing/Redial, Extension Dialing, Anonymous Call Rejection and e-mail notification of voicemail, all at no additional charge. Clients are also able to choose additional features include: Multibox Voicemail, Music on Hold, Corporate Conference calling, Reassign Phone, Find me/Follow me, Auto Attendant, among others.

Arbinet Introduces Paid Voice Peering

Arbinet-thexchange, Inc., has announced a new service designed to allow voice over Internet protocol (VoIP) companies to get paid for telephone calls to their customers.

Local access charges, the fees for terminating a call to a customer, which total an estimated $14 billion annually in the United States alone, have been unavailable to a VoIP service provider until now because calls destined for a VoIP customer are typically sent through a local telephone company’s network and that company retains the termination charge. Two key puzzles needed to be solved to enable commercial settlement for inbound Internet phone calls. The first was a way to match a traditional phone number to the IP address of a VoIP customer, and the second was the ability to charge for routing the call directly to the customer over the Internet.

“There are companies today who provide free termination of calls between VoIP service providers, but, in our view, that will not be of interest to large telecom companies,” explains Curt Hockemeier, President and CEO of Arbinet. “Our new service commercializes this activity in such a way that we expect it will eliminate an expense and create a new revenue stream for VoIP service providers.”

Arbinet’s new commercial voice peering service provides a central telephone number to IP address mapping database and settlement system. An inbound call from an Arbinet buying Member will be checked against this database and sent directly to the selling Member’s VoIP equipment. The seller can charge a termination fee for receiving the call. Arbinet will collect and settle the transactions with buyers and sellers every 15 days.

Arbinet operates a voice exchange service, which processes approximately 1 billion minutes of traffic monthly for 375 telecommunications service providers worldwide. Arbinet’s new paid voice peering service will allow telephone calls to find their way through the Internet to the called party, bypassing the local, incumbent telephone companies.

Keynote Launches VoIP Quality Benchmark Survey

Keynote Systems has launched the first study to benchmark the service quality of the leading providers of VoIP phone services, as perceived by end users.

The study is designed to assess the market readiness of the leading Internet phone service providers by comparing the call quality of those VoIP providers in San Francisco and New York to traditional PSTN. The study compares the quality of Internet phone service providers based on 10 performance factors to accurately benchmark typical scenarios including placing coast-to-coast VoIP and local PSTN calls between New York to San Francisco and back as well as calls from a VoIP-enabled phone to a traditional phone.

Additionally, the study ranks leading providers on every major factor that affects the end-user experience with Internet phone service, including service availability, outage minutes, dropped calls, audio delay, listening quality, consistency, and geographic uniformity. The study also analyzes various diagnostic metrics that constitute the service performance factors and evaluates the impact of the underlying network carriers on the VoIP quality perceived by end-users.

The six leading VoIP providers included in the study are: AT&T CallVantage, Packet 8, Primus Lingo, Skype, Verizon, and Vonage. To understand the impact of underlying network performance on call quality, the VoIP telephone calls are carried on three T1 networks: AT&T, Sprint, and UUNET. In addition to measuring underlying network performance, the study captures the impact of the last-mile on call quality by measuring each of the six providers on residential DSL and residential cable lines as well.

Acterna Launches VoIP Test And Management Solution

Acterna recently announced comprehensive VoIP support features for its NetComplete Service Assurance solution. Driving cost-effectiveness, efficiency, and helping to generate revenue for service providers, NetComplete’s VoIP capabilities are designed to deliver end-to-end network coverage by enabling performance monitoring, turn-up test, and troubleshooting functions that meet the challenges of delivering successful, efficient, and reliable voice-over-IP service.
NetComplete’s VoIP capabilities are a key strategic focus of Acterna’s Service Assurance Solutions (SAS) division. Acterna recently launched SAS to provide “triple play” next generation network operators with a robust, modular portfolio of interoperable applications for end-to-end quality of service management across their converged broadband IP infrastructure.

NetComplete enhances the efficiency and reliability of the service quality management processes by enabling VoIP service providers to obtain visibility of the end-customer voice experience throughout the entire lifecycle of a call. It allows service providers to perform critical service-assurance functions such as service monitoring, rapid fault isolation, and tier-two troubleshooting so they can effectively address the unique challenges of delivering carrier grade VoIP. In doing so, NetComplete helps providers identify, isolate, and overcome common quality of service (QoS) issues, such as poor call quality, dropped calls, and slow dial tone response.

As components of NetComplete, Acterna’s NetAnalyst Test Management and NetOptimize Performance Management applications work in conjunction with the QT family of distributed test heads. Together, they provide VoIP demand testing and proactive monitoring by means of active call generation or passive call monitoring.


Serverless Peer To Peer VoIP Meets WiFi
By Johanne Torres

Popular Telephony Inc., a serverless peer-to-peer technology and Peerio provider, made news recently when it announced that Clipcomm, a Korean manufacturer of VoIP solutions and supplier to Korea Telecom, licensed the Peerio middleware technology to enable end-to-end serverless operability in a range of WiFi and fixed-line IP terminals.

“Clipcomm’s WiFi and Bluetooth technology coupled with Popular Telephony’s serverless Peerio middleware creates a fully featured SIP- or H.323-based PBX system for enterprises that requires neither servers nor wires,” said Dmitry Goroshevsky, CEO and founder of Popular Telephony. “As a result of this new partnership Clipcomm is able to bring a new approach to wireless telephony technology to the North American, European and Asian markets.

Popular Telephony explained that by embedding Peerio into the Clipcomm terminals Ethernet cables and switching hardware become a thing of the past, as the user just plugs in the terminals and the power cord to deploy a Peerio enabled Clipcomm telephony system. Clipcomm’s built-in WiFi and Bluetooth capabilities provide the interconnection between fixed and mobile phones.

The company claims that security and safety have also been designed into the system from conception, allowing gateways to drive phones using the power from the PSTN in case of catastrophic power failure in the office. Clipcomm phones also support a specialized high-performance DSP technology, as well as a range of different encryption systems including DH key exchange and various DES options.

¨We think this relationship between Clipcomm and Peerio will be a great opportunity for the market in the near future, and we are sure that this combination of the technologies will bring tremendous success to both Clipcomm and Peerio.” says Randy Ahn, Sales Director at Clipcomm.

Peerio core technology eliminates the need for any centralized server and allows any IP phone, PC or other terminal to interconnect and materialize into a bespoke communications system that is self-servicing, self-healing, redundant, and secure. A Peerio intelligent device or system can support a wide range of telephony features and services by delivering up to 250 features and scaling to over 4 billion lines.

Nextel Offers WiFi Service For Business Travelers
By Robert Liu

In a recent announcement, Nextel said it has partnered with Boingo Wireless and Wayport to offer the Nextel WiFi HotSpot service at more than 7,000 airports, hotels, convention centers, retail stores and other hotspot locations throughout North America.
The new service addresses “key concerns of individual customers as well as IT managers,” said Greg Santoro, vice president, products and services, Nextel.

The service is enabled by Nextel’s Connection Manager software client which sits in the user’s PC. Unlimited service is available for $39.99 per month and the first month is free.

But if the customer moves out of the Boingo or Wayport’s service areas, Nextel Connection Manager can automatically connect users to Nextel’s data network via its new im240 PC card for an additional fee. However, because that service uses Nextel’s pre-existing data network, the access comes at a rate of only 24-40 kbps and, bundled with the WiFi service, costs $54.99 per month.


Telchemy Intros Wireless And VoWiFi Handset Monitor
By Johanne Torres

Telchemy, Inc., a provider of VoIP Fault and performance management technology, announced it introduced what the company says is the first performance management technology suitable for direct integration into cellular and wireless handsets. The VQmon Release 2.2 supports the measurement of performance parameters for Push-to-Talk and other near-real time applications, adds support for key 3G cellular vocoders such as AMR narrowband and wideband, SMV and EVRC, providing a wide range of performance metrics.
Telchemy explained that it designed the VQmon/EP 2.2 for direct integration into IP based mobile and wireless handsets, IP phones, media gateways and residential gateways. VQmon/SA 2.2 provides cellular aware midstream monitoring for probes and analyzers and can be integrated directly into routers and session/ border controllers.

VQmon/4G bundles in the company’s new non-intrusive video quality monitoring technology to provide a single solution for IP based Voice and Video solutions. The company said the system is ideal for monitoring multimedia cellular handsets. Telchemy says the technology will help wireless service providers to deliver higher-quality advanced video applications including gaming, streaming video, and podcasting.

“Mobile and IP telephony are converging, which is opening up opportunities for service providers and enterprises to deliver multimedia over wireless handsets,” said Alan Clark, CEO of Telchemy. “VQmon Releases 2.2 and 4G meets the growing need for improved service quality to the mobile marketplace.”

The company says that network operators and system administrators currently use the VQmon technology to detect, monitor and resolve call quality and network related problems for networked multi-media services, including Voice and Video over IP, IP Centrex, 3G Cellular, Voice over WLAN, and streaming audio/video. The system also provides listening and conversational call quality metrics in both R factor and MOS formats as well as detailed diagnostic information, giving network managers both high level metrics and the ability to drill down to identify specific problems.

Telchemy is making available the VQmon/EP Release 2.2 and VQmon/SA Release 2.2 in June 2005, VQmon 4G will be available in August 2005.

PT Intros Higher-Capacity SEGway Signaling Gateway
By Johanne Torres

Performance Technologies (PT), a developer of integrated systems, platforms, components and software, announced today it introduced the SEGway 4300 Signaling Gateway. The company this is its most robust and highest capacity signaling gateway system to date.

The company explained that as global SS7 traffic continues to increase, carriers are seeking ways to increase network capacities. PT believes that in order to facilitate a gradual migration from legacy SS7 circuits to next-generation VoIP, carriers require devices capable of supporting both traditional SS7 protocols and IP protocols. The company says its gateway products such as the SEGway 4300 would provide the signaling bridge to allow traditional TDM and next-generation, IP-based SS7 networks to work together seamlessly.

Performance Technologies’ signaling gateways use the latest Internet Engineering Task Force (IETF) SIGTRAN standards to ensure reliability for these demanding VoIP applications. Scalable to 96 signaling links, the SEGway 4300 ensures OEMs and carriers a solution that can expand to meet the demands of high-growth, SS7 circuit-based or IP networks.

“The SEGway 4300 is our highest capacity signaling gateway offering to date. Our field-hardened signaling software combined with our robust hardware platform offers the flourishing VoIP market a flexible, cost-competitive solution for next-generation networks,” said Deb Brunner-Walker, signaling product manager for Performance Technologies. “As global IP convergence moves closer to the mainstream, our gateways are being deployed around the globe. This latest offering was created to meet the burgeoning demands of ever-growing IP networks and new application providers.”

The SEGway 4300 is offered in a 7U CompactPCI form factor. It offers 1024 virtual point codes and supports both circuit-switched and IP-based SS7 protocols. Beta trials for the SEGway 4300 are scheduled for July with general availability scheduled for September 2005.

TI’s Digital Media Processors Enable High-Def Conferencing

At a recent industry trade show, Texas Instruments demonstrated full-feature, high-definition multipoint conferencing based on the TMS320DM642 DSP-based digital media processor.
According to Codian CEO David Holloway, “High-Definition videoconferencing is a reality. As the HD resolution of 1280x720 at 30 frames per second becomes the new standard for videoconferencing, we will use our investment in the DM642 architecture to quickly and cost-effectively deliver HD quality multipoint video, recording and streaming to our customers.”

TI’s DM642 digital media processor is designed to provide developers with an optimized platform for enterprise-class videoconferencing applications that scale easily from client endpoint devices to the multipoint control units/gateways using multiple DM642 devices that enable IP-based videoconferencing across several sites. Multiple DM642s can be seamlessly connected via the 66 MHz PCI bus interface for high-speed connectivity. The DM642 digital media processors have numerous integrated I/O and streaming capabilities — including on-chip HD-capable video ports, glueless 10/100 Ethernet connectivity, multi-channel audio, and 66 MHz PCI connectivity — to develop true high-definition systems.

TI officials maintain that they are committed to the high-definition videoconferencing market and will continue to provide the hardware and software support essential for driving new services and enhanced functionality. DM642 processors running at 720 MHz are shipping in volume production today, and TI plans future code-compatible devices that offer higher performance and additional system integration to continue to bring cost-effective and scalable high-definition systems to market.

Surf Announces SurfRider-812

Surf Communication Solutions, has announced the launch of SurfRider-812, a fully-integrated high-capacity PTMC DSP farm comprised of eight digital signal processors (DSPs). The SurfRider-812 provides simultaneous Triple-Play media-processing capabilities for developers of voice and video gateways, CTI applications, media servers, and a multitude of other voice, video, fax, and modem applications.

Based on Texas Instruments’ TMS320C6412 DSPs, the SurfRider-812 can process voice, video, and data (fax/modem) simultaneously, on the same DSP. It integrates with PCI, Compact PCI (cPCI) and AdvancedTCA (ATCA) carriers.
The SurfRider-812 can be used in a variety of products, including Voice and Video gateway applications connecting mobile, broadband IP, wireless (WiFi/WiMAX), and PSTN networks; CTI Applications, such as Voice, Video, and Fax mail, Interactive Voice/Video Response (IVR) servers, announcement servers, unified messaging servers, and call recording servers for Voice, Video, and Fax; Inter-Working Function (IWF); and Stand-Alone Termination Applications, such as RAS, Conferencing Servers, Voice quality monitoring, and non-intrusive interception and security applications.

AudioCodes Intros SDK For AdvancedTCA Platforms

AudioCodes has announced the availability of its TP-12610 Software Development Kit (SDK) for AdvancedTCA applications. This new product is a development system for application developers seeking a quick start for an ATCA compliant platform using AudioCodes technology.

The ATCA PICMG 3.x family of specifications targets the requirements for the next generation of carrier-grade communications equipment providers. The TP-12610/SDK is designed to enable short time-to-market for software developers requiring a VoIP building block compliant with ATCA standards. The new design kit features a high-capacity blade of 2,016 channels and is based on AudioCodes field proven API common to all AudioCodes boards. Customers can use the same API for future platforms and thus reduce their development cycle and time-to-market.

The TP-12610/SDK is based on an AudioCodes cPCI VoIP board (either the TP-6310 or the TP-1610) hosted in a cPCI to ATCA adaptor. The adaptor-based board enables GB Ethernet base and fabric interfaces. The platform contains a 5-slot / 4U chassis with an IPMI shelf manager, a fabric (PICMG 3.1) and base switch blade, and an optional application processor blade.

Alcatel, Intel Collaborate On AdvancedTCA

Alcatel and Intel Corporation announced that they have signed an agreement to improve time to market of Advanced Telecom Computing Architecture (or AdvancedTCA) platforms for mobile service providers. Both companies will further work together on the design and development of AdvancedTCA subsystem building blocks to increase standardization and component interoperability across the mobile telecommunications industry. The new Alcatel solutions resulting from this collaboration are expected to be available in early 2006, and use an Intel Pentium M processor-based AdvancedTCA single board computer.

The companies will engage in joint development activities in such areas as system architecture definition and board and solution design based on Intel processors, with the resulting products to be available across the industry. Alcatel will participate early in Intel’s standard product definition process, ensuring a tight match between requirements and customer usage models. In addition to this implementation effort, the two companies remain closely aligned in their participation with standards bodies defining the specifications for modular communications platforms, such as PICMG for AdvancedTCA, Service Availability Forum for high-availability middleware and OSDL for Carrier-Grade Linux.

Wind River Linux Available For Artesyn Blades

Artesyn Communication Products, recently announced the availability of Wind River Systems’ Platform for Network Equipment for Artesyn’s KatanaQp AdvancedTCA telecom blade.

Wind River’s Platform for Network Equipment combines Linux 2.6 and OSDL Carrier Grade Linux with Wind River’s Eclipse-based Workbench development suite and a rich set of networking middleware. The platform is designed to assist telecom OEMs to develop and deploy network infrastructure equipment based on Artesyn’s KatanaQp ATCA blade.

The KatanaQp is a high-performance ATCA telecom blade that combines two PowerPC MPC7447A processors with four PTMC expansion sites, IPMI-based system management, and a PICMG 3.1-compliant ATCA interface. The KatanaQp’s high-speed PowerPC processors, switched fabric ATCA interface, flexible mezzanine expansion, and integrated system management make it easy to configure for a wide variety of control and packet processing applications, including WAN access, SS7/SIGTRAN signaling, media gateways, traffic processing, wireless base stations and softswitches.

Wind River’s Platform for Network Equipment provides a complete implementation of Linux 2.6 with CGL 2 extensions. The pre-emptive kernel features high-resolution timers, fast user-space mutexes, and a native POSIX thread library. CGL v2 extensions include an Intelligent Platform Management Interface (IPMI), Hardware Platform Interface (HPI), heartbeat monitor, hot-plug, and Ethernet link aggregation failover. Networking support includes IPv4/IPv6, SNMP management, and a comprehensive suite of open source networking protocols and applications, including DHCP, FTP, HTTP, NFSv4, NTP, PPP, SCTP, Telnet, VLAN, SSL, SSH, and IPsec.

The KatanaQp, equipped with a single MPC7447A processor and a four-channel Gigabit Ethernet fabric interface, sells in OEM quantities starting at $3,498. Wind River’s Platform for Network Equipment is available directly from Wind River Systems.


UCS Solutions Implements FrontRange Contact Center
By Johanne Torres

UCS Solutions, an IT systems provider to South African businesses, announced that it implemented a contact center system provided by FrontRange that would enable the company to enjoy advances in the productivity and effectiveness of contact center staff. Thanks to the partnership between both companies, UCS Solutions and its 40-consultant contact center will now be able to provide support to about 15,000 users at more than 20 major customers.

Debby Webster, Customer Contact Center manager at UCS Solutions, explained that the company signed up as pilot implementers of FrontRange Contact Center in May 2004 and were fully aware of the challenges and risks associated with being the first HEAT customer globally to implement a new, mission critical application.

“We’re also in the IT business. We knew that being the first with a new application would have its challenges. However, we drew on the strength of our five-year partnership, experience of their HEAT application and their knowledge of contact center business to deliver a great solution,” Webster added.

It took UCS Solutions several months of testing and implementing the system. After this lapse of time, the company reported it was “now ready to respond to requests to visit the facility from people who are interested in seeing at firsthand how it works,” stated Webster. “Quality monitoring and training is most important in any contact center. FrontRange Contact Center gives us the ability to ensure that consultants take calls on specific clients only when they are properly trained to do so. This ensures a quality service to our clients.”

UCS Solutions is currently using the voice recording on demand functionality within FrontRange Contact Center, but plans to expand this to recording all calls soon.

UCS Solutions described the FrontRange Contact Center as a system that integrates with any other service and support application. The company found that the fast integration with the existing HEAT implementation a further benefit.
Future development for UCS Solutions could be to expand the use of FrontRange Contact Center to its Cape Town and Durban facilities, enabling the functionality of FrontRange Contact Center to be applied to the same queue of calls from any location. “Ultimately, we’ll be able to exploit the potential of the idea of the virtual contact center, where individual consultants are able to take calls from home, for example. We plan to be ready for the growth and possibilities of VoIP by having the technology and the systems in place to make VoIP work effectively,” explained Webster.

“It’s not often that a totally new development such as FrontRange Contact Center is implemented first in South Africa, and we’re especially pleased to have been able to extend and deepen our strong existing partnership with UCS Solutions during this implementation. Since this deal, FrontRange Solutions has closed a further 15 sites to date, thus vindicating the enormous potential we all see with this product,” said Tracey Newman, managing director of FrontRange Solutions South Africa.

“The FrontRange Contact Center provides comprehensive telephony functionality in a single modular package that allows organizations to simply plug directly from Telkom into a SIP (session initiated protocol) compliant voice gateway and from there to a single server. It eliminates points of failure and extensive maintenance, makes upgrades easy and slashes implementation time and cost,” concluded Newman.

Today’s news follows FrontRange’s recent extension of its global access of its new modular IT Service Management (ITSM) system with a new version and languages including English, German, Russian and Polish. The company announced this month that French and Chinese will be made available soon. The company said the release of ITSM 5.0.2 is designed to improve the performance of IT and support organizations for new customers as well as offer additional modules to increase functionality for HEAT customers.

Michael McCloskey, CEO of FrontRange stated: “We will continue to add significant new functionality and languages to our new offerings to extend user’s access as well as our global reach. These solutions, to be introduced over the course of the year, will offer our customers an exceptional opportunity to obtain enterprise-class functionality at an exceptional value.”

Nuasis Integrates IP-Based Contact Center With Siebel CRM
By David Sims

Nuasis Corporation, the IP contact center company, has announced that it has successfully integrated its software-only, IP-based contact center system with Siebel CRM. The Nuasis NuContact Center handles customer inquiries via the phone, e-mail, Web and fax. The integration of the Siebel CRM application with the software-only, IP-based Nuasis contact center system extends the CRM investment and should increase contact center efficiency and productivity.
When CRM applications are integrated with the NuContact Center, companies can “pop” existing customer information from the CRM database onto the contact center agent’s desktop, thereby providing the agent with detailed customer information so that the call/e-mail/Web chat can be handled faster and more efficiently.

Nuasis customers are reporting that the combination of the IP-based contact center integrated with their existing CRM applications allows them to shave 30–45 seconds from every customer contact.

“Our ability to quickly deploy integrations with enterprise applications such as Siebel is a result of our software-only, single network product architecture,” said Kevin McPartlan, vice president, product direction, Nuasis. “Our choice to use open standards such as VoIP, SIP, and SOAP rather than proprietary CTI protocols simplifies deployment and decreases implementation times from many weeks to days. The open standards, software-only, model dramatically lowers the cost of CRM integrations.”

Nuasis reports that nearly 100 percent of its customers are integrating the NuContact Center with CRM applications. For Nuasis customers, no costly hardware or middleware, nor CRM APIs are required. Furthermore, there is also no requirement for complex and costly professional services.

As a single, distributed system, the NuContact Center is designed to replace multiple legacy ACD systems. It intelligently routes and queues customer contacts across multiple geographically dispersed service centers taking full advantage of VoIP technology.

Siemens Announces Contact Center App Upgrades
By David Sims

Siemens Communications, Inc. has announced Versions 6.5 of its HiPath ProCenter Agile and ProCenter Standard contact center applications, marketed towards enterprise customers small to large and across myriad industry sectors in more than 60 countries.

The Siemens HiPath ProCenter V6.5 portfolio is designed to provide faster call center administration, call processing and multimedia routing and reporting, including enhanced user and management visualization tools.
Presence-driven and permission-based collaboration tools have also been extended to reach enterprise-wide communication sources such as the telephone, e-mail and instant messaging.

The upgraded HiPath ProCenter Agile solution, with support for as many as 150 active agents, is designed for small and mid-sized enterprise contact centers or informal call handling groups. Its new features include e-mail management, scheduled callbacks and agents in multiple groups. It also provides integrated design capabilities for a basic IVR, and integration with Microsoft CRM.

As a ready-to-run solution, Agile is engineered to be easy to implement, configure, and use, delivering intelligent call routing, graphical reporting, and productivity tools for both call handling agents and managers.

Streamlined for call handling and communication using a smaller footprint, the HiPath ProCenter Agile solution includes an associate desktop that can extend intelligent call routing features to employees who serve as overflow agents during peak traffic periods.

The upgraded HiPath ProCenter Standard solution, scaling up to 750 active users, includes enhancements such as full IVR support, multimedia skills-based routing and a software developer toolkit for vertical business process integrations.
As with the HiPath ProCenter Agile, presence tools such as Team List and Team Bar are integrated into the Standard’s application desktop, allowing agents to visually monitor the real-time availability of other enterprise agents, managers and subject matter experts. As needed, agents can engage colleagues with the solution’s one-click collaboration capabilities, whether they are in the next office or remotely connected to the enterprise via an IP network.


AltiGen To Partner With Commtech For Irish IP
By David Sims

AltiGen Communications, Inc., has announced it is partnering with Commtech, a distributor of Internet, networking and security products in Ireland, to distribute AltiGen’s award winning IP phone systems and call center tools to the Irish market.
The partnership is another step in AltiGen’s continuing strategic plans for growth in Europe.

“We see Commtech as a strategic distribution partner to support the unique needs of this emerging market,” said Mike Plumer, AltiGen’s Vice President of Sales. Commtech will be a specialist distributor of AltiGen technology and a key component in delivering complete VoIP and IP convergence solutions in the small and mid-sized enterprise market.

Commtech will also provide independent advisory and installation services to support the AltiGen reseller channel community.
Justin Owens, Managing Director of Commtech thinks AltiGen’s “superior product offering” is “well suited to the Irish market.”

Alcatel And Polycom Converge Voice, Video, And Data
By Johanne Torres

VoIP technology providers Alcatel and Polycom, Inc. announced that they have partnered to deliver converged collaboration systems to enterprises. The multi-year agreement will bring together Polycom’s video conferencing systems with Alcatel’s unified communications offering. The companies explained that they will now focus on delivering enterprise users integrated collaboration solutions that span voice, video, and Web conferencing.

With the new system in place, enterprises will now be able to instantly connect with people and workgroups through an instant messaging client or via the telephone to launch rich media collaboration sessions that include video. Users will also be able to escalate from one form of communication to another with just one touch, adding video to an Alcatel phone call or audio and video to an instant message.

“Alcatel is committed to delivering SIP-based collaboration solutions to enterprise users, enabling them to instantly initiate rich media sessions regardless of the device they are using and regardless of their location,” said Jean-Luc Fourniou, senior vice president, Alcatel enterprise solutions activities. “Partnering with Polycom supports our strategy to deliver video as part of our converged collaboration solutions, improving both individual and workgroup productivity for enterprises.”

Nortel And IBM Sign Deal To Broaden VoIP Offerings
By Johanne Torres

Nortel and IBM announced that the companies agreed on a deal designed to jointly support customized products across a range of market segments. As the first step in this technology, research and services relationship, the companies will establish a Nortel-IBM Joint Development Center based in Research Triangle Park, North Carolina, to collaborate on the design and development of new products and services.

At the Nortel-IBM Joint Development Center, personnel from both companies will work to accomplish the following goals:

• Work together to enhance and extend current products, drawing from various divisions within Nortel and IBM, to drive new revenue growth while reducing R&D costs
• Collaborate on focused research on a project-by-project basis, enabling a new level of product creation by tapping the deep skills of each firm to introduce solutions more rapidly
• Work together on technology, initially, a new class of blade servers that would meet the specific and demanding data flow, reliability and security needs required by the network equipment marketplace embracing next generation network solutions.


Ingate Intros Firewall 1180 And SIParator 18
By Johanne Torres

SIP security products provider Ingate Systems announced two new products: the Firewall 1180 and the Ingate SIParator 18. The company describes both as powerful tools that offer complete support for SIP-based IP communications, such as VoIP, to small businesses, branch offices, or home offices.

Ingate explained that both the Firewall 1180 and SIParator 18 have three ports and offer a 25 Mbit throughput. The company describes the products as small and versatile with no fan, making them virtually silent, therefore, eliminating the need for a separate server room.

The Firewall 1180 and SIParator 18 bring VoIP, IM, and a host of SIP-based applications to the small business, branch office or home worker, said Olle Westerberg, CEO, Ingate Systems. With these products, any business, regardless of size, or location, can enjoy the benefits of increased productivity and the cost-savings associated with IP applications such as Internet telephony.

Included with the 1180 are ten SIP user licenses that can be used for the registration of SIP user agents, such as phones and soft clients, on the SIP registrar. Five SIP traversal licenses also come standard, allowing up to five calls to traverse the Firewall at the same time. Additional SIP user licenses and SIP traversal licenses can be purchased at any time.

The Ingate Firewall 1180 offers control over SIP signaling, traffic, and network security. TLS encryption ensures privacy when communicating, making eavesdropping, call hijacking and call spoofing difficult.

The Ingate SIParator is a stand-alone device that connects to an existing network firewall to enable the traversal of SIP communications. Included with the SIParator 18 are ten SIP user licenses that can be used for the registration of SIP user agents, such as soft phones and soft clients, on the SIP registrar. Five SIP traversal licenses also come standard, allowing up to five calls to traverse the firewall attached to the SIParator at the same time. Additional SIP user licenses and SIP traversal licenses can be purchased at any time. The SIParator 18 can be configured as a part of the DMZ or in a standalone mode.

The Ingate Firewall 1180 and Ingate SIParator 18 are currently available for $900.

Genesys Announces SIP Contact Center System
By Johanne Torres

Genesys Telecommunications Laboratories, Inc., recently launched a new SIP-based contact center system. The system would provide entities with customer interaction control for any SIP-enabled infrastructure, regardless of vendor.

The company says it designed the Genesys SIP system to handle contact interaction control among SIP-enabled devices, such as gateways and end-points. The system provides agent state tracking and monitoring functions, and delivers a set of interaction management functions needed in a contact center, including customer segmentation, call queuing, call routing, reporting, and call control. Furthermore, the solution takes traditional end-to-end IP calls and mediates them as a central IP server. Genesys explained that it is integrated into the open, standards-based Genesys Customer Interaction Management Platform for managing and tracking customer contacts from beginning to end.

“Our new SIP contact center solution extends Genesys’ long-standing commitment toward platform-independence to the SIP world,” explained Elliot Danziger, chief technology officer, Genesys. “This allows us to incorporate interactions from any SIP-enabled component and centralize the interaction data for improved reporting and management, delivering the full value of IP contact centers to our customers.”

“From the beginning, Genesys has stood out in the contact center market by offering an open, software-based platform. By extending its open architecture into the IP contact center environment, enterprises are able to realize the significant cost and business benefits of IP, while utilizing a proven contact center solution,” said Seema Lall, industry analyst, Frost & Sullivan. “Genesys has always been focused on open contact center software and this latest release demonstrates the company’s commitment to be an industry leader in IP technology.”

Nokia 770 To Support SIP-complaint VoIP
By Robert Liu

Nokia’s new 770 Internet Tablet is designed with VoIP capabilities, thanks largely to the combined efforts of fellow countrymen of the Finnish handset maker.

The Nokia 770 Internet Tablet, which connects to the Internet via WiFi or using Bluetooth with a compatible mobile phone, is built upon the software applications platform, a license-free set of developer tools. Using this software environment, Helsinki-based Movial Corp. developed VoIP Connect — a third-party, SIP-compliant application designed to run on the 770’s Linux environment.

But even though Skype also offers a Linux-based download for the desktop, Movial’s CEO doesn’t view the popular peer-to-peer (P2P) application as a competitor to his own product because his revenue stream is based on OEM agreements with handset makers and operators.

“For the operators, it’s a lot better to rely on SIP than on peer-to-peer,” Jari Ala-Ruona, Movial’s CEO, explained.
According to Ala-Ruona, VoIP Connect supports both iterations of SIP: IETF and IMS. While the IETF’s model for SIP is more widely used as the industry standard, cellular operators have adopted an infrastructure that uses IMS in order to better control access and usage of their closed networks.

The PC client is available immediately. VoIP Connect for the Nokia 770 will be available this summer. The product was never announced because Movial didn’t want to pre-empt Nokia’s launch of the 770.

Additional features of the Nokia 770 Internet Tablet include an Internet Radio, RSS News reader, Image viewer, and Media players for selected types of media. The Nokia 770 Internet Tablet is planned to start shipping in the third quarter of 2005 in selected countries in the Americas and Europe.

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