More Popular Telephony news. On a related note, I'm moderating the P2P VoIP session tomorrow, which includes Popular Telephony, Skype, and others. Can't wait!
 

EWOOPHONE BECOMES FIRST VOIP SERVICE PROVIDER IN CHINA
TO MARKET VOIP P2P SERVICES USING PEERIO SERVERLESS TECHNOLOGY

Miami, FL and Sophia Antipolis, France, February 23 2005 - Popular Telephony, the telecommunications middleware company and creator of Peerio - the innovative technology for server-free, peer-to-peer communications, today at the Internet Telephony conference in Miami, FL announced that China-based Ewoophone will be the first VoIP service provider in Asia to offer PC-to-PC and PC-to-phone VoIP service using Peerio serverless technology.

Ewoophone customers in China will be able to place domestic and international VoIP calls via Peerio-enabled client software, IP phones and media gateways operated by Ewoophone.

Peerio is a groundbreaking patents pending core technology based upon middleware that implements a serverless voice and data communications system. Peerio eliminates the need for any centralized server and allows any IP phone, Personal Computer (PC), endpoint or other terminal to interconnect and materialize into a complete communications system that is self-servicing and self-healing, redundant and secure. The Peerio autonomous telephony architecture brings greater efficiency for communications as well as substantial cost-effectiveness, saving up to 80% of the system or application cost. A Peerio-intelligent device or system is capable of supporting the widest possible range of telephony features and services, delivering up to 450 features, and can seamlessly scale to over 4 billion lines simultaneously.

Ewoophone has chosen the Peerio serverless architecture as the foundation for their VoIP services, which will expand with serverless capabilities and new enhanced features. Enterprise customers and home users alike will use Ewoophone-branded Peerio-powered soft phones, IP phones and other communications devices to make low-cost VoIP peer-to-peer (P2P) calls. Ewoophone will also support the Peerio Global Numbering Plan (GNUP). GNUP enables interconnection between Ewoophone’s users in China with the global community of VoIP and Peerio users, as well as provides a termination point for PSTN calls routed to China over the Internet. 

GNUP works by assigning users with a unique identification number for seamless accessibility with any GNUP-enabled terminal or other network user through a GNUP partner service provider, while supporting full portability. A GNUP user can activate their number wherever they have Internet access and the GNUP software is installed. GNUP can easily be carried on a smartcard or other storage device or simply downloaded from the web site by users keying in their log on information. GNUP then detects their online presence and makes them accessible for interaction.

“We are very glade to partner with Popular Telephony to deliver real P2P VoIP services across the Asia region. This will help us greatly reduce the burden on our future expansion as Peerio is just the pie we are looking for,” said Wilson Shan, the co-founder and CEO of Ewoophone.

"We are pleased to be working with Ewoophone to pioneer the business of serverless telephony in one of the fastest growing regions for telecommunications," said Dmitry Goroshevsky, CEO of Popular Telephony. "Ewoophone is an important addition to our rapidly growing list of partners in Asia, and around the world, who are contributing to the expansion and utilization of serverless communications."

About Ewoophone Technology

Innovative and inspired, Ewoophone Technology (ET) is a leading edge VoIP service provider and strategic telecom partner offering telephony services and other leading value added Internet services. With experience in telecommunications operations and broadband Internet services Ewoophone has become a leading developed high-tech enterprise in China.

ET promotes VoIP applications and terminals to businesses, families and individuals, targeting those segments to use the Internet to the fullest of its capabilities.

http://www.ewoophone.com/ewoophone/index.jsp

Voxeo Corporation today announced its new line of standards-based turnkey communication servers today. The new VoiceCenter Fusion Server product consolidates seven previously separate platforms into a single, integrated communications solution.

Voxeo claims that it's VoiceCenter Fusion Servers are the first turnkey telephony platforms to offer VoiceXML IVR, CCXML call control, speech recognition, speech synthesis, call conferencing, call recording and a PSTN-to-SIP Voice over IP (VOIP) gateway in one integrated, rack-mount server. All of the Fusion Server's capabilities are built around the SIP VoIP standard, enabling rapid integration with both existing and new PSTN, PBX, and VhIP deployments.

"VoiceCenter Fusion Servers offer flexible, universal telephony ports with best-of-breed technology from Voxeo, Nuance, ScanSoft, Paraxip, and Intel/Dialogic," said Zack Angelo, Director of Software Development at Versabar. "Voxeo's turnkey servers uniquely enable us to deploy a single, low-cost platform for any telephony application requirement."

The VoiceCenter Fusion Server's capabilities include:

VoiceXML IVR - enables enterprises to create and deploy telephony applications that integrate with their existing web infrastructure to automate recurring sales, service, collections, inquiry and support calls to and from their customers.

Call Control XML (CCXML) - delivers capabilities to intelligently screen, transfer, and initiate traditional and SIP telephony calls, including integration with call center platforms via a built-in Computer Telephony Integration (CTI) interface.

Speech Recognition and Synthesis - allows callers to use comfortable, spoken commands to control the Fusion platform, and allows the Fusion platform to respond with human-like spoken audio output.

Call conferencing - lets up to 48 callers meet in a crystal-clear telephone conference. Plus, the CCXML-based conferencing engine allows enterprises to create custom conferencing applications that integrate with existing directory servers or applications.

Call recording - provides call logging / recording capabilities on all or a portion of all calls to address call center audit, third party financial verification, corporate security, FCC telesales, and other call recording requirements.

PSTN-to-SIP VoIP Gateway - connects the Fusion Server's SIP-based features - and external SIP-based telephones or platforms - to any PBX, call center, local phone company, or long distance carrier via built-in analog or digital T1/E1/PRI phone jacks.

Analysts and industry experts at Forrester Research predict that VoIP-based IVR - used at the core of Voxeo's Fusion Server - will be one of "only a few technology areas [that] will generate solid enterprise demand" in 2005.

The new product includes four models with varying telephone port types and densities:

- VoiceCenter IVR Fusion Server 4 - 4 analog ports starting at $4,495
- VoiceCenter IVR Fusion Server 12 - 12 analog ports starting at $9,995
- VoiceCenter IVR Fusion Server 24 - 24 digital T1/PRI ports starting at $16,995
- VoiceCenter IVR Fusion Server 48 - 48 digital T1/PRI ports starting at $26,995

The VoiceCenter Fusion solution has already won customers in a number of fields, including one of the nation's largest HMO's and the United States Department of Defense. Voxeo's VoiceCenter Fusion Servers complement Voxeo's existing VoiceCenter IVR Software and VoiceCenter IVR hosting solutions, and are available immediately.

I reviewed Epygi Technologies Quadro 4X product when it literally just came out - it may have even been a beta version, I don't recall. Epygi is offering a "discounted show special" at Internet Telephony Expo. Yet another reason why you should attend this show - the discounted VoIP products you can get. Of course, I get these VoIP products for free - it's called the "Tom Keating VoIP Blogger Special". Actually I write reviews for Internet Telephony, so maybe I should call it the "Tom Keating Internet Telephony Reviewer & VoIP Blogger Special"

Epygi Technologies Lowers Price 25% on Quadro 2x For TMC Internet Telephony Conference and Expo

DALLAS, TEXAS (February 22, 2005) Epygi Technologies, Ltd. announces consolidation of its entry level IP PBX products into a single product offering, the Quadro 2x IP PBX. Also, the company has increased IP Phone extensions (now included) for all Quadro IP PBX products. New models will be exhibited Feb. 22-25 at the TMC Internet Telephony Conference and Expo at the Hyatt Regency Hotel in Miami, Florida. A limited quantity of Quadro 2x units are available for purchase at the show at a special buy and carry show price. Join us at Booth 202, ABP and at Booth 506, NETXUSA.

“Epygi continues delivering high value and low cost to its small business customer. Even the smallest office or home office can now afford its own IP PBX.”, said Tom Boone of NETXUSA.

Epygi’s IP telephony products offer a feature-rich, scalable migration into IP telephony for the small business, SOHO and Teleworker markets. Epygi meets a market need combining analog lines, analog and IP phones with VoIP carriers in one straightforward product. The Quadro 2x now supports 2 IP phones and 2 analog phones. It offers a VPN, firewall and a host of PBX and Internet features at a reduced price. The Quadro 4x supports 4 IP phones. The Quadro 16x supports 8 IP phones.

Epygi is building a world-wide network of distributors and resellers. “By consolidating our entry level product into one offering we’ve increased the value of the offering while keeping the price low. Our on-going mission is that Epygi will be known for its depth of useful features, ease of use, and low cost to the end user.” says Jeff Kirchner, CEO/President of Epygi Technologies.

Net2Phone (NASDAQ: NTOP) and Empresa Telecomunicaciones de Bogota
(ETB), the largest telecom operator in Colombia, today launched a joint
offering of a complete suite of co-branded consumer and corporate VoIP
services across Colombia.

Uh oh, the Columbians have VoIP now! What's the DEA going to do now? It's much harder to intercept VoIP calls. Before I offend someone, I am only kidding...

I just received the following pitch... (followed by my comments below)

Human resources departments spend an inordinate amount of money each year trying to streamline labor management processes.  Due to human error from manual processes, many employees are getting away with working very few hours. Companies are looking for a variety of technology solutions to decrease labor management, thereby, increasing productivity.

Thanks to the advances in technology and phones, specifically IP phones, they can now be used to help with labor management processes and help companies save money and time overall. Using IP phones for labor management saves money by eliminating hardware needs, such as time clocks or biometric devices, while providing broader functionality.

Now employees can clock in and out, review time sheets, check schedules, receive text messages, and view accrued benefits - all from IP Phone-enabled workstations. Clocking in via the phone also tends to reduce the stigma of punching into a time clock. Employees feel more at ease and are more willing to comply with the system. In turn, managers can track employee activity and produce highly accurate job costing and productivity reports.

While I think this is an interesting application of IP phones, I could have sworn traditional PBXs with ACDs already have this time-sheet capability. I could be mistaken, but ACDs track logons, logoffs, lunch breaks, etc., so it shouldn't be hard to integrate the ACD data into time sheets, payroll, etc.

Still, I'm sure IP phones due to their very nature of residing "on the network" are a much "cleaner" integration and are also much easier to integrate with databases and other external databases.

Check out Rich Tehrani's blog entry from yesterday about Popular Telephony announcing Commoca and Texas Instruments agreement to embed Peerio in color IP Telephony terminals. The news was made to coincide with the "opening day" of Internet Telephony Expo. I can't wait to see Popular Telephony's product live in action on the exhibit floor!

I would have blogged Popular Telephony's news yesterday, but I was too busy trying to get these damn Linksys WRE54G wireless expanders/extenders to work so I could extend the WiFi range at Internet Telephony Expo. Yes, in addition to VoIP blogging, writing TMC Labs reviews, and managing TMC's computers, I along withVahid Hashemian are responsible for configuring the show's WiFi network. So you know who to complain to if WiFi doesn't work. Maybe I shouldn't have admitted to being the WiFi guru? Ahh too late now. Cat's outta the bag.

You would figure Linksys (now Cisco) would be a plug and play affair - after all, Linksys targets the home consumer.

Well, let me tell ya, it was anything but plug and play. The WRE54G has two LEDs that are both supposed to be blue (though nothing in the manual tells you this, an exhibitor said he has one at home and it's supposed to be all blue). The units we had had installed had one blue light and one red light each. After some tinkering, Vahid and I got it to work, but still we had 1 red light and 1 blue light. I then upgraded the firmware on the WRE54G and finally was able to get both lights to be blue, but only on the unit closest to the access point.

Unfortunately, we were only able to use 1 extender. We thought maybe we could place an extender every 30 feet and it would amplify the 2.4Ghz radio frequency, but alas, we couldn't get more than 1 extender to work. Looks like unlike cellular phones which can "hop" from tower to tower as you drive, these extenders don't work like that. As far as we could tell, only 1 extender can communicate with the main access point (AP).

Fortunately, we have multiple APs, so we'll just wire up the weak spots where needed.

Interesting news from a Latin America-based VoIP company which is launching a new service at the Miami-based VoIP show I am covering. I find it fascinating how many Latin American companies come to the Miami-based Internet Telephony Expo to launch new products and services. I've been to all the past Miami Internet Telephony Expos and it seems like year after year this show draws a stronger Latin American presence.

Sure, the Los Angeles-based Internet Telephony Expo draws several Latin American companies and I've seen a few Latin American companies at VON, but neither of these shows have the massive number of Latin American companies and even Latin American attendees. I've looked at the attendee demographics for the Miami show and the number of attendees from Latin America is staggering! The Miami show is it if you are looking for Latin American-based VoIP solutions period.

In any event, GlobalNet is launching a turn-key VoIP solution targetting service providers, ISPs, etc. called iDial IP. Think of it as a hosted Vonage solution. If you want to be the "next Vonage", you can use GlobalNet's hosted solution. There are other companies offering hosted broadband VoIP solutions, including VoIP Americas (now owned by VoIP, Inc.), Telic.net, and others.

Check out the release:

GlobalNet to Unveil The Ultimate Turn Key Consumer VOIP Solution for Service Providers at the INTERNET TELEPHONY Conference & EXPO Miami, Feb. 24, 2005.

GlobalNet Corp, Ranked Among the Top 10 Service Providers of International Wholesale Voice and Fax Termination to Mexico, Central and South America, is Proud to Announce the Launch of the Most Flexible Private Label Consumer VOIP Platform designed and built specifically for Service Providers.

The Completely Customizable Solution known as iDial IP, (www.idialip.com) provides everything required to turn an existing Service Provider into a Leader in VOIP technology without undertaking any of the obstacles involved in building out a Carrier Class VOIP Platform.

iDial IP is a Private Label Solution designed for ISPs, Cable Companies, Telcos, and all other Providers desiring to offer VOIP services to their Local Market.

The iDial IP platform is operated by GlobalNet's Proprietary VOIP Software Package which allows Clients to build a completely custom solution to fit the needs of specific local markets.

iDial IP's Flexible Billing Engine Provides Web Based functionality for managers to setup Custom Calling Plans by Country or Region with Custom Rate Management for Profitability Control.

The Consumer experience is powered by Linksys VOIP Hardware and is completely Plug and Play while providing advanced calling features such as Follow Me Services, Web Based Voicemail a nd Real Time Accounting features.

GlobalNet Corp is ranked among the top ten US service providers of International Wholesale Voice and Fax Termination to Latin America, and counts more than 30 Tier 1 and Tier 2 carriers as customers, apart from a host of other reputed global service providers and major international telecommunications organizations including AT&T, MCI/WorldCom, Qwest, Global Crossing, IDT, Broadwing and ITXC.

This was worth sharing. More positive VoIP news...

Worldwide Carrier VoIP Equipment up 36% to $1.7B in 2005 and North American Subscribers Grow 1M to 17M 2004 to 2008

February 23, 2005-Worldwide service provider next gen voice product revenue totaled $1.71 billion in 2004, a healthy 36% gain over 2003, setting a new high, according to Infonetics Research's quarterly market share and forecast service, "Service Provider Next Gen Voice Equipment."

Infonetics projects revenue to reach $5.9 billion in 2008, a five-year CAGR of 36%.

"We're starting to see strong equipment sales translated into tangible services," said Infonetics Research's Kevin Mitchell, directing analyst and author of the report. "For instance, there were 1.1 million residential/SOHO voice over IP subscribers in North America in 2004--with almost half coming directly from MSOs--and we expect that number to soar to 17.4 million by 2008."

"North America was a hot spot in 2004 as carrier adoption moved into the big time, and we expect Europe to start taking shape this year," Mitchell continued. "As we move deeper into the 21st century, it becomes more apparent that IP networks are the next gen networks for all forms of communication. It's hard to find a carrier not modernizing their network with VoIP or planning to do so."

The question of why so many network operators are converging their fixed networks or contemplating it is being addressed today at the IEC 21st Century Communications World Forum in London, where Mitchell is participating in the "Making Convergence Pay: Results to Date and Future Prospects" panel. As more and more providers invest in convergence to increase profitability via reduced costs and revenue growth from new services, the VoIP equipment market will continue to take off.

4Q04 and 2004 Market Highlights
- The media server and voice application server segments posted significant quarter-over-quarter gains
- Class 5 softswitch revenue made up over half of all softswitch revenue in 2004
- In a very fragmented market, Sonus is the worldwide media gateway market share leader with 16% share in 2004, and Cisco was second for the year
- Nortel leads the softswitch market for 2004 and for both class 4 and class 5 applications
- The 2004 geographic breakdown for total next gen voice products shifted to North America as many carriers in that region started serious deployments last year: 48% North America, 19% EMEA, 28% Asia Pacific, and 5% CALA

The Service Provider Next Gen Voice Equipment report tracks VoIP subscribers, media servers, session border controllers, media gateways (including RAC VoIP gateways, ATM switch voice gateways, and packet voice gateways), voice application servers, softswitches, and class 5 packet switches. Forecasts are updated quarterly and cover all regions (worldwide, North America, EMEA, Asia Pacific, and CALA).

Companies tracked in this service include Acme Packet, Alcatel, AudioCodes, BayPackets, Broadsoft, CIRPACK, Cisco, Convedia, CopperCom, Ericsson, Huawei, IP Unity, Italtel, Jasomi, Kagoor, LongBoard, Lucent, Marconi, MetaSwitch, Mera, Netrake, NexTone, Nortel, Pactolus, Sansay, sentitO, Siemens, Sonus, Sylantro, Tekelec, Ubiquity, UTStarcom, Veraz, Xener, and others.

For the table of contents, log on to Infonetics Research's Information Portal at www.info.infonetics.com.

$1.7B1 billion VoIP minutes served by Veraz Networks per month? Nice! Not to rehash the recent past, but I can't help but be reminded of Dvorak's claim that VoIP isn't ready for primetime. (see http://blog.tmcnet.com/blog/tom-keating/voip/voip-blog/galitzine-vs-dvorak-on-voip.asp)

Here's an excerpt of the release:

Veraz Networks Records Strong Worldwide VoIP Sales Growth in 2004

  Network and Enhanced Service Solutions Deployed in 30 countries

February 23, 2005-Veraz Networks announced that its VoIP solutions have been deployed in 30 countries, representing both complex network environments and newly deregulated markets. Veraz VoIP solutions are carrying over 1 billion minutes of traffic per month. Service Providers in newly deregulated markets such as India, Turkey, Bulgaria, and Russia require a stable solution that works with their IP environment, including the demands of handling satellite, wireless and international traffic. Even with mobile originated traffic carried over satellite links, Veraz’s voice quality and stability stand out under the most rigorous conditions, due to best-in-class handling of latency, jitter and packet loss.

This is an interesting release if only for the simple fact that its claiming a 2% drop in enterprise routers purchased. I have to wonder if VoIP will be the "shot in the arm" that will boost the enterprise router market. There will be more and more "converged devices" that feature an integrated router, firewall, VoIP gateways, SIP Registrar, WiFi, and maybe even integrated 3G or WiMAX. So maybe Cisco's outlook, (the "big enterprise boy on the block") isn't so bad afterall. Time to change from "sell" or "hold" to "buy"? Any stock analysts want to pipe in? Actually, I don't think Cisco's stock is positioned for any substantial short-term gains (next 1-2 years), but then again, I said the same thing about Dell 7 years ago and they keep surprising the analysts and me...

Enterprise Router Market Totals $3.4 Billion in 2004, Down 2%

BOSTON, Massachusetts, February 22, 2005--Worldwide enterprise router revenue is down 2% both for the quarter and the year, totaling $816 million in 4Q04 and $3.4 billion in 2004, while unit shipments are up 8% quarter-over-quarter and 4% year-over-year, according to Infonetics Research's "Enterprise Routers" quarterly worldwide market share and forecast report.

Revenue is projected to return to the 2003 level of $3.5 billion in 2006, and should hit $3.6 billion in 2008, a five-year CAGR of 1%.

Despite router revenue being down 1%, Cisco actually gained a point in market share in 4Q04, maintaining its strong lead with 84% revenue and 77% unit market share. Vanguard is in second place for revenue share with 1%, and Allied Telesyn is second in units with 5%.

"Twelve percent of 4Q04 enterprise router revenue came from the sale of secure routers, up a percent from 3Q04," said Infonetics Research's Matthias Machowinski, directing analyst for enterprise voice and data. "We project secure routers to continue to take up a growing slice of the router revenue pie, accounting for 29% of total router revenue by 2008. Over time, router vendors could add security features into their routers as a default offering, at no extra charge, causing the standard enterprise router category to disappear."

Enterprise Routers tracks standard and secure high-end, mid-range, and low-end/SOHO routers. Forecasts and market share are updated quarterly and cover all regions (worldwide, North America, EMEA, Asia Pacific, and CALA).

Companies tracked in this service include 3Com, ADTRAN, Allied Telesyn, Cisco, Enterasys, Huawei, Juniper, Lucent, Nortel, Siemens, Tasman, Vanguard, and others.

For the table of contents, go to www.info.infonetics.com. For sales, contact Larry Howard, vice president, at larry@infonetics.com, 916.933.3543.

I'm here safe and sound in sunny Miami. I'm currently sitting in on the "SIP's Role in Open Source" speaking session which is moderated by Rich Tehrani. I missed a few of the speakers since I arrived in the middle of the session.Right now SIPfoundary is speaking about GNU and GPL.Well, I'm a good multitasker, but I should listen to the rest of the speeches. But I'll post one bit of news before I sign off...

ISN Telcom, the Miami based CLEC, has deployed Kagoor's VoiceFlow session border controllers for its ipFONE VoIP service (see release below). Kagoor enables ISN to provide its customers with unlimited calls to Latin America in addition to the United States, and Canada for a single monthly fee.

Jon Arnold, VoIP Program Leader of Frost & Sullivan stated that deployment of Kagoor's session border controllers "will be a key factor in helping service providers such as ISN gain market leadership."

ISN Telcom Selects Kagoor's Session Border Control For NAT Traversal and Network Protection

Kagoor enables Miami-based CLEC to provide unlimited calls to Latin America

SAN MATEO, CA and MIAMI, FL - February 22, 2005 - Kagoor Networks, a leading provider of session border control solutions, today announced that ISN Telcom, the leading pan-regional, phone, data, wireless and Internet services company, has deployed Kagoor's VoiceFlow 1000 session border controller for its ipFONE VoIP service. Kagoor enables the Miami-based CLEC to provide its customers with unlimited calls to Latin America in addition to the United States, and Canada for a single monthly fee.

"VoIP will fuel the rapid growth of telephone service between Miami and Latin America", said Jon Arnold, VoIP Program Leader of Frost & Sullivan.
"Deployment of session border controllers provided by companies like Kagoor will be a key factor in helping service providers such as ISN gain market leadership."

Kagoor's session border controllers provide network protection and security for ISN's VoIP network. Kagoor's network protection applications include topology hiding, intrusion prevention, call admission control, and protection from denial of service (DoS) attacks, which provides ISN's customers with secure, high-quality VoIP calls.

Kagoor also provides its hosted NAT traversal solution, which permits incoming VoIP calls to securely pass through firewalls. Kagoor's unique 3-way architecture ensures the optimum platform for the NAT traversal applications. With the VoiceFlow network-hosted NAT traversal solution, ISN delivers its customers VoIP service without adding additional premises equipment. VoiceFlow fully supports all major VoIP protocols (SIP, H.323, MGCP), and provides carriers with the most extensive vendor library of protocol extensions available.

"Kagoor delivers all of the capabilities that ISN's VoIP network will need at its borders," said Jonathan Lieberman, president of ISN. "VoiceFlow gives us a real competitive edge by enabling economical VoIP calls between Miami and Latin America for our business and residential customers."

ISN Telcom, one of the first competitive local exchange carriers (CLEC) nationwide to roll out VoIP services, supplies telephone service directly to residential and business customers through its ipFONE service without interconnecting through a local carrier.

"ISN Telcom is becoming the market leader in the rapidly growing, Miami metropolitan area," said Jim Greenway, vice president of marketing for Kagoor. "We're happy we could be a part of their continued success."

About Kagoor Networks
Kagoor is a leading supplier and innovator of session border control technology that ensures customers fast, expert delivery of IP communications services (voice, video, multimedia). Our award-winning VoiceFlow series of session border controllers overcomes technical roadblocks typically encountered at VoIP network borders. With approximately 100 worldwide customer deployments and the largest number of Tier 1 telecom partnerships, Kagoor offers customers the most powerful, and scalable family of session border controllers. The company is headquartered in San Mateo, CA, with offices throughout the world. More information is available at www.kagoor.com.

Some interesting news...AudioCodes and Vail Systems have partnered with Microsoft to SIP enable Microsoft Speech Server 2004

Here's the release:

AudioCodes and Vail Systems offer SIP-enabled product bundles to enable Microsoft Speech Server 2004 deployments.

February 21, 2005 AudioCodes (NASDAQ: AUDC), a leading provider of Voice over Packet (VoP) technologies and Voice Network products, and Vail Systems a leading provider of SIP-based IP telephony platform technology, speech application solutions and hosting services today announced availability of a set of SIP-based product bundles to enable development and deployments for Microsoft Speech Server 2004. The new product bundles consist of the newly certified version of the Vail SIP Telephony Interface Manager (TIM) for Microsoft Speech Server 2004 and either an AudioCodes TP260/SIP or an AudioCodes Mediant 2000 Voice over IP Media Gateway. These bundles offer Microsoft Speech Server developers and integrators a combination of software and hardware products that are ready-to-deploy, leveraging SIP to enable high performance, distributed and scalable speech applications.

"We're pleased to see two premier vendors partner in enabling SIP-based speech applications", said James Mastan, director of marketing for Microsoft's Speech Server product group. "The availability of the SIP bundle today will protect customer investments in traditional TDM telephony systems while providing an evolution path to future Microsoft SIP based products."

"The bundle of the Vail SIP TIM and AudioCodes SIP media gateways enables developers to immediately leverage SIP in creating powerful distributed speech applications based on Microsoft Speech 2004", said Lior Aldema, VP of Marketing for AudioCodes. "As the speech marketplace continues to mature and further adopt SIP, AudioCodes hopes to enable the evolution with a wide range of gateway and transcoding enabling technologies."
AudioCodes' TP-260/SIP offers value added resellers and integration partners a PCI-based digital media gateway board for enhanced voice services and enterprise applications. Offered in densities of 1, 2, 4 and 8 E1/T1s on a single PCI slot, the TP-260/SIP is an excellent building block for space-saving single-server enterprise solutions. The TP-260/SIP supports a range of central office and PBX TDM protocols including ISDN, CAS and E1R2 signaling. Leveraging an on-board SIP stack, the TP-260/SIP eliminates the need for PCI drivers and associated operating system compatibility issues.

AudioCodes' Mediant 2000 is a stand-alone digital media gateway in a 19 inch-wide 1U package. Offered in densities of 2, 4, 8 and 16 E1/T1s, the Mediant 2000 is ideal for distributed systems that require a separate media gateway or geographical diversity. Sharing a common software base with the TP260/SIP, the Mediant 2000 also supports SIP and a range of central office and PBX TDM protocols including ISDN, CAS and E1R2 signaling
"Vail is pleased to announce this offering with our long-time partner AudioCodes," said James Whiteley, President and CEO of Vail Systems. "Their industry-leading PSTN/VoIP gateway products provide Microsoft Speech Server and Vail SIP TIM customers even more deployment options at highly competitive price points compared to TDM-only alternatives."

The Vail SIP TIM enables enterprises to connect Microsoft Speech Server 2004 to industry standard SIP-based VoIP telephony equipment including AudioCodes SIP Gateway cards and systems. Meeting Microsoft's high performance standards, the Vail SIP TIM combines excellent service quality with the compelling business value of standards-based IP communications. For additional Vail SIP TIM product information please visit http://www.vailsys.com/mss/.

Ahhh yes, these two Ingate releases below are but the first of many VoIP-related announcements to be made at Internet Telephony Expo... Let the massive amount of VoIP news announcments begin!

I leave my house at 4:45am (fun fun) - for a 7:29am flight, so this will be may last blog entry until I get to Miami. Hmm, "this will be my last blog entry" sounds 'dire' doesn't it?

Ingate SIParator Wins Product Leadership Award From Frost & Sullivan

Company's Innovation in SIP Recognized by 2005 Science & Technology Awards

INTERNET TELEPHONY CONFERENCE & EXPO, MIAMI, February 22, 2005 - Ingate Systems (www.ingate.com), whose award-winning security products enable enterprises to solve the problem of letting Session Initiation Protocol (SIP) traverse their firewalls, has been named the recipient of the Frost & Sullivan 2005 Science & Technology Award, Product Leadership for IP Protocols (specifically, SIP) category. 

The Award is in recognition of the Ingate SIParator, a unique device that connects to an existing network firewall to seamlessly enable the traversal of SIP communications, including Network Address Translator (NAT) traversal. The Award also acknowledges the company's leadership in developing technologies based on the SIP protocol.

Winners are chosen based on the level at which their product contributes to the industry in terms of market acceptance. Products are evaluated based on the degree of innovation, the estimated market penetration rate, and acceptance by intended end-users. Ingate's successful product development strategies and the degree with which the SIParator has met customer needs and requirements was a significant factor.

The selection process included primary participant interviews and interviews with end users, distributors, suppliers as well as extensive primary and secondary research. 

"The Ingate SIParator is a one-of-a-kind product that brings SIP - and access to applications such as VoIP - to any enterprise, while solving critical issues such as NAT traversal," said Olle Westerberg, Chief Executive Officer, Ingate Systems. "It is an honor to have our leadership in developing SIP-enabling technologies recognized with such a prestigious award." 

"Ingate's SIParator has garnered an enthusiastic response from industry end-users," said V.R. Yoges, Research Analyst, Technical Insights, Frost & Sullivan. "The SIParator provides users with a range of options while bringing down the cost of providing SIP-capability to the enterprise network: it is compatible with legacy firewalls, and its modular nature allows a user to add more functionalities later, if need be - as in a growing organization. The Frost & Sullivan Award for Product Leadership acknowledges Ingate Systems' development of this truly industry-leading product technology."

In addition to the SIParator, Ingate produces and sells the world's only fully SIP-capable enterprise firewalls. Ingate's products also include optional software modules for VoIP survival in hosted SIP-based hosted PBX environments, far-end NAT traversal solutions for remote VoIP and Quality of Service.

For more information, please visit Ingate at Booth 604 at Internet Telephony.

Ingate VoIP Survival - Full Redundancy CPE for Hosted VoIP

Validated by BroadSoft as a Customer Premises Backup Enhancement to BroadWorks

 
INTERNET TELEPHONY CONFERENCE & EXPO, MIAMI, February 22, 2005 - Ingate Systems (www.ingate.com), whose award-winning security products enable enterprises to solve the problem of letting Session Initiation Protocol (SIP) traverse their firewalls, showcases Ingate VoIP Survival at Internet Telephony Conference & Expo 2005. Ingate VoIP Survival is an application module for hosted VoIP communications services that secures full redundancy in a SIP-based Hosted PBX or IP Centrex environment all the way out to the customer premise. 

Ingate VoIP Survival, which can be used with any hosted VoIP service, has been validated by BroadSoft, Inc., the leading provider of VoIP application software, to serve as a customer premise backup to enhance the reliability and availability of the BroadWorks VoIP application platform.

In the hosted environment, Ingate VoIP Survival ensures continued communications service even if the hosted server goes offline due to connection failure or malfunction. This increased availability means that unnecessary downtime is minimized.

"Ingate VoIP Survival is suitable for any enterprise using a Hosted PBX platform looking to maintain the highest reliability and availability for their network," said Olle Westerberg, Chief Executive Officer, Ingate Systems. "We are proud to deliver this functionality to BroadWorks as part of their interoperability validation program, our first implementation of the product." 

 
Reliable, Hosted VoIP with CPE Redundancy 

A hosted communication platform offers many advantages to enterprises adopting VoIP. However, there are occasions when it is desirable to have a redundant capability at the customer premise. Should a network outage occur, Ingate VoIP Survival offers continuous, uninterrupted availability of mission-critical VoIP service.

"Ingate VoIP Survival offers the benefits of traditional outsourcing while securing the availability of one of the most mission-critical applications for businesses today. In addition, VoIP Survival maintains vital perimeter control and security at the enterprise. Combined, these functionalities minimize risk and encourage enterprises to move to VoIP to gain the cost and productivity benefits of hosted voice communications," said Scott Wharton, vice president of marketing for BroadSoft.

Ingate VoIP Survival Module

In a Hosted PBX environment, the server situated in the service provider's core network is in constant contact with - and dynamically sending data to - the Ingate Firewall or Ingate SIParator situated on the company LAN. The intelligence built into the hosted solution is automatically transferred to the local side, including settings and registrations. The Ingate CPE keeps track of updates and stores a mirror image of user settings.

In the event that contact with the hosted server is lost, the Ingate CPE will automatically detect the failure and make use of stored settings to ensure that communication methods are maintained. By using the mirror image of user settings, local calls within the domain are handled by the CPE. Any calls made to the outside will automatically be routed towards the SIP/PSTN gateway, given that there is SIP/PSTN gateway on the LAN. The CPE will automatically detect when the communication link is restored.

Ingate VoIP Survival is available now for all Ingate Firewall and SIParator products. 

Well, Blogads (www.blogads.com) accepted my application to join their "blog advertising network". From what I understand Blogags is very discerning as to which blogs they accept. I guess I should feel proud. They cited my coverage of not only VoIP, but gadgets as one of their reasons for accepting my blog into Blogads.I rebuilt my blog to include the Blogads javascript code, so it's live now. If you are interested in advertising on my blog, you can simply click on the Blogads advertisement located beneath the Google ads.

I head out to Miami tomorrow for Internet Telephony Expo and this trip couldn't have come at a better time. for two reasons. One, it snowed today, which means I will be doing some back-breaking snow shovelling sometime today - so I will be glad to enjoy Miami's warmer climate. Secondly, my Mitsubishi 3000GT died on Saturday while attempting to teach my wife how to drive a stick shift. DOH! I think she threw out a throw bearing or bent a fork when she floored it and dropped the clutch. The damn clutch won't "catch" now. I lift up the clutch pedal and car goes nowhere. I don't think it's the clutch though - it's only a year old.

Well, the good news is that at least while my car is getting worked on, I will be in Miami - no need to pay to rent a car to get to work. So the show timing couldn't have been better. I could of course drive my Viper to work, but the snow storm will be leaving a wake of slush, sand, and salt - none of which are good to be driving in, especially with this car. The Viper's wide tires are great on dry traction, but notoriously bad in slushy or even wet conditions (forget about driving a Viper in icy conditions unless you have a deathwish). Too much torque for its own good!

Anyway, I will be blogging from the show and reporting the latest happenings here.

p.s. I really should pack, I'm such a procrastinator..