VoIP Monitor

eBay and Silver Lake Investor Group Settle Skype Litigation with Joltid Limited

VoIP Monitor — Fri, 06 Nov 2009 22:10:35 GMT

eBay announced that the investor group led by Silver Lake, which had previously entered into a definitive agreement to acquire a majority stake in Skype from the company, has reached a settlement agreement with Joltid Limited and Joost N.V. that gives Skype ownership over all software previously licensed from Joltid and ends all litigation currently pending against the investor group and eBay at the closing of the acquisition.

As part of the settlement agreement, Joltid and Skype founders Niklas Zennström and Janus Friis will join the investor group, contributing Joltid software and making a significant capital investment in exchange for a 14 percent stake in Skype. As a result, Silver Lake and other investors including Andreessen Horowitz and the Canada Pension Plan Investment Board, will together hold 56 percent of Skype and eBay will retain 30 percent. As previously announced, eBay will receive approximately $1.9 billion in cash upon the completion of the sale and a note from the buyer in the principal amount of $125 million. The deal, which values Skype at $2.75 billion and is not subject to a financing condition, is expected to close in the fourth quarter of 2009.

Commenting on the agreement on behalf of the investor group, Silver Lake Managing Director Egon Durban said: “We are very pleased to have the litigation resolved. We remain confident in a great future for Skype, and we look forward to working with Niklas, Janus and the other investors as partners to help the company achieve its full potential.”

The investor group will no longer include Index Ventures, which has withdrawn from participation. Commenting on its decision to withdraw, Danny Rimer of Index Ventures, said: “We are pleased that Skype will now be able to put litigation behind it, and we wish Josh Silverman, his team and the Skype investors well in continuing to grow a great business. Although Skype has the potential to be a great investment, the deal terms changed for Index such that it no longer matches our investment criteria and thus we have decided not to participate in the transaction.”

FreedomVOICE Proposes Telecommuting to Tackle the H1N1 Virus

VoIP Monitor — Thu, 05 Nov 2009 19:42:51 GMT

FreedomVOICE has proposed telecommuting as a viable option to control the spread of the H1N1 virus in the workplace.

Whether we like it or not, the H1N1 virus is among us. While the virus affects our health, it can also affect our business. But smart CEO’s are taking precautions to end the virus’s daily commute.

“If an employee is feeling ill, they should be given the opportunity to work from home if they can still be productive,” said Eric Thomas, CEO of FreedomVOICE systems. “If the entire company is infected with the flu at the same time it could be devastating.”

Telecommuting is a weapon being used at the forefront of the fight against H1N1. And with today’s advances in telecommunications technology there are a variety of options available to business owners who want to enable their employees to work from home. A popular telecommuting technology is hosted VoIP PBX phone systems. These Internet based phone systems allow you to seamlessly connect all members of your office, regardless of where they are located. One such product is the FreedomIQ Hosted VoIP PBX by FreedomVOICE.

“FreedomIQ is a VoIP phone system that you can utilize anywhere you have Internet access,” explains Thomas. “The device is perfect for telecommuters. It enables unified messaging, advanced call reporting, automatic call distribution, and more for a value driven rate.”

While working from home used to be frowned upon, today, the practice has been largely adopted by companies in the US.

“According to recent research conducted by Techaisle, currently over 80% of businesses in the US have a work at home policy,” says Thomas. “But all businesses have the ability to help alleviate fears associated with the H1N1 virus by allowing those concerned to work from home.”

FreedomVOICE offers the following as a basis for companies to evolve their own H1N1 work at home policy from.

How Managers Should Handle Employees Feeling Ill

Employees feeling ill should immediately be given the opportunity to work from home.
Employees can continue to work from home as long as their manager feels they are being productive.
The manager should delegate a set amount of work to the employee each day.
Each day the employee should submit a list of what they have accomplished.
Based on the accomplishments, the manager can decide whether the employee can work at home again or should take a sick day.
Employees should only return to the office once they have been cleared by a doctor.

What Employees Need to be Productive While Working at Home?

Employees must be able to access all of the same items they can when they are in the office.
Employees should be able to take their computer home or connect to their work computer remotely.
Employees must set up their call forwarding to forward to a line that can be answered from home.
Employees must have an area at their home where they can work without being distracted.

Other Important Items

Employees should be able to be reached by phone at all times.
Employees should be able to be reached by email at all times.
Employers should be able to be reach employees by dialing their work extension.
If an employee misses a call they should call the employer back ASAP.

FreedomIQ is a business phone system which allows employees to access all of the same VoIP phone features at home as they can at the office. Managers can see how long employees have been on calls, who they have been on calls with, how many calls they’ve missed, how long they’ve been logged into their phone, and other important information. With these tools, managers can keep a close eye on employees, and employees can have all the advantages of office right at home.

snom and ESTOS Partner to Deliver Unified Communications and CTI to VoIP Phones

VoIP Monitor — Thu, 05 Nov 2009 19:08:07 GMT

snom has teamed with ESTOS to ease the deployment of Microsoft TAPI-enabled business applications, like unified communications and CTI solutions. The partnership allows ESTOS’ ECSTA for snom middleware to convert snom’s Computer Telephony Integration interface protocol to the Microsoft TAPI standard. ESTOS ECSTA for snom, when used with ESTOS ProCall solutions, allows both companies to deliver unified communications and CTI functionality to medium-sized businesses and service providers.

The new ESTOS ECSTA for snom middleware enables snom’s 3XX Series and 8XX Series intelligent VoIP telephones to support additional call center, CTI and unified communications applications in both 32-bit and 64-bit Windows operating system environments.

ESTOS ECSTA for snom also enables snom phones to take advantage of several telephony functions through TAPI-enabled applications. Those functions include making, answering, holding and retrieving calls; consultation calls; single-step transfers; alternating calls; setting and receiving forwarding status; and setting and receiving do not disturb status.

Together with ESTOS ProCall, ESTOS ECSTA for snom is available in two versions: a solution for single seats and a solution for networks. The single seat version of ESTOS ECSTA for snom and ESTOS ProCall One is designed to enhance productivity for small offices and telecommuters. It enables features like click-to-call dialing from Microsoft Outlook. The network version of ESTOS ECSTA for snom and ESTOS ProCall Enterprise is aimed at medium-sized businesses with larger corporate networks or service providers that want to take advantage of unified communications functionality such as CTI, instant messaging and presence management.

Both the single seat and network versions are offered in license packages based on the number of lines required and both are compliant with 32-bit and 64-bit Windows operating systems, including Windows XP Professional, Vista, Windows 7 and Windows 2000, 2003 and 2008 Server.

ESTOS ECSTA for snom and ESTOS ProCall also offer intuitive first-time installation and easy control of snom phones. It supports all snom 3XX and 8XX series telephones equipped with firmware 7.3.4 and higher, including the new snom 870, the latest snom desktop phone that features a large, full-color touch-screen, HD voice compatibility and integration with Microsoft Office Communications Server.

Telchemy Ships over 50 Million Units of VQmon VoIP/IPTV Performance Analysis Software

VoIP Monitor — Thu, 05 Nov 2009 18:55:12 GMT

Telchemy announces that over 120 equipment manufacturers have successfully integrated Telchemy’s VQmon software into a wide range of products, such as IP phones, VoIP Gateways, Routers, IPTV Set Top Boxes, Residential Gateways, eMTAs, Probes and Analyzers. Over 50 million units of VQmon have been shipped, representing an annual growth rate of over 50%. This number does not include an estimated 200 million additional units shipped by a number of large ISPs who license VQmon on a subscription model basis.

The VQmon product family incorporates:

- VQmon/EP is the most widely used technology for IP Phones and Gateways to provide real-time VoIP performance measurement and reporting for a wide range of narrowband and wideband codecs. VQmon/EP generated RTCP XR and SIP RTCP Summary reports can be collected and analyzed using Telchemy’s SQmediator.

- VQmon/SA, designed for Routers, Probes and Analyzers to provide in depth analysis of VoIP traffic and IP performance, supports automatic detection of codec type and codec rate producing listening and conversational quality metrics

- VQmon/HD is integrated into Routers, Probes, Analyzers, and IP Set Top Boxes to provide a rich set of real time IPTV and IP Videoconferencing performance metrics. VQmon/HD is capable of analyzing encrypted/ scrambled IPTV streams and now supports the analysis of IPTV streams with FEC and Retransmission. VQmon/HD is in early commercial deployments in IPTV set top boxes and is already extensively used for service assurance by leading IPTV service providers.

VQmon products are compact, efficient, reliable and highly portable software agents that are integrated into a wide range of software and hardware platforms, producing accurate real-time, performance and QoE metrics for IP voice and video traffic. VQmon is unique in its ability to model the impact of time varying impairments on user perceived quality, which has been shown to materially improve the accuracy of QoE metrics.

Digium and TMC Host Sold Out Digium|Asterisk World 2010 at ITEXPO East in Miami

VoIP Monitor — Thu, 05 Nov 2009 18:34:15 GMT

Technology Marketing Corporation and Digium announce that all exhibit space has sold out at their Digium|Asterisk World 2010 conference, collocated with ITEXPO East 2010, January 20-22, at the Miami Beach Convention Center. Digium and TMC officials are in negotiations with Convention Center staff to identify additional space to for exhibiting opportunities at the event.

The Fourth Annual Digium|Asterisk World will be the largest and most comprehensive to date, with expanded content covering: call centers; cloud computing; government/enterprise/carrier applications; and a number of sessions identifying opportunities for the entire Asterisk ecosystem. This three-day conference and expo attracts a broad group of Asterisk developers, resellers and professionals who design and manage large enterprise, government and carrier networks.

Asterisk, created and maintained by Digium, is the world’s leading open source telephony engine and tool kit. Today, open source telephony solutions account for more than 18 percent of all PBX sales in North America, equaling more new lines than any single conventional PBX or key system manufacturer. Open Source PBX technology creates significant revenue for businesses that install Open Source PBXs, and provide GUIs, PBX appliances, gateways, SIP phones, VoIP trunking and installation services.

The upcoming Digium|Asterisk World will identify opportunities for all sectors of the Asterisk Community. The educational content covers the entire spectrum of issues related to Asterisk-based, open source telephony.

“Asterisk, celebrating its 10th birthday in 2009, is moving into the mainstream – at large enterprises, in government sectors, and amongst carriers,” said Mark Spencer, CTO and founder of Digium, and the creator of Asterisk. “The upcoming Digium|Asterisk World will help developers, resellers, and Asterisk end users learn more about the opportunities emerging today, and it will provide the hands-on training needed to capitalize on these opportunities.”

"We’re extremely pleased to once again host a sold out Digium|Asterisk World at ITEXPO. The event provides the perfect forum for open-source developers, users, and channel resources to make the connections they need to develop the next generation of open-source telephony applications,” said Rich Tehrani, CEO and group editor-in-chief of TMC. “We’re looking forward to a tremendous event in Miami, and we’re proud to feature Digium|Asterisk World as one of the featured collocated events at ITEXPO East 2010.”

TMC and Digium will publish a complete list of conference tracks and sessions in the weeks ahead.

Sponsorship opportunities remain available for companies to showcase their offerings to Digium|Asterisk World and ITEXPO attendees. Interested parties should contact TMC’s Joe Fabiano for more details. Registration for Digium|Asterisk World and ITEXPO is now open.

Telanetix Announces New Cordless Handsets for Small Business VOIP Market

VoIP Monitor — Wed, 04 Nov 2009 18:07:10 GMT

Telanetix announced the addition of cordless phones to its Digital Phone System suite of services based on the DECT standard. The Digital Phone System, marketed under the AccessLine brand, is available in several configurations and models for either home or small business offices. The system comes bundled with service that includes a complete business phone application suite that is based on the Telanetix nationwide VoIP Network.

The new cordless handsets offer more than 30 important business features, including: inter-extension dialing with non-cordless handsets on the same phone system, intercom, call park/pickup, headset compatibility, and many more. As with all AccessLine Digital Phone Systems, the cordless handsets arrive custom-configured to the specific business needs of each individual small business.

InCharge Systems Announces Reference System for SIP Security Interoperability Testing

VoIP Monitor — Wed, 04 Nov 2009 17:59:12 GMT

InCharge Systems released a hosted reference system of its ACerted Trust solution, available immediately for interoperability testing. The reference system, will be demonstrated Thursday, October 29, 2009 at the Illinois Institute of Technology 5th Annual VoIP Conference and Expo.

ACerted Trust is a solution for assuring the identity of end users and their operators that originate SIP requests for voice, video, presence or messaging communication sessions, based on the Internet Engineering Task Force standard RFC 4474, “Enhancements for Authenticated Identity Management in the Session Initiation Protocol”

In VoIP telephony today, the calling line identity presented to the receiver of a phone call can be easily altered if the communication traverses the open Internet, making those communications subject to abuse by unscrupulous telemarketers, vishing attacks for financial and identity information, theft of toll service and other serious problems. ACerted Trust assigns a cryptographic signature to SIP:INVITE messages, allowing receiving entities to check with a trusted certificate authority to verify the identity asserted by the caller. This enables the identity of the caller to be verified at any point in a communication, regardless of network operator.

The reference system released today is intended to support the implementation by the industry of RFC 4474 in various VoIP and SIP products and services, such as:

IP-PBXs, SIP-aware firewalls, session border controllers and softswitches.
End user devices such as IP phones, analog telephone adapters and software user agents such as PC softphones and mobile VoIP clients on smart phones.
Services and gateways such as SIP peering federations, SIP call termination providers and VoIP communication services providers
Consumer Internet and business process applications using SIP to embed voice, instant messaging and presence.

Today’s release for interoperability testing comprises the following hosted elements: a public / private key provisioning system, a certificate authority and a SIP proxy server that will allow the validation of calls based on their digital signature, as well as demonstrate the signing of SIP messages.

In addition to the reference system, ICS has contributed to the popular open source Asterisk IP PBX system a software module that implements support for RFC 4474 and interoperates with the ICS hosted reference system.

Grandstream Introduces its GXV3140 Desktop IP Multimedia Phone

VoIP Monitor — Tue, 03 Nov 2009 20:39:50 GMT

With an estimated 57% of adults in the U.S. having broadband access to the home1, the Internet makes it easy to deliver real-time video and web entertainment to enhance people’s lives. Today, people are spending longer periods of time away from loved ones due to work factors and the economic climate forcing Americans to travel less. An all new consumer multimedia phone has hit the market making it easy to see and hear family and friends around the world for free via an existing broadband connection. Using the broadband service provider of your choice and Grandstream’s built-in IPVideoTalk, the GXV3140 desktop IP Multimedia Phone offers, for no additional monthly fee, face-to-face two-way or three-way video calling out of the box.

High quality multimedia video devices packed with rich applications also bring an exciting social networking experience that lends to creating lasting business and personal relationships. In addition to real-time video/voice calls, the GXV3140 includes instant access to Web and social multimedia applications. With a 4.3” digital color LCD screen, the ‘always-on’ GXV3140 has an integrated web browser giving consumers convenient, one-touch access to personalized RSS feeds of real-time online information services (news updates, stock updates, weather forecasts, directory searches, etc.), thousands of Internet radio stations, social networking sites (Facebook) and popular streaming Internet entertainment sites like YouTube.

"Consumers have historically shied away from video calling due to its perceived complexity, the cost associated with endpoint hardware and of course the 'dreaded' monthly service fee. But with the Grandstream 3104 users will find themselves delighted by its ease of use, value price point and quality build. Plus it comes with the free IPVideoTalk service. Where I come from, we call this a great deal," said Garrett Smith, Director of Marketing at VoIP Supply and author of the popular VoIP site www.SmithOnVoIP.com.

GXV3140 Features

See and Hear the Person You Are Talking To (No Additional Monthly Fee)
Video Conferencing Bridge for 3 Way calling (Free of Charge)
Portable and Works Anywhere in the World with Broadband Connection
4.3” 480x272 digital color LCD display and 1.3M pixel camera with privacy shutter
Plug the Phone into the Stereo and TV for Larger Picture Projection and Sound
Full HTML web browser
Media Player that Supports YouTube streaming
IM with Yahoo/MSN/Google
Thousands of Internet radio stations & popular online music networks such as Last.fm
Digital Photo Frame
Web photo album with Google, Yahoo Flickr, Photobucket, Phanfare
Personalized RSS feeds of news/weather/stock, horoscope
Full duplex speakerphone
Supports 9 different languages
Facebook and Twitter
Optional WiFi Adapter

The GXV3140 desktop phone can be purchased for $249 through Grandstream’s worldwide distribution channel.

SonicWALL Discusses VoIP Firewall at VoiceCon

VoIP Monitor — Tue, 03 Nov 2009 20:18:48 GMT

SonicWALL will be exhibiting at the VoiceCon conference in San Francisco, November 2-5, where the company will be discussing the rising use of VoIP, how to maintain high Quality of Service and low latency for voice traffic in a secure network environment. SonicWALL invites attendees to join them at booth #503 to view product demonstrations, join associated seminars and meet with executives to learn more about the new threats facing companies today and which questions to ask vendors when considering such voice deployments.

Managing the VoIP Bandwidth Conundrum

The movement to VoIP technologies for many companies brings more traffic to already crowded networks, busy with multiple other applications and data transfers. Network managers and administrators are now asked to work to maintain security levels at an absolute maximum to ensure the integrity of the corporate network, while securing suitable bandwidth for "heavy network-usage" applications such as VoIP. Key to the success of such deployments includes the ability to prioritize applications on the network (i.e. VoIP based apps), keeping connections clean to prevent viruses and other malware accessing the network and having visibility into all network traffic to better control it.

To achieve the above, companies should plan on using an Application Firewall within the IT infrastructure, providing them with granularity to decide which apps have access to the network and which don't. Such Firewalls also ensure a clean connection is always made by allowing it to tag VoIP traffic guaranteeing it the highest priority when receiving, inspecting, assembling and accepting VoIP content and is the best path to maximizing call quality and providing network protection.

VoIP Introduces New Threats

While the expansion of digital telephony and teleconferencing in today's business has brought VoIP into the mainstream, so too has it brought new threats into the corporate environment. Not only is VoIP vulnerable to classic network threats such as SPIT (Spam over Internet Telephony), phishing, Trojans, worms and viruses; it also introduces a slew of other threats to the corporate network including Eavesdropping, Interception (or call hijacking), Denial of Service attacks and more.

More key considerations when implementing a VoIP based deployment can be found at: http://www.sonicwall.com/us/products/resources/10661_14513.html

Blabbelon Introduces a Game Changer in VoIP

VoIP Monitor — Tue, 03 Nov 2009 19:59:01 GMT

As communication companies continue to look for new ways to generate revenue with VoIP applications, keep current customers and lure new ones, Blabbelon enters the market and completely changes the rules of the VoIP game. The first browser-based VoIP chat tool using Skype’s SILK super wideband audio codec, as well as one of the first massively multi-user VoIP services, Blabbelon can handle thousands of simultaneous users at a fraction of the cost per user of other traditional services. Blabbelon provides a platform for a wide variety of users such as: gamers securely competing in team-based quests; businesses running 7,000 person global conference calls; or grandparents taunting grandkids over a game of Facebook Scrabble.

An HD-quality, push-to-talk VoIP chat service, Blabbelon requires no software downloads or complicated sign-on procedures – users register once, click and talk. With Blabbelon, session leaders can use, create and maintain their own private or public groups through a number of features such as texting, e-mail notifications and privacy management. Users can run Blabbelon in their Web browser, while simultaneously using other online applications – with no impact on their overall network performance, since Skype’s SILK codec was designed to use approximately 50% less network bandwidth than previously required, while delivering improved richness in audio quality.

Unlike other voice chat services that max out at 300 users, Blabbelon sessions can scale up to more than 7,000 simultaneous users, while still maintaining a low cost business model and offering an easy-to-use portal-style entry. A breakthrough in VoIP services, Blabbelon is ideal for a variety of consumer and business applications, such as e-commerce, online auctions and sales meetings/seminars.

Blabbelon will initially focus on a core audience of the millions of online gamers and players of Massively Multiplayer Online games such as World of Warcraft. The combination of Blabbelon’s technology and features and the SILK codec provides a perfect tool for this market, which, because of the cooperative nature of games, needs a no-download, high-quality, secure, reliable and scalable voice chat.

Report: Mobile VoIP - Plenty Of Talking The Talk, But Can It Walk The Walk?

VoIP Monitor — Tue, 03 Nov 2009 19:28:19 GMT

Research and Markets has announced the addition of the "Mobile VoIP - Plenty Of Talking The Talk, But Can It Walk The Walk?" report to their offering. Mobile VoIP is often perceived as a threat to mobile network operators, eating into their voice revenues and reducing them to mere "dumb pipes" It has also been perceived as "the next big thing" for a while now in the mobile market This report will look at the myths and realities of mobile VoIP, the potential and threats of managed VoIP, and draw mobile VoIP penetration forecasts.

Key questions

Is there potential to turn managed VoIP into increased data revenue?
Why does Skype dominate the independent "cheap calls" VoIP market and how do the other players compete?
Does Google, the latest big name to enter the mobile VoIP fray, have the potential to disrupt the market?
How are mobile operators reacting: fighting or cooperating?
How and when will operators start to offer managed mobile VoIP?
What are the key factors for enabling mobile VoIP to take off?
What will mobile VoIP user penetration levels be up to 2014?

Ribbit Announces Cloud-Based Service Ribbit Mobile

VoIP Monitor — Tue, 03 Nov 2009 18:45:36 GMT

Ribbit announces the release of Ribbit Mobile, a cloud-based service that provides a new way to manage calls, messages and phones. Ribbit Mobile's core set of features - including voicemail transcription, smart call routing and Web-based calling - transforms the way people communicate by letting them decide how, when and where they want to stay in touch.

Ribbit Mobile lets users communicate in ways never before possible by enabling them to merge multiple phones, route mobile calls to other phones and even answer their mobile calls on the Internet. Ribbit Mobile users can stay connected on their own terms-wherever they are-so they never miss another important call.

Unlike other Internet voice services, Ribbit Mobile doesn't require a new phone number, a new phone or a software download-and the service can be made active on any mobile phone in less than five minutes. Ribbit Mobile also converts voicemail to text and sends it to users via SMS and email, eliminating the need to dial into voicemail and listen to messages. It also provides an online message inbox where messages can be managed and stored forever.

Ribbit Mobile also offers users a back-up phone online for those times when a users' device is lost, stolen, broken or mobile service is unavailable. The Web-based version of the mobile device enables Ribbit users to continue to make and take "mobile" phone calls via an Internet connection, regardless of their location or device accessibility.

Service Increases Mobile Professional Productivity
For mobile professionals dependent on their phones to close deals and win new business, Ribbit Mobile is a must-have service. "Ribbit Mobile gives me a competitive advantage and helps me close deals faster," said Mortgage Loan Officer Jerry McClain. "I receive over 30 voicemail messages a day, and Ribbit Mobile gives me instant visibility into which ones are important so I can focus on what matters most."

Open Platform Drives Telephony Innovation
Ribbit Mobile is built on the Ribbit Platform - a robust, extensible platform with open APIs that lets developers add features to Ribbit Mobile. As more developers create applications, the service will become increasingly powerful and innovative.

Ribbit is also announcing the "Developer Rewards" program which will compensate developers based on how much their applications are used, even if their applications are offered for free to end users. Unlike traditional app stores, Ribbit developers are rewarded on creating the most useful and popular applications that drive platform and network usage.

During the introductory beta period, Ribbit is making its "Professional" package free for everyone on a trial basis, available via download at http://www.ribbit.com/mobile.

Aruba Network Certifies the Wireless snom 800 Series VoIP Desk Phones

VoIP Monitor — Mon, 02 Nov 2009 17:16:30 GMT

Aruba Networks and snom announce that the wireless snom 820 and 870 VoIP Desktop Phones have been certified interoperable with multi-use Aruba 802.11n wireless LANs. Multi-use networks share wireless infrastructure among different voice, data, and video applications, and are increasingly popular because they eliminate redundancy, enhance user mobility, and significantly lower deployment and operating costs. Cost savings come in part from scaling down -- or rightsizing -- expensive wired Ethernet infrastructure in favor of low-cost, more energy-efficient Wi-Fi networks. Desk phones have traditionally been tethered to wired Ethernet connections, but the new wireless snom phones allow these connections to be rightsized, too.

Network rightsizing is a three step process in which the utilization of existing wired Ethernet infrastructure is assessed, unused or underutilized ports and switches are consolidated to lower recurring costs and energy usage, and mobile users and devices are migrated to 802.11n Wi-Fi networks. The cost savings and carbon footprint reduction obtained from network rightsizing have been demonstrated in projects large and small. Migrating desk phones from wired to wireless connections promises to boost rightsizing savings even further, while delivering the freedom to locate and move phones at will.

Industry analysts report an advancing wave of SIP-capable equipment, and the technology is the clear choice for new phone deployments. Major UC/PBX vendors now offer SIP connectivity on the station side, avoiding the need to implement multiple proprietary signaling protocols.

The snom 820 VoIP phone has been compliance-tested by Avaya for compatibility with Avaya IP Office 5.0, and features a large high-resolution TFT color display, integrated XML browser, speakerphone with interference noise suppression, secure VPN, 5-way conferencing, multiple ringtones, and up to twelve different SIP identities. The 870 VoIP phone includes a larger touch-screen display, larger address book, and additional security features.

Aruba uses wireless snom phones in its Sunnyvale HQ facility and new Rockefeller Center offices in New York City. The latter is a completely wireless VoIP deployment anchored by an Avaya IP-PBX in Sunnyvale.

Esnatech Introduces Desktop Video Integration with Skype Desktop VoIP Software

VoIP Monitor — Mon, 02 Nov 2009 16:13:31 GMT

Esna Technologies has added Skype VoIP/Video support to its market leading unified communication desktop software, UC Link. As part of its ongoing goal for interoperability and Rich communication and collaboration tools, Esnatech has integrated its UC Link desktop software with Skype VoIP and Video services for enhanced real-time communication services. Users of the Esnatech Telephony Office-LinX UC platform will have their live Skype VoIP & Video status tied into existing office infrastructure and applications so other users can see if they are available or not, on their office telephone, or off the network. It will bridge the Skype VoIP/Video world with Enterprise telephony and Groupware status.

The benefits to enterprise users are they can integrate their daily business communications with Skype VoIP & Video and start using the popular application for daily business use. Co-workers and customers will know their true availability and status and using the Esnatech UC link desktop software communicate in the most efficient and collaborative manner. Users can in one click initiate Live Secure IM, Live desktop to phone voice calls, Ad Hoc group conference session and Live Skype VoIP and Video Session.

Skype support will be part of the upcoming UC Link desktop software for the esnatech UC server Telephony Office-LinX expected in Q1 2010. Esnatech will commence beta trials in December/January with its partners.

UC Link desktop software is a rich multi-OS desktop unified communications software that delivers Rich Presence, Mobility, messaging for the Telephony Office-LinX unified communications platform. This Esnatech UC server is already tightly integrated with all major PBX's and VoIP services as well it provides Mobile UC services to RIM Blackberry, Apple iPhone, Google Android, Windows Mobile & Symbian devices. It can be deployed as a standalone UC solution or an integrated application to existing PBX's and messaging solutions such as Octel voice mail platforms.

Smoothstone SIP Trunking Application Now Rated 'Avaya Compliant'

VoIP Monitor — Mon, 02 Nov 2009 15:36:20 GMT

Smoothstone announced that its SIP Trunking Service is compliant with key Internet protocol telephony solutions from Avaya. Smoothstone's SIP Trunking service helps businesses maximize IP-PBX functionality by accessing the rich feature set of Smoothstone's VoIP services, delivering reliable, secure and cost-effective connections that provide greater redundancy and higher aggregation of voice traffic (compared to T1/PRI), without costly equipment upgrades. The application now is compliance-tested by Avaya for compatibility with Avaya Aura Communication Manager and Avaya Aura(TM) SIP Enablement Services.

Smoothstone is a member of the Avaya DevConnect program -- an initiative to develop, market and sell innovative third-party products that interoperate with Avaya technology and extend the value of a company's investment in its network.

As a Gold member of the program, Smoothstone is eligible to submit products for compatibility testing by the Avaya Solution Interoperability and Test Lab. There a team of Avaya engineers develops a comprehensive test plan for each application to verify whether it is Avaya compliant. Doing so ensures businesses can confidently add best-in-class capabilities to their network without having to replace their existing infrastructure -- speeding deployment of new applications and reducing both network complexity and implementation costs.

Save an Additional $200 Off VoiceCon San Francisco – November 2-5, 2009

VoIP Monitor — Sun, 01 Nov 2009 20:36:40 GMT

Make Smart Decisions in this Tough Environment

The industry is being shaken up — Consolidation means some companies will leave the field as new ones enter. There are emerging technologies and architectures.

The fate of a huge number of products and services is uncertain, and therefore so is the fate of the networks that rely on them.

VoiceCon San Francisco will deliver the insight you need to make the critical decisions that will shape your enterprise communications platform, systems, applications and services.

Conference Tracks Include:

Buyer’s Guide
Next-Gen Communications Technologies
Wireless/Mobile Communications
Planning & Implementing IP Telephony and Converged Networks
Managing Staff Technologies & Cost
Unified Communications

Keynote: Dr. Alan Baratz, Senior Vice President;President, Global Communication Solutions, Avaya

See the complete conference program and session descriptions.

VoiceCon San Francisco Exhibition is THE place to see the equipment and software suppliers and the service providers serving the enterprise communications market in one location. Check out our sponsors and exhibitors.

Special offer for Voip Monitor readers – take an additional $200 off the current discounted price or register for a FREE Expo Plus pass at www.voicecon.com/sf. Use VIP Code: CNECVC10

States Approve Frontier’s Acquisition of Verizon Wireline Operations

VoIP Monitor — Fri, 30 Oct 2009 15:44:22 GMT

Frontier Communications announced its pending acquisition of Verizon Communications’ local wireline operations has received approvals from the California Public Utilities Commission, the Public Utilities Commission of Nevada, and the Public Service Commission of South Carolina. The transaction, announced May 13, 2009, includes Verizon’s local exchange businesses in 14 states, including parts of California, and certain customer relationships for long distance services, broadband Internet access and broadband video.

This week, on October 27, 2009, Frontier’s stockholders voted overwhelmingly to approve the merger agreement and related proposals.

Frontier has also received approvals from 10 of the 41 FiOS video franchise communities the company will serve in Washington state and Oregon. On September 1, 2009, the transaction received early termination of the waiting period required under the Hart-Scott-Rodino Antitrust Improvements Act of 1976.

In addition to the remaining local franchise approvals and the approvals of six other states, the Federal Communications Commission (FCC) must approve certain license transfers as well. The FCC review is in process.

After the transaction, Frontier will have approximately 7 million access lines in 27 states, 8.6 million voice and broadband connections, and approximately 16,000 employees, based on data as of December 31, 2008. The transaction is still expected to close during the second quarter of 2010.

Report: VoIP Services Bring in $21 Billion for Service Providers in 1H09; Strong Demand Continues

VoIP Monitor — Thu, 29 Oct 2009 19:31:49 GMT

Market research firm Infonetics Research released its biannual VoIP and UC Services and Subscribers report, which tracks business and residential/SOHO voice over IP services and includes a North America Business VoIP Services Leadership Matrix that analyzes and ranks the top service providers in the VoIP business services market.

ANALYST NOTE

"Demand for residential and business VoIP services continues to grow even as spending in other communication areas tightens. For the first half of 2009, the worldwide VoIP services market grew to $20.7 billion. Residential VoIP services remain healthy, comprising the majority of worldwide VoIP services revenue, and subscribers are up 14% from the end of 2008. On the business VoIP side, while managed IP PBX revenue growth has slowed in line with IP PBX shipments, we are expecting IP Centrex and hosted UC service revenue to grow 26% year-over-year in 2009," explains Diane Myers, directing analyst for service provider VoIP and IMS at Infonetics Research.

REPORT HIGHLIGHTS

IP connectivity services currently make up about a third of total VoIP business service revenue, growing to 40% of the total by 2013 (managed IP PBX services and hosted VoIP and UC services make up the balance)
The current sweet spot of the North American IP Centrex market is small business (those with fewer than 100 employees)
While the largest VoIP services opportunities are in North America and EMEA (Europe, Middle East, Africa), demand for VoIP services is growing fast in Asia Pacific and Central and Latin America
For the first half of 2009, Japan's NTT, France Telecom, and Comcast in North America retain their leadership as the world's largest residential VoIP service providers, together holding nearly 20% of the world's VoIP subscribers
The number of residential/SOHO VoIP subscribers is forecast to top 225 million by 2013

REPORT SYNOPSIS

Infonetics' VoIP and UC services report provides worldwide and regional market size, forecasts, and analysis of residential VoIP services, business VoIP and UC services, and residential/SOHO VoIP subscribers. Business voice services tracked include managed IP PBX services, hosted VoIP and unified communication services (IP Centrex by business size, UC applications, UC telephony), and IP connectivity services (VoIP VPN/IP access, SIP trunking, IP integrated access).

The report includes regional consumer VoIP subscriber market share, tracking AT&T, BT, Cablevision, Charter, Comcast, Cox, Embratel NetFone, France Télécom, Free, KDDI, LG Dacom, LibertyGlobal, neuf cegetel, NTT, ONO, PCCW, Portugal Telecom, Rogers, SK Broadband, SoftBank, Tele2 Netherlands, Telecom Italia, Time Warner Cable, Vonage, and many others.

The report includes a VoIP Service Provider Leadership Matrix, which provides rankings of service providers currently leading the North American market and best positioned to succeed long-term. The leadership matrix ranks the top 10 IP Centrex service providers and top 9 IP connectivity providers based on criteria such as number of IP Centrex subscribers, installed base of IP connectivity lines/trunks, financial stability, market strategy, service capabilities, and support options.

BT Brings Unlimited Calls to Businesses

VoIP Monitor — Thu, 29 Oct 2009 17:57:21 GMT

BT announces it is offering unlimited fixed line calling packages for small businesses. The new high value BT Business One Plan Inclusive packages offer unlimited calling minutes to UK landlines, UK mobiles, international destinations and 0845 and 0870 numbers for a fixed fee*.

BT Business is also offering businesses greater savings by introducing 500 free UK calling minutes per calendar month per line (fixed or VoIP).**

Businesses opting for one of the new BT Business One Plan Inclusive packages can also combine unlimited calls with a range of communications options including broadband and mobile, consolidating their fixed, mobile and broadband spend under One Plan.

Under BT Business One Plan all service and billing enquiries are dealt with through one number, and customers will receive one bill for all services. BT Business Broadband offers a 99.99% reliable broadband service; nine out of 10 faults are resolved in under four hours and 95% of calls to the 24/7 UK help desk are answered within 30 seconds.

*Available to customers that (re)sign to OnePlan by 30.04.10. Minimum yearly spend and requires commitment to 2 core products. Fee - £15 per month - further reduced to £12.50 per month if customers take the unlimited package on a VoIP line. Fair usage policy applies.

**Offer ends 30.04.10. Exclusions, terms and conditions apply.

Voxbone's iNum Service Speaks in High Definition with Skype

VoIP Monitor — Wed, 28 Oct 2009 21:18:13 GMT

Voxbone announces that its geographically-independent number service, iNum, now supports HD calling between dozens of VoIP networks and Skype, the global leader in Internet communications.

Greatly expanding availability of emerging HD voice technology, Voxbone will transcode, or translate, between Skype's super-wideband SILK codec and the widely deployed HD codec, G.722, Voxbone announced at eComm Europe 2009.

Prefaced with the ITU-assigned 883 code, iNum numbers refer to the Internet as the 44 country code refers to the U.K. Calls to iNum numbers are routed first to Voxbone, which delivers them to the appropriate service provider over IP.

Because iNum numbers can be dialed from PSTN phones as well as IP endpoints, Voxbone's move enables conference calls with attendees on conventional phones, who will hear within the PSTN's audio constraints, and others with high-definition IP endpoints, who will enjoy the richer sound of HD calling.

Touchfon Introduces Residential Phone Service at $15.99

VoIP Monitor — Wed, 28 Oct 2009 20:38:19 GMT

Touchfon International is pleased to announce the addition of TouchfonLive VoIP Phone Service to its comprehensive suite of communication products. TouchfonLive VoIP Phone Service is available to residential and small business users, as well users in many foreign countries. TouchfonLive uses customer's high-speed internet connection to provide clear, crisp and cost saving calling to anyone, anywhere. Touchfonlive, which includes features such as Caller ID, Call Waiting, Call Forwarding, and more allows customers to keep their current number and phone equipment, while cutting their monthly costs in half. TouchfonLive includes unlimited long distance to the United States, Canada and Puerto Rico. International calling plans are available starting at just $.02/minute to help alleviate long distance costs associated with most phone services.

With the release of TouchfonLive VoIP Phone Service, Touchfon is offering the Standard Residential Phone package at special introductory rate of just $15.99 per month. There are no contracts or commitments. Customers who join before November 30, 2009 qualify for the introductory price, which is guaranteed for the life of the account.

VocalTec’s VoIP Solutions Power Far East Telecom’s VoIP Network

VoIP Monitor — Wed, 28 Oct 2009 16:59:46 GMT

VocalTec announces that Far East Telecom has selected VocalTec as the provider for its core Class-5 and Class-4 VoIP network. The project will be implemented in conjunction with Inline Telecom, a leading Systems Integrator in Russia.

Far East Telecom is transforming its network to a core IP network. This deployment represents the first stage of Far East Telecom’s migration plan, including the deployment of VocalTec’s Essentra Class-5 and Class-4 VoIP solutions across five distributed locations with initial support for tens of thousands of subscribers. The project will provide for a seamless transition and will enable Far East Telecom to offer a wide range of innovative subscriber services, as well as efficiently managing the interconnection to its existing traditional telephony network.

VocalTec’s SIP-based VoIP solution will enable greater capacities and efficiencies and is designed to support Far East Telecom’s future network evolution to hundreds of thousands of subscribers; VocalTec’s comprehensive certification by the Russian Ministry of Communications further supports Far East Telecom’s requirements and service needs.

V Phone: Mobile & Wireless VoIP on the Apple iPhone

VoIP Monitor — Wed, 28 Oct 2009 16:56:26 GMT

The V Phone offers a key advantage that is sure to grab the attention of users of other mobile VoIP software. Rather than locking iPhone users into working with a single VoIP provider, it lets users choose between multiple providers of their choice for voice calling. The V Phone nullifies the single provider protocol and lets users choose the most cost-effective provider for each task, a capability that is unique among such programs. In addition, the V Phone app is the first ‘open and configurable’ app to also offer SMS messaging over IP.

The V Phone’s SMS features offers two standard VoIP provider networks, as well as an option for entering the user's own provider of choice. The app’s VoIP calling and text messaging options works with WiFi, allowing for convenient coverage without the need for a SIM card. Additionally, iPhone users can now make 3G/GPRS/HSDPA VoIP calls. With an intuitive, attractive graphical user interface and simple installation steps, the V Phone makes it easy for anyone to save money and start getting more out of their mobile or cell phones.

Matrix has released a video on You Tube demonstrating to iPhone users on how easy it is to configure and use the app.

Matrix has submitted the app to Apple for approval and believes it should be available in the App Store within 2 to 3 weeks.

Philly and New York Service Providers Set MLB World Series Wager

VoIP Monitor — Wed, 28 Oct 2009 16:38:31 GMT

As the New York Yankees take on the reigning champion Philadelphia Phillies in the 2009 Major League Baseball World Series, there's more on the line for two unified communications providers. After last year's World Series bet with a Tampa-based company (Oranges for Cheesesteaks), Philadelphia-based Alteva, North America's largest enterprise hosted VoIP and unified communications solutions provider, has initiated another friendly competition with Stage 2 Networks, New York's leader in hosted PBX and converged network solutions.

This year, Alteva decided to up the ante by adding a humiliation factor. The bet is Philadelphia pretzels for New York bagels, but the 2009 loser will also don opposing teams fan wear and have a photo session with a banner acknowledging the winner in front of well known landmarks. If the Phillies win, Alteva has chosen to have Stage 2 take their "photo of shame" in Times Square. If the Yankees win, Stage 2 has selected the Rocky statue in front of the Philadelphia Museum of Art.

"After winning bushels of Florida oranges last year, we bought a juicer and got accustomed to drinking fresh squeezed orange juice at breakfast meetings... the only thing missing is a good bagel," said William Bumbernick, CEO of Alteva. "We enjoy doing things like this as its fun for the company, overall morale and we get to show our support of our home team. Alteva is proud to be part of the Philadelphia business community, and we have an avid group of Phillies fans here."

"While this is a great way to show support for our home teams, it also demonstrates how two companies who have a common business goal can use competitiveness in a fun and spirited way," said Joseph P. Gillette, CEO of Stage 2 Networks. "The only previous Yankees-Phillies World Series was played back in 1950, so we're looking forward to going head-to-head almost 60 years later. Who knows when we'll get this opportunity again? We're also looking forward to eating lots of Philly pretzels."

Rebtel Introduces VoIP Services Between Pakistan and 50 Countries

VoIP Monitor — Wed, 28 Oct 2009 16:32:20 GMT

Pakistanis living in 50 countries around the world can now call friends and family in Pakistan free of charge using any phone - landline or mobile - with the introduction of Rebtel service in Pakistan.

The same goes for people in Pakistan who can now use their landline or mobile phones to call friends and family in 50 other countries free of charge, or make ultra-cheap calls to anywhere else in the world using Rebtel, the Internet phone company.

There is no charge to set up a Rebtel account. No contract. No monthly fees. And no changes whatsoever are required to the users calling plan with their landline or mobile phone provider.

How Rebtel works

Rebtel gives people local phone numbers in the country where they live that connect them directly with their friends and family abroad. That means, for example, Rebtel gives someone living in Lahore a local Lahore phone number that connects him directly to a friend or family member living in New York. At the same time Rebtel gives the friend or family member in New York a local New York phone number that connects them directly to their friend or family member in Lahore.

There's no access number to dial, no PIN to remember, or long international phone number to enter. The user only dials a local phone number where they live and Rebtel connects the call over the Internet to their loved one abroad.

Free Call: Tricky but worth it
Rebtel's standard rates to and from Pakistan are incredibly low - almost always better than calling card rates.

All calls - to landlines or mobile phones anywhere in Pakistan - from any of the 50 other countries where Rebtel is available are just 7.4 cents (U.S.) per minute.

And calls from Pakistan are equally cheap. For example, a call from Pakistan to:

Saudi Arabia is 8.23 PKR per minute on a landline, 16.30 PKR on a mobile phone
U.K. is 1.58 PKR per minute on a landline, 14.97 PKR on a mobile phone
UAE is 16.55 PKR per minute on a landline, 17.55 PKR on a mobile phone
U.S. is 1.24 PKR per minute on either a landline or mobile phone

But if those rates are still too high, Rebtel recommends Pakistanis use the Free Call service and get their international call for the cost of making a local call.

Here's how to make a Free Call: When a person in Karachi calls a friend in London, for example, they ask their friend - while they stay on the line in Karachi - to hang up in London and call right back using the local London phone number they've been given by Rebtel to reach their friend in Karachi.

When that happens:

The person in Karachi is making a local call
Their friend in London is making a local call
Rebtel connects the two local calls over the Internet, and
The international minutes are free.

Free Calls (when the person who receives the call hangs up and calls back while the original caller stays on the line) today can be made between 51 countries around the world.

Getting started with Rebtel is fast, easy and free

1. Go to http://www.rebtel.com, enter your number and a friend's number abroad.
2. Rebtel creates a local number you can use from any phone, anywhere.
3. Save the number in your mobile phone's address book and use it any time.

Panasonic Announces Interoperability of Its SIP Cordless Phones with Broadsoft's Broadworks Platform

VoIP Monitor — Tue, 27 Oct 2009 15:51:53 GMT

Panasonic announces the interoperability of BroadSoft's BroadWorks platform with the KX-TGP500 series of SIP DECT cordless telephones, which offers outstanding voice quality and a range of productivity-boosting capabilities in a full-featured desk phone replacement. DECT 6.0 ensures no interference with wireless networks, and the convenient cordless design eliminates the need to run dedicated network wiring to each employee workstation. It is ideal for home and business environments.

The Panasonic TGP500 series phones debut at BroadSoft Connections 2009: Voice & Vision, October 25-28 in Scottsdale, where Panasonic is a Platinum Sponsor. This is Panasonic's second year sponsoring BroadSoft's annual users' conference, validating the company's continued commitment to its partnership with BroadSoft.

With flexible configuration options, it has never been easier to deploy and expand a SIP-based phone system. The benefits of SIP are especially compelling in today's business environment, where every dollar counts. The reduced hardware costs and simplicity of routing calls over an Internet connection can add up to huge savings on monthly telephone bills. With the Panasonic KX-TGP500 series SIP phone system, it is quick and easy to add up to six cordless handsets -- each with its own number. All that is needed is a single Internet connection and an electrical outlet near the location of the handset. Because it employs DECT technology, the connection is secure and the sound crystal-clear. Panasonic's SIP DECT phones use 100 percent recycled packaging materials, and all new models are Energy Star® qualified, which means they use about one-third less energy than non-qualified models.

Compatibility of the TGP500 series phones include:

BroadSoft BroadWorks VoIP application platform
CAT-iq
Asterisk
IETF SIP version 2 (RFC3261 and companion RFCs)

GP500 Series Details

KX-TGP500

The system features a wall-mountable base unit and one cordless handset. It is expandable up to 6 DECT 6.0 cordless handsets and supports up to 8 phone numbers and 3 simultaneous calls. It boasts Wide Band Audio (G.722) and 5 hours Talk Time, 5 days Standby. Its elegant design features a white backlit large LCD on the handset and a Handset Call Button on the base unit. It also has a handset speakerphone, 2.5mm headset jack and belt clip. MSRP $199.95

KX-TGP550

The KX-TGP550 has all the features and benefits of the KX-TGP500 and adds a corded base unit with a large white backlit LCD and 5 hours Talk Time, 5 days Standby, plus a hands-free speakerphone, Handset Call Button on the base unit, and one-touch call transfer with Busy Lamp Indication. MSRP $329.95

KX-TPA50

The TGP500 systems can be expanded up to a total of 6 cordless handsets by adding the KX-TPA50 cordless handset. MSRP $99.95