VoIP Monitor

Apple Gives Green Light to Mobile VoIP iPhone App from Rebtel

VoIP Monitor — Fri, 20 Nov 2009 16:33:27 GMT

Rebtel lets iPhone owners make international calls to any phone, anywhere in the world for just pennies per minute without having to search for a Wi-Fi hot spot. Free calls are possible between the 53 countries where Rebtel is available.

Using the new Rebtel iPhone app is super easy and fast. Just open the app, select the phone number of an international friend from your usual iPhone address book or type in their number, tap the Call with Rebtel button, and Rebtel takes over from there to connect the call instantly.

Celebrate Thanksgiving with Free Calls from Truphone

VoIP Monitor — Thu, 19 Nov 2009 18:24:43 GMT

Truphone believes in the importance of staying in touch with friends and family, especially during the holiday season. That’s why for 12 hours this Thanksgiving (Thursday November 26th, between midday and midnight GMT), all calls to a US landline or mobile phone made using Truphone will be completely free!

It is open to all new and current Truphone users, anywhere in the world, on all Truphone applications, and you don't need to sign up to anything or change your settings to make free calls.

American Airlines Launches Wi-Fi Widget, However Cell Phone and VoIP Use Are Prohibited

VoIP Monitor — Thu, 19 Nov 2009 18:16:54 GMT

American Airlines is launching its new Wi-Fi Widget that enables customers to identify Wi-Fi equipped flights 24 hours prior to departure. Customers can visit aa.com/wifiwidget to find and use the Wi-Fi Widget. The tool allows travelers to determine if their American Airlines flight will be scheduled with Wi-Fi enabled aircraft. Customers can also embed the Wi-Fi Widget in their own blogs, Web sites, and social network sites, which makes finding Wi-Fi on American Airlines flights even easier.

Simply enter your flight number or select your flight's departure airport from a drop down menu to find out in seconds if your flight is scheduled on a Wi-Fi equipped aircraft. For a quick video demonstration of how to use the Wi-Fi Widget.

Recently American completed the installation of Wi-Fi on 150 MD-80 aircraft. Combined with the 15 Boeing 767-200 aircraft that also have the service, American Airlines fleet now has 165 planes that are capable of providing customers with the ability to surf the Web, check e-mail, send instant messages, and more on a greater number of flights. In August 2008, American became the first U.S. airline to launch the Gogo service. Since then, thousands of customers traveling on American aircraft have enjoyed onboard Wi-Fi service while flying throughout the United States.

Coming soon, customers will also be able to determine if their flight has Wi-Fi by looking for the Wi-Fi symbol on the AA electronic gate information display screens at the airport.

Gogo service turns an American Airlines flight into a Wi-Fi hotspot. Once the aircraft has reached 10,000 feet, users can simply turn on their Wi-Fi enabled devices - such as laptops, smartphones and handheld PDAs - then open their browsers and be directed to the Gogo portal page where they sign up and begin surfing. Gogo also provides complimentary access to all that AA.com has to offer, including booking flights, hotel stays and rentals, checking AAdvantage accounts, and more.

Once connected, passengers can surf the Web, check e-mail, send instant messages, access a corporate VPN, and even start holiday shopping with SkyMall. By placing an online SkyMall order onboard, customers will receive a $12.95 discount on orders totaling more than $100 - covering the price of the Gogo session.

Each paid Gogo session includes full Internet, e-mail and VPN access. Cell phone and VoIP service use are prohibited.

Digium Adds Affiliate Partnership to Authorized Reseller Program

VoIP Monitor — Wed, 18 Nov 2009 18:52:12 GMT

Digium has added an Affiliate partnership level to the company’s Authorized Reseller Partner Program. This new Affiliate level provides an entry-level tier for resellers who are interested in selling Digium Unified Communications and IP Telephony solutions, without a required minimum annual revenue commitment.

Affiliate partners will have access to Digium’s ongoing partner communications, as well as the partner portal, which includes marketing toolkits, videos and sales and technical training. As partners migrate to the higher levels in the program, Registered and Select, Digium provides greater benefits, including discounts, demo kits, sales and marketing support and enhanced technical and sales training. This tiered approach is designed to help partners grow their businesses with Digium’s Switchvox VoIP phone system and Asterisk custom telephony solutions.

Digium is actively recruiting new partners to join its program. More details can be found at www.digium.com/ucpartner.

fring Now Offering VoIP Support on All Android Devices

VoIP Monitor — Wed, 18 Nov 2009 18:27:23 GMT

fring announces the world’s first full voice over internet support on all Android devices. Android users join millions of fringsters worldwide with free calls, live chat, contact availability status, file sharing and more with friends on fring and several popular social services including Skype, MSN, GoogleTalk, ICQ, Twitter, Yahoo! and AIM. Centralized from one integrated contact list, fringsters also make calls to regular phone lines via Skype-Out and hundreds of popular SIP services over 3G, GPRS or Wi-Fi internet connection.

Avi Shechter, Co-Founder & CEO of fring said, “fring is glad to bring Android users the first free voice calls over fring, Skype, GoogleTalk and SIP, realizing the real potential of their new open, internet-enabled devices.” Shechter continued, “Now Android users can join the millions of fringsters on thousands of fring-supported mobile devices who enjoy the rich features and cost benefits that mobile communication over internet can provide: communicating and sharing social Internet experiences over Wi-Fi or data plan with their friends wherever they are, on the web or on their phones.”

In addition to Android, fring is supported worldwide on thousands of Symbian S60, iPhone, iPod touch, Windows Mobile, J2ME and Linux devices.

The brand-new Android version can be downloaded via fring’s WAP site at m.fring.com or directly via the Android Market.

CosmoCom Announces SIP Trunking Certifications from AT&T, Global Crossing and XO Communications

VoIP Monitor — Tue, 17 Nov 2009 18:08:05 GMT

CosmoCom has recently been certified for direct SIP Trunking interoperability with AT&T, Global Crossing, and XO Communications. As a pioneer in IP Telephony, CosmoCom has been deployed in dozens of carriers' VoIP networks. The native VoIP architecture of CosmoCall Universe has enabled CosmoCom's IP-based contact center solution to easily integrate with these carriers' VoIP infrastructure.

As carriers roll out their VoIP services, they have added SIP Trunking to their service offerings to allow businesses to directly connect corporate VoIP systems to the carriers' networks. SIP Trunking enables corporations to benefit from the advantages of IP-based contact center solutions without having to also deploy VoIP gateways on the corporate premises. Such a solution greatly reduces both hardware and IT costs as well as the technology expertise required for the corporations deploying VoIP solutions.

Junction Networks Offers Alternative to Blocking Numbers in Response to ''Traffic Pumping'' Practices

VoIP Monitor — Tue, 17 Nov 2009 17:10:36 GMT

Free conference calling services, adult chat lines and other "traffic pumping" services are often reached through the telephone exchanges of very small, rural operators. In a legal but questionable arbitrage scheme, these calling services choose these rural exchanges precisely for their high termination charges -- the fees that sending carriers pay them to complete (terminate) the calls. Charging as much as twenty times the typical domestic termination rate, the rural telco then splits the profits with the service. While carriers have responded by blocking calls to those numbers, Junction Networks has taken another tack, announcing that it will begin charging a higher fee for outbound calls to those exchanges.

The move allows Junction Networks' customers to continue using these services at their discretion. They have the option to control access to any call costing more than 2.9 cents per minute by simply completing an online extended dialing form.

"Free conference calling and other 'traffic pumping' services exist because the current carrier compensation system allows rural carriers to pass extremely high fees on to other carriers, who often cannot come close to recovering the cost of calls," said Rob Wolpov, president, Junction Networks. "As a result, we have been left with an overwhelming increase in fees for calls to a number of rural locations where these services operate."

"In order to maintain our low-cost business VoIP options and at the same time, allow our customers to call any number they choose, we have decided to charge the market rate for calls to the designated areas used by these services."

"We've all heard the saying, 'There's no such thing as a free lunch.' Well, Junction Networks has chosen to take the approach of providing access, but also has chosen not to lose money by providing it," said Andy Abramson, author of VoIPWatch, a widely read industry site that covers all things IP communications. "Each side gets what it wants, and as the carrier, Junction Networks eliminates the subsidization that had been at play. I think it's a model more access providers

Phone Power Adds Free International Calling to VOIP Phone Service

VoIP Monitor — Tue, 17 Nov 2009 16:58:04 GMT

Phone Power announces that their VOIP phone service is now offering 60 minutes of free international calling to 88 countries every month, including major calling destinations like Mexico, China, India, and most of Europe. This program extends to all new and existing customers on current residential and business plans. Existing customers will not need to do anything to activate this feature. Phone Power offers unlimited local and long distance calling starting at $14.95 per month.

Official: Google Acquires VoIP Firm Gizmo5

VoIP Monitor — Mon, 16 Nov 2009 21:04:52 GMT

Google said on Thursday that it has acquired California-based VoIP company Gizmo5 for an undisclosed sum. Reports said that the purchase price was approximately USD30m.

This acquisition will further broaden Google's range of communication services, adding technologies for voice, text and video communications.

Google confirmed that Gizmo5's engineers will be joining the Google Voice team "to continue improving the Google Voice and Gizmo5 experience".

Current Gizmo5 users will still be able to use the service but no new customers will be signed up for the time being.

Zultys Empowers Voice Solutions Through Internet Telephony Service Providers

VoIP Monitor — Mon, 16 Nov 2009 18:46:11 GMT

Zultys has announced the launch of their Zultys ONE SIP bundles. These bundles provide everything a customer needs to have a full Unified Communications suite of applications running over SIP trunks. Customers can also leverage this deployment as a means of failover or to obtain numbers from different markets.

SIP, being the leading standard for VoIP Communications, is now the choice of all customers who want high availability, increased productivity and reduced costs. To accelerate and launch this deployment, Zultys is working with multiple premier ITSPs. To that end, Zultys has developed solution packages that include the ability to deploy 60 SIP trunks with each of its MX250 SIP Packages called: SIP/20, SIP/50, SIP/100, SIP/200, and pre-licensed with all the software and hardware needed to rapidly deploy a pure SIP solution -- just add phones!

"Our objective was to enable the activation of SIP connectivity and trunking in a simple, elegant and cost efficient manner," said Neil Lichtman, Zultys CEO. "Zultys delivers a standard interface to activate the service with multiple ITSP providers in a matter of minutes not hours." Making communications cost effective, Zultys offers compelling bundled SIP trunk pricing that leverages the 60 free SIP licenses included with every bundle.

All the power of SIP in the elegant simplicity of a single purpose built server with no additional licenses or hardware to purchase: The Zultys difference!

Android Mobile Application Developers Can Now Build VoIP-Enabled Clients

VoIP Monitor — Mon, 16 Nov 2009 18:03:20 GMT

Global IP Solutions is enabling Android mobile application developers to build first-rate VoIP-enabled clients with GIPS VoiceEngine Mobile. Nimbuzz will be the first customer to offer HD voice (VoIP) on Android phones.

“GIPS VoiceEngine allows Android developers to build VoIP-enabled applications that offer HD voice, while tackling all the typical IP network issues - such as delay, jitter, packet loss, bandwidth constraints, noise and echo,” said Roar Hagen, GIPS’ Chief Technology Officer. “Today, mobile users desire a smartphone that offers a unique experience, which includes the ability to have great quality communication. With GIPS engineering expertise, developers can quickly and effectively build their applications enabling them to concentrate on their core business.”

Hagen added, “We’re thrilled that Nimbuzz will be GIPS’ first customer to deploy their application on Android mobile phones. Nimbuzz continues to offer their users a distinctive unified social messaging application that connects popular social and instant messenger networks into one simple, user-friendly offering.”

Evert-Jaap Lugt, CEO of Nimbuzz said, “Nimbuzz is delighted to offer free mobile VoIP calls to Android users. VoIP is fast becoming established as a mainstream standard for voice calling and will represent a major part of the mobile eco system of the future. With the addition of high-quality HD Voice to our free IM and chat functions, thousands of new and existing Nimbuzz users on Android will now be able to enjoy a complete communications and messaging service in one.”

With nearly 3 billion users worldwide, the mobile phone has become the most personal and ubiquitous communications device. Research firm Gartner has predicted that Android will become the second most popular smartphone by 2012 with 14.5 percent market share and iPhone with 13.7 percent market share.

Voxeo Corporation Announces Strategic Investment

VoIP Monitor — Fri, 13 Nov 2009 16:12:37 GMT

Voxeo Corporation announces a $9 million minority investment by North Atlantic Capital and the Florida Growth Fund. Voxeo will use the funds to take advantage of rapidly growing opportunities in Unlocked Communications, including VoIP platforms, Interactive Voice Response, text and instant messaging, and unified communications, both by accelerating internal development and through corporate acquisitions. Voxeo has acquired three companies over the last 14 months.

This is the first investment made by the Florida Growth Fund, a $250 million partnership between the State Board of Administration of Florida and Hamilton Lane. Hamilton Lane is one of the leading private equity asset managers worldwide, with over $88 billion under management and supervision. The Florida Growth Fund was created in 2009 as a result of the passage of Florida State Bill 2310 with the goal of investing in technology and growth-based businesses in the State of Florida.

North Atlantic Capital is a venture capital firm dedicated to helping strong teams accelerate the growth of their expansion-stage, technology-based companies. North Atlantic Capital found and identified Voxeo as a high quality investment opportunity in 2007 and in addition to leading the investment, introduced Voxeo and the Florida Growth Fund. Concurrent with the investment, Mark Morrissette, Managing Director of North Atlantic Capital, has joined Voxeo’s board of directors.

Voxeo is an employee-owned company with headquarters in Orlando, Florida and offices in Beijing, Cologne, London and Silicon Valley. In 2009, Voxeo was named one of the 100 best places to work in Florida. Voxeo's offerings include:

Prophecy Interactive Voice Response and Unified Self Service platform helps enterprises and service providers automate their most common customer and employee interactions via phone, SMS, IM, Twitter, mobile web and more.
VoiceObjects Application Server helps enterprises and service providers design, create, deploy, manage, and analyze personalized self-service applications.
Micromethod VoIP platforms and solutions help enterprises and service providers deploy highly-scalable, customizable infrastructure for VoIP solutions.
IMified technology delivers a web service that enables applications to interact with any text messaging network, including SMS, IM, Twitter, Facebook, and more.

All of Voxeo's products and services are available as both an on-premise, traditional software license and as on-demand, hosted, Software as a Service solutions.

TalkSwitch Announces the TS-350i and TS-550i IP Phones and Management Software 6.12

VoIP Monitor — Thu, 12 Nov 2009 17:42:59 GMT

TalkSwitch announces the launch of two IP phones, giving customers more choice, features and functionality. The phones run with TalkSwitch Management Software 6.12, which features a newly engineered line optimization tool and automatic VoIP service provider configuration.

These sophisticated, easy-to-use phones feature crystal-clear sound and exceptional speakerphones. Customers will hear the difference. They have big, bright multicolor programmable appearance keys for one-touch access to lines, extensions, queues, and features.

The TS-350i IP phone sports six programmable appearance keys, and rich features. It’s a powerful business workhorse that fits a wide range of users, and it’s value-priced at MSRP US$174. The TS-550i IP phone is a premium business phone for power users with 22 appearance keys and advanced expansion and connection options at MSRP US$264. The TS-50e expansion unit (MSRP US$99) adds 12 appearance keys to the TS-550i. Attach up to two for 46 keys in total.

Broadsoft Expands Belfast Operations

VoIP Monitor — Thu, 12 Nov 2009 17:05:00 GMT

BroadSoft announces the expansion of its Belfast operations to support the company's growth in the Europe, Middle East and Africa region. BroadSoft has worked with large carriers in Europe since 2002, and opened its European Headquarters at Northern Ireland Science Park in Belfast, Northern Ireland in 2006. Due to rapid growth, BroadSoft has more than doubled its space requirement in the past three years. BroadSoft's BroadWorks VoIP application platform is currently deployed by more than 100 service providers in the European market, including Telefonica, T-Systems, TDC, Slovak Telecom, Belgacom and KPN, and in more IP Multimedia Subsystem networks worldwide than any other application server.

With BroadSoft's "follow-the-sun" approach to global customer support, the company provides customers with technical expertise and assistance, regardless of location. BroadSoft's network of worldwide Operations Centers offers customers direct access to highly-skilled technical experts for resolving critical operational issues on a 24x7 basis.

San Isabel Telecom Deploys XOP Networks' VoIP Based Voice Mail Application

VoIP Monitor — Thu, 12 Nov 2009 16:56:08 GMT

XOP Network announces that San Isabel Telecom is using the XOP Networks' Universal Service Node with simultaneous T1 and SIP connectivity to provide voice mail to both traditional and VoIP customers. San Isabel Telecom is a facilities-based provider of voice and data services in several mountain regions of Colorado. The company was started by Doug Wagner in 1996 to meet the communications needs of the ever-growing Vail Valley. San Isabel also serves the Aspen and Basalt areas.

San Isabel Telecom has a TDM network with a Alcatel-Lucent 5ESS switch, and a next generation packet based VoIP network with a Cisco Gateway and Acme Packet Session Border Controller. San Isabel is meeting the challenge of providing voice mail to customers on the TDM network and the VoIP network at the same time by using the XOP Networks' USN to connect to the 5ESS switch and the VoIP based SBC equipment.

In addition to providing TDM and SIP connectivity, the USN based Voice Mail application allows voice mails to be accessed by telephone, web portal and PDAs such as iPhone and Blackberry. The PDA access is unique in that voice mails can be accessed in random order simply by a click on the PDA based web portal.

Zoiper.com Launches the First IAX and SIP WEB-based Softphone

VoIP Monitor — Wed, 11 Nov 2009 20:33:00 GMT

Zoiper WEB introduces communication functionality to any web application and VoIP-enables any website making Zoiper WEB beneficial for any business environment.

Zoiper WEB requires no desktop client-software installations, which have to be stored and configured – it simply eliminates any complexity by providing instant access to any services and products upon website visit or upon first use of a web application. This results in increase of the sales conversion rates, and drastically reduces website, shopping cart, or web application abandonment.

Zoiper does not require development or infrastructural hardware investments – it is simply uploaded to a web-server. Zoiper WEB per-domain licensing allows unlimited concurrent calls and unlimited users, avoiding the usual endless license fees related to communication software. Zoiper API allows customization of Zoiper’s standard functionality to perfectly suit any business’ needs. Zoiper API also allows delivery of a preconfigured ready-to-use solution to any audience thus providing smoother customer experience and reducing customer support costs.

Zoiper WEB is fully customizable to fit any website or web-application layout and feature requirements in its OEM version – Zoiper WEB’s development team can build a wide range of personalized communication solutions based on Zoiper’s architecture.

Zoiper WEB suite includes:

Active-x and npapi plugins in order to seamlessly run on the most popular browsers - Internet Explorer, Firefox, Safari, Chrome on Windows
Powerful API for functionality and provisioning configuration
JavaScript and HTML examples
smart or no interface
Autoanswer, Voice mail, Transfer achieved through JavaScript

VoIP-PAL.com Announces Reseller Website Development

VoIP Monitor — Wed, 11 Nov 2009 20:28:19 GMT

VoIP-PAL.com announces the expansion of its Product Development Division to fast track the launch of its first reseller program,www.PointsPhone.com. In addition, they will build the reseller websites and back-end for www.loyaltyphone.com and www.rewardsphone.com, all of which will be "powered by VoIP-PAL" and will include VoIP-Pal's new Mobil VoIP service to be offered at launch.

"Customers, through their purchase of a product from www.voip-pal.com, can experience cost-effective communications, without the necessity of additional hardware or equipment," said Richard G. Kipping, Chief Executive Officer. In addition, Kipping said, "VoIP-PAL management continues to take actions to achieve its goals, providing state of the art communication services that will enable local, North American and international capabilities and dedicated social networking."

VoIP-Pal anticipates a quick implementation of its platforms, and will be announcing several new programs during 2009.

Mr. Kipping further commented, "This is an exciting time for VoIP-PAL, as we are now moving the company to the next level of distribution with a Reseller program, and by launching our new IPHONE and IPOD compatible Mobil VoIP service, we will be current with the demands of consumers in our market niche."

VocalTec and Runcom Offer Integrated WiMAX and VoIP Solutions

VoIP Monitor — Wed, 11 Nov 2009 18:48:44 GMT

VocalTec disclosed their integrated WiMAX VoIP solution today at the AfricaCom Conference and Exhibition. The companies announced that they will collaborate on the provisioning of WiMAX VoIP solutions to service providers in emerging telecom markets.

The joint offering enables the delivery of both broadband and voice services to remote areas where traditional access is unavailable, assuring interoperability and ease of deployment. It also allows service providers to effectively provide high quality VoIP services to their subscribers at low cost.

VoicePulse Now Offers Numbers in 32 Rate Centers Across South Dakota

VoIP Monitor — Tue, 10 Nov 2009 20:37:20 GMT

VoicePulse announces the immediate availability of phone numbers in the 605 area code for business and residential customers. VoicePulse now offers numbers in 32 rate centers across South Dakota, more than four times as many as well-known competitor, Vonage who currently has availability in just seven rate centers. VoicePulse now offers more coverage in the state of South Dakota than any other pure-play VoIP provider.

The addition of South Dakota's 605 area code allow consumers to port their current phone number to VoicePulse's residential or SIP Trunking services in addition to the option of choosing a new phone number. For those customers looking to port a number from a rate center not listed above, portability can be checked with VoicePulse's online availability tool at voicepulse.com/availability.

VoicePulse Residential and Home Office

Over 27 innovative features available, including:

Telemarketer Blocking, Do Not Disturb, Anonymous Call Rejection, and Call Transfer
Voicemail with optional e-mail delivery of messages as sound attachments
Free Local Number Portability

Plan Pricing:

Unlimited local, regional and US long distance calling for $24.99 per month
Unlimited local, regional and 200 US long distance minutes for $14.99 per month

VoicePulse SIP Trunking for Business and Wholesale

VoicePulse also offers SIP Trunking and origination/termination services to businesses and commercial users. VoicePulse services are highly compatible with PBX clients such as Allworx , Asterisk, AsteriskNow, Fonality's PBXtra, FreePBX, FreeSwitch, PBX-in-a-Flash, 3CX, Switchvox, trixbox and more. In addition, VoicePulse offers a trixbox/FreePBX based auto-configuration module, giving businesses the ability to fully configure their own PBX.

VoicePulse provides comprehensive support services for businesses of all sizes - knowledgeable support teams answer questions via email and phone while advanced technical support is available to optimize VoicePulse service for the company's IP PBX. VoicePulse has received industry recognition for its excellence in customer support and service.

Ingate Systems Now Certified as ''Avaya Compliant'' with Aura

VoIP Monitor — Tue, 10 Nov 2009 17:49:22 GMT

Ingate announces that the Ingate SIParator and SIP Trunking software module are compliant with key SIP trunking and IP telephony PBX and Unified Communication solutions from Avaya. The SIParator helps businesses connect IP-PBXs and Unified Communication solutions to Internet telephony service providers' SIP trunk offerings instead of conventional PRI trunks, giving them both lower cost and more features for their telephony connection. The SIParator also makes it possible for home workers and road warriors to use the company PBX and allows Unified Communication to be global, stretching outside the Local Area Networ. The SIParator and Ingate SIP Trunking software module are now compliance tested by Avaya for compatibility with the Avaya Aura Session Manager 1.1 SIP Trunking, Avaya Aura Communication Manager 5.2 and Avaya Aura SES 4.2. This is in addition to Ingate's existing certification with other Avaya IP telephony solutions.

The SIParator is an Enterprise Session Border Controller that enables the secure, simplified adoption of SIP communications, including VoIP and SIP trunking. It solves the SIP traversal of firewalls and Network Address Translation and can also enable far-end NAT traversal to extend the SIP capabilities within the corporate network to remote users. SIParators feature Ingate's Firewall SIP proxy architecture, which delivers maximum control over SIP signaling, traffic and network security, creating one truly converged network for both data and VoIP.

In SIP trunk installations, SIParators ensure seamless interoperability between virtually all service providers' SIP trunk offerings and the Avaya Aura Communication Manager and the Avaya Aura Session Manager. The Ingate E-SBC offers Avaya customers many advantages, including converging voice with data on a single LAN to obtain maximum use of multimedia terminals such as desktop and laptop computers and SIP-enabled mobile phones.

Ingate 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, Ingate is eligible to submit products for compatibility testing by the Avaya Solution Interoperability and Test Lab in Basking Ridge, N.J. 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.

Open Source BroadVoice to Improve Voice Fidelity and Cost-Effectively Transition to HD VoIP

VoIP Monitor — Tue, 10 Nov 2009 17:13:56 GMT

Broadcom is offering its BroadVoice family of voice codecs royalty-free and without any licensing fees. As a direct response to customer demand for advanced, high-quality voice solutions and development tools, Broadcom is releasing its wideband and narrowband BroadVoice codecs in both floating-point and fixed-point C code as open source software under the GNU Lesser General Public License, version 2.1, as published by the Free Software Foundation.

By eliminating the royalties and licensing fees (required by competitive solutions), Broadcom is driving a cost effective transition to HD VoIP applications by enhancing the quality of voice transmissions enabling a higher quality audio experience. The availability of BroadVoice source code, under an open source software license, provides the industry with maximum flexibility in how it can be deployed and has the potential of addressing a wide range of next generation voice-related applications.

Highlights/Key Facts:

Broadcom developed the BroadVoice family of voice codecs with two variants including a 32 kb/s version called BroadVoice32 for wideband (HD) speech sampled at 16 kHz, and a 16 kb/s version called BroadVoice16 for narrowband telephone-bandwidth speech sampled at 8 kHz.
BroadVoice advanced voice codecs reduce the latency, complexity and bandwidth usage on a wide range of wideband and narrowband voice applications including voice-over-cable, voice-over-DSL, Ethernet IP phones, Wi-Fi VoIP phones and software-based VoIP client solutions. Additionally, for VoIP applications, distortion and echo are also reduced.
BroadVoice is available on Broadcom’s cable, DSL and VoIP system-on-a-chip solutions enabling manufacturers and service providers to improve the customer’s audio experience.
When standardized by SCTE and ANSI, the BroadVoice16 and BroadVoice32 codecs are called BV16 and BV32, respectively. BV16 is a standard codec in PacketCable(TM) 1.5, PacketCable 2.0, ANSI/SCTE 24-21 2006, and ITU-T Recommendation J.161 specifications. BV32 is a standard codec in PacketCable 2.0, ANSI/SCTE 24-23 2007, and ITU-T Recommendation J.361 specifications.
The real-time transport protocol payload formats for BroadVoice16 and BroadVoice32 are specified in RFC4298.
BroadVoice16 and BroadVoice32 have very similar codec structures. Both variants share most of the algorithm modules so when implemented together, substantial code sharing and memory reduction can be achieved.
Broadcom is providing both the floating-point and fixed-point C source code of BroadVoice16 and BroadVoice 32 under an open source license (LGPL version 2.1) and on a royalty-free basis.
For more technical information on BroadVoice codecs, or to download the BroadVoice open source C code, please visit: http://www.broadcom.com/broadvoice.

Small Biz Expert Steve Strauss Explains 'Why I Vote for VoIP Over Standard Telephone for Small Business' in Upcoming Webinar

VoIP Monitor — Tue, 10 Nov 2009 17:01:38 GMT

Internationally recognized small business author, columnist and speaker Steven Strauss will conduct a free webinar discussing the advantages of VoIP phone service for small business over traditional phone service on Thursday, November 12, 2009 at 2:00 p.m. ET. Sponsored by IP communications solution provider 8x8, the webinar is intended to explain, in layman's terms, how VoIP technology works and how it enables companies to operate smarter and more efficiently while simultaneously reducing monthly overhead expenses.

As the author of 15 books, including the best-selling Small Business Bible, Strauss is dedicated to providing entrepreneurs and small business owners with strategies for growth covering every aspect of a business' operations -- from sales and marketing to finance and insurance to equipment and technology. His column, Ask an Expert, appears weekly in USA Today and is one of the most highly syndicated business columns in the world.

According to Strauss, although the economy has forced small businesses to do more with less, advancements in technology are enabling companies to maintain their competitive edge despite a reduction in orders, employees and budgets. "The truth is, even today, there are ways to remain competitive, look professional, get the job done and even save money in the process," says Strauss. "By doing something as simple as switching to a VoIP phone system, you can get superior phone quality and powerful features at a fraction of the cost of an old-fashioned phone system."

During the one-hour webinar, Strauss will explain how VoIP phone service easily integrates into any business environment, improves productivity, provides powerful features and benefits and saves significant cost. Strauss will also discuss how VoIP phone service gives companies the flexibility to work and call from any location equipped with broadband Internet service.

"One of the great changes that the Internet has produced for small business is that not only can you run your business from anywhere at any time, but you can actually create a virtual business," explains Strauss. "With VoIP phone service, if you are working from home one day and you have high speed Internet access there, then your VoIP phone at home will act just like your VoIP phone at the office. This sort of mobility and flexibility is not only a great benefit, it can also save you money as it helps eliminate the need for costly office space."

To participate in the November 12, 2:00 p.m. ET free webinar, go to http://www.get8x8.com/landingpage/WS-WC-VoIPSB-12NOV09/ and complete the registration form.

Nimbuzz Launches NimbuzzOut for Low-cost International Calling

VoIP Monitor — Mon, 09 Nov 2009 19:00:57 GMT

Nimbuzz launches NimbuzzOut, offering great value international calls at the lowest possible rates, to complement its multi award-winning mobile application offering free VoIP calls, SIP calls and social IM aggregation.

NimbuzzOut Credits will be available for purchase, enabling users to save up to 95% on the cost of a traditional international mobile call.

Nimbuzz' free to download, free to use mobile IM and VoIP application, launched in May 2008, already enables millions of users worldwide to enjoy free calls and instant messaging with friends across both its own network, and users' favourite online social communities (Skype, Windows Live Messenger, Yahoo! Messenger, ICQ, Google Talk (Orkut), AIM, Facebook, MySpace and many more)

The launch of NimbuzzOut, starting with Symbian, iPhone and iPod Touch handsets and rolling out to others later this year, adds great value and great quality calling from mobiles on 3G or Wi-Fi to any other mobile or landline worldwide.

Nimbuzz technology has pioneered next-generation mobile calling, with over 3million Nimbuzz calls made every single week! NimbuzzOut represents the company's first revenue-generating feature, launched in direct response to user requests of whom 69% recently surveyed* said they expect to use mobile VoIP more than ever in the coming 12 months.

NimbuzzOut comes at a time of phenomenal growth for Nimbuzz, with more than 10million registrations of the application since launch, a current growth rate of more than a million new registrations each month (that's a new Nimbuzz user every 3 seconds!), a daily active user base of over 30% and an international user footprint covering every corner of the globe.

Adopting a standard pay-as-you-go model, NimbuzzOut credits will be sold in bundles of 10, 20 and 40 Dollars or Euros. Users can learn more and check call savings per country via the rate calculator on the Nimbuzz web and wap sites.

Doddle Makes VoIP Easy as Pulling up a Website

VoIP Monitor — Mon, 09 Nov 2009 18:52:12 GMT

Doddle allows users to make VoIP calls anywhere in the world via Doddle’s web based SIP phone directly from their webpages. With Doddle’s free online phone service, no registration is required and VoIP is as easy as accessing a webpage. Users just begin using.

Doddle’s web based phone allows anyone to add phone capabilities to their web applications, including websites, portals, social networks and blogs such as Facebook, MySpace, Orkut, Blogger and more, so that web visitors can make calls directly from the webpage. Twitter tweets can truly make sounds with an instant call link.

All that is required is to embed Doddle’s free linked phone gadget, which empowers users’ webpages with a webphone / click2talk feature. The options and features provide a completely customizable solution for both personal and business use:

Webphone for blogs, homepages, social networks and virtual business cards
Internet service and VoIP providers, IT companies, SOHO
SIP compliant phone: seamless integration with VoIP providers
Compatible with VoIP Analog Telephone Adapters (ATA/Hardware)
Server side integration: J2EE (Java) /.NET / PHP / Database
JavaScript API (Mac OS X, Windows, Linux)
No need to keep computers powered on to receive calls
WebPhone, Click2Call, Call Me Button, PhoneBooks, etc.
VPN Support

Doddle’s web-driven phone makes it possible to quickly add SIP-based calls features in web applications. There is no desktop application to be downloaded. Doddle’s webphone is supported in all major browsers for both PC and Mac. Doddle offers several free widgets: webphone, call button, and iGoggle. Fully customizable versions are also available.

Skype Appoints Dr. Jonathan Rosenberg as Chief Technology Strategist

VoIP Monitor — Mon, 09 Nov 2009 18:46:55 GMT

Skype has appointed Dr. Jonathan Rosenberg as Chief Technology Strategist, effective immediately. Reporting to Daniel Berg, Skype’s Chief Technology Officer, Rosenberg will be responsible for shaping and driving the company’s technology strategies and overall architecture to continue delivering a best-in-class Skype experience to users around the world.

“We are tremendously excited to have Jonathan Rosenberg join us. His experience and his track record of innovation within the industry will be a great asset for us,” said Berg. “Having someone of Jonathan’s caliber join our talented technology team will help us continue to stay ahead and build for the future.”

Rosenberg brings with him a wealth of industry experience. He is most well known as the co-author of the SIP and has authored many of the standards that are in wide use today within the telecommunications industry. Prior to joining Skype, Rosenberg was a Cisco Fellow at Cisco Systems, where he directed technology strategy for its enterprise voice product portfolio. Before that, he was CTO of dynamicsoft, one of the industry's first VoIP technology companies. In that role, he helped build products and networks for several of the earliest VoIP deployments. Rosenberg has received many industry accolades for his role as a pioneer and innovator in telephony.

Rosenberg will be based in the U.S and will work with Skype's engineering teams worldwide.

Digium CEO Danny Windham to Deliver Keynote Address at ITEXPO East 2010 in Miami

VoIP Monitor — Mon, 09 Nov 2009 18:30:20 GMT

Technology Marketing Corporation announces that Digium CEO Danny Windham has accepted its invitation to deliver a Keynote Address at ITEXPO East 2010, taking place January 20-22, 2010 at the Miami Beach Convention Center in Miami, FL.

Mr. Windham’s presentation, which takes place Thursday, January 21st at 9:45AM EST, will address the evolution of Open Source as a mature telephony platform that is experiencing extensive growth in enterprise, government and carrier markets. Open to all ITEXPO attendees, this session will also be a centerpiece of the fourth annual Digium|Asterisk World conference, which is collocated with ITEXPO East 2010.

ITEXPO is the world’s largest conference and trade show focused on communications and technology. Launched in 1999, the event is expected to draw as many as 200 exhibiting companies and more than 7,000 attendees to Miami.

“Developers, resellers, IT professionals and executives will benefit from hearing Danny’s views on the increasing number of opportunities based on Asterisk, the world’s most popular open source telephony software,” said Rich Tehrani, CEO and group editor-in-chief of TMC. “ITEXPO and Digium|Asterisk World will expand on this notion, with three days of content, education and training for anyone interested in learning more about open source telephony.”

Digium is the creator, sponsor and primary developer of Asterisk. Digium offers Asterisk software free to the open source community and offers Asterisk Business Edition and Switchvox IP PBX software to power a broad family of products for small, medium and large businesses. The company's product line includes a wide range of hardware and software to enable resellers and customers to implement turnkey VoIP systems or to design their own custom telephony solutions.

Danny Windham joined Digium in February, 2007 as CEO. He is responsible for setting the company’s corporate strategy, and executing its day to day business operations. Prior to joining Digium, Windham served as president and chief operating officer of ADTRAN, a global provider of networking and communications equipment. He joined ADTRAN in 1989, following ADTRAN's successful acquisition of Processing Telecom Technologies, a company Windham co-founded in 1986. Windham holds a Bachelor of Science degree in Electrical Engineering from Mississippi State University, where he was named a Distinguished Engineering Fellow in 2001 and he also holds an MBA from the Florida Institute of Technology.

“Whether building a corporate phone system from the ground up, or extending the functionality of a legacy telephony system, organizations of all sizes are increasingly turning to open source VoIP as a flexible and inexpensive solution,” said Windham. “I’m honored to be delivering the keynote at ITEXPO East 2010 and sharing some of the great accomplishments of open source in the enterprise with attendees.”

Registration for ITEXPOE and Digium|Asterisk World remain open.