Business News Holding

Category: voip

  • XConnect Offers Trial of High-Definition Calling

    XConnect announced a trial of the first IP peering federation specifically for service providers capable of offering high-definition voice services.

    The trial, open to qualified operators, waives sign-up and monthly fees for its April-June duration. “Multiple providers using the G.722 wideband codec will be able to test the interoperability, scalable interconnection, reliability and support of XConnect federation services,” says the company.

    High-definition voice is being adopted increasingly by fixed, mobile and Web 2.0 telecom service providers, as it delivers a much richer audio experience than the PSTN makes possible.

    Using wideband codecs, HD achieves a wider frequency range, providing almost the clarity of face-to-face conversation.

    However, for HD to work across networks, the entire call path and endpoints themselves must support high definition. According to Eli Katz, XConnect CEO, the mass-market adoption of high-definition voice and other new IP services demands “trusted, scalable cross-network interconnection.”

    “Service providers are eager for a solution. We look forward to working with the industry to help bring the benefits of HD voice to these operators and the consumer and enterprise markets they serve,” he said.

    Jeff Rodman, Polycom co-founder and CTO, said, “Because voice is the most critical way that we communicate, the significantly improved sound quality of HD voice is an important step in making communication clearer and more effective.”

    Trial participants will form a private peering community under the Private Alliance feature of XConnect’s Global Alliance, which combines ENUM-registry and multimedia interconnection hub services. Supporting multiple protocols and codecs, the Global Alliance enables new IP services, including HD voice, to be delivered across networks.

  • Dialogic to Provide “Any-to-Any” PBX Connectivity for SIP Trunking

    Dialogic announced that it has entered into an agreement with Ingate Systems and says this allows them to incorporate the SIP Trunking software module from Ingate into a new enterprise border element designed to connect virtually any SIP trunk with virtually any PBX, to facilitate seamless SIP trunk deployments in legacy TDM and hybrid PBX environments, as well as new SIP-based PBX systems.

    “PBX’s are transitioning from the traditional TDM PBX’s to hybrid PBX’s, IP-PBXs, and Unified Communications solutions creating a heterogeneous TDM/SIP trunk environment and there is a significant opportunity to provide connectivity and security between public and private networks,” said Franz-Josef Eberle, Vice President and General Manager for the Enterprise Market Group at Dialogic.

    The Ingate SIP Trunking software module provides enterprise session border control along with the routing capabilities necessary to connect SIP trunks to enterprise networks and branch offices by employing Ingate’s proxy-based traversal and security technology.

    The SIP Trunking software also is designed to resolve interoperability issues between service providers’ SIP services and the SIP-based systems being deployed inside corporate data networks today.

    Dialogic’s media gateway technologies provide the protocols and interfaces necessary to connect with a wide variety of legacy telephony equipment and networks, both TDM and IP.

    Dialogic says its new enterprise border element will combine the functions normally found in a media gateway and an enterprise session border controller into a single product. “The result will be a solution that is designed to connect virtually any trunk with virtually any PBX, thus helping to resolve the connectivity issues with the heterogeneous environment,” says the company.

    According to Steven Johnson, President of Ingate Systems, the Dialogic solution will open the opportunity of a rapid return on investment to enterprises with mixed PBX environments, including those using traditional PBXs.

    Dialogic plans to make a first set of enterprise border elements available later this year with product configurations offering SIP trunking legacy PBX connectivity via PRI/E1/T1 and ISDN BRI.

  • Skype’s On-Net International Traffic Growing Fast

    New data from TeleGeography show that the growth of international telephone traffic has slowed, while Skype’s growth has accelerated.

    Over the past 25 years, international call volume from telephones has grown at a compounded annual rate of 15 percent. In the past two years, however, international telephone traffic annual growth has slowed to only 8 percent, growing from 376 billion minutes in 2008 to an estimated 406 billion minutes in 2009, according to recent TeleGeography research.

    The deep recession has had a marked impact on many routes. "Demand for international voice has been remarkably robust, but it’s clearly not recession-proof," said TeleGeography analyst Stephan Beckert.

    “While international telephone traffic growth has slowed, Skype’s traffic has soared,” he added.

    Skype’s on-net international traffic (between two Skype users) grew 51 percent in 2008, and is projected to grow 63 percent in 2009, to 54 billion minutes. That means that about 13 percent of international calls are made on Skype.

    "The volume of traffic routed via Skype is tremendous. Skype is now the largest provider of cross border communications in the world, by far," said Beckert.

    He claims that the proliferation of alternatives to telephone calls—including Skype for mobile devices, and Google’s gradual entry into the voice market—will present ever greater challenges to international carriers.

  • Richard Shockey Named New Board Chairman of SIP Forum

    The SIP Forum, an IP communications industry association that engages in numerous activities that promote and advance SIP-based technology, has announced the recent re-election of industry veteran and VoIP pioneer Richard Shockey to the Board of Directors, and the election of Shockey as new Board Chairman.

    Additionally, the Forum has re-elected Dr. Eric Burger to the Board of Directors and named him Chairman Emeritus, and elected Dr. Alan Johnston to the Board.

    Richard Shockey, founder of Shockey Consulting, is an industry veteran with a decades-long and distinguished track record in helping shape numerous technical standards that have become the foundation for today’s SIP-based next generation network infrastructure and application ecosystem.

    Richard Shockey

    He served as a Director and Member of Neustar Inc.’s Technical Staff, which provides a number of critical services to the communications industry including the administration of all telephone numbers in North America, management of the wireline and wireless Number Portability Administration, number pooling, and OSS products for carriers.

    “I look forward to continuing to build on the solid foundation left by my predecessor, Dr. Eric Burger, and ensuring the successful completion of the important work in progress in the SIPconnect, Fax-over-IP and User Agent Configuration task groups. I also look forward to expanding the work of the SIP Forum into new and exciting industry sectors, including Smart Grid and Unified Communications,” said Shockey.

    Rejoining the SIP Forum Board of Directors, Dr. Alan Johnston brings nearly two decades of industry experience. He has been involved with SIP and VoIP since the mid-1990s, helping to spearhead the development and adoption of SIP and VoIP in both the service provider and enterprise markets.

    Alan Johnston

    He served as an architect on the first enterprise SIP VoIP product in the U.S. as a Distinguished Technical Member at MCI.

    He co-authored the SIP protocol specification RFC 3261 and edited the Basic and PSTN call flows Best Current Practices documents, RFC 3665 and RFC 3666, along with additional RFCs. He has also worked on SIP Service Examples, Peer-to-Peer SIP and security, and co-authored the ZRTP protocol.

    He is currently a Consulting Member of the Technical Staff of Avaya.

    “As SIP approaches critical mass in the market, the SIP Forum continues to play a significant role breaking down the barriers to true interoperability between vendors, platforms, applications and more,” said Johnston.

    “This highly respected organization is shaping the future of how companies, customers and users communicate, and I am honored to be rejoining the board.”

  • Tight Competition for Biz-News.com Product of the Year Awards

    Contenders for the Biz-News.com Product of the Year Award are on a tight struggle to obtain the leading amount of votes for the 2009 Awards.

    The awards are people’s choice; users are being called upon to express their consumer experiences during 2009.

    What VoIP service worked for you best? Which would you recommend to your friends and colleagues?

    Let us know and award them a vote by filling this form.

    Currently the top contenders are:

    • IPsmarx SIP Based Calling Card Platform by IPsmarx Technology
    • Asterisk by Digium
    • IXC Softswitch by IXC
    • Fonolo by Fonolo
    • Phybridge Uniphyer by Phybridge Inc.
    • Truphone by Truphone
    • FreePBX by FreePBX
    • 3CX by 3CX

    You can vote for them or nominate your personal favourite.

    If you have more than one nomination for "Best Product" you can make multiple submissions – but you can only vote once for any product.

    Voting closes on the 15th of February!

    May the best be rewarded.

  • Voxbone: Revenue Rises 60 Percent, Network Traffic More Than Doubles in 2009

    Despite one of the worst global economic recessions in decades, Voxbone, a provider of IP network services, "saw another year of substantial growth in revenue, network traffic and new customers in 200", the company informed today.

    According to Voxbone, 2009 revenue was up 60 percent over the prior year, and minutes of use for inbound traffic grew 110 percent to 1.5 billion. In addition, Voxbone attracted more than 150 new customers to its base of service providers, multinational corporations, call center operators, conference call providers and others in the IP industry.

    Voxbone provides these customers with direct-inward-dial (DID) and iNum numbers for local presence in foreign markets while enabling them to deploy new advanced services via backbone dedicated to voice-origination services.

    Demand from conferencing providers, mobile VoIP providers, Web and telephony integrations, and enterprise customers accounted for much of Voxbone’s 2009 growth.

    Businesses’ global expansion, transition from the PSTN to IP, and deployment of high-definition voice and video services helped drive growth for the company.

    Voxbone reported it has assigned 18 million iNum numbers – global phone numbers with the “area code for Earth” – since introducing the service in November 2008.

    The company plans to make SMS services available through iNum and increase interoperability with mobile operators in 2010.

    Rod Ullens, Voxbone CEO, said, “What makes us different, and one of the main reasons for our success, is that Voxbone has been built from the beginning with an unusual mix of software expertise, a passion for open-source, deep understanding of telecom infrastructure and widespread global presence; we’re now in 49 countries.”

  • 3CX Announces 3CXPhone 4.0 – a Free Softphone in Smartphone Look

    3CX released a new version of its free VoIP softphone for Windows – 3CXPhone 4.0.

    3CXPhone is a free, SIP-based VoIP phone that allows to use any PC or laptop as a phone. Making calls to any VoIP, mobile or landline number is possible after connecting 3CXPhone to a VOIP provider or to a VoIP PBX. With 3CX Gateway for Skype users can also make and receive calls to Skype numbers.

    3CXPhone allows to choose from several popular phone interfaces (Nokia, iPhone), features multi-line capability (up to 3 lines), ability to transfer calls, calls to disk recording and works with Asterisk and popular VoIP providers.

    It supporst G.711, GSM, iLBC and Speex codecs as well as USB and Plantronics headsets and also features STUN support for NAT/firewall traversal.

    3CXPhone is provided completely free of charge to individuals and organizations including commercial entities (all features, including call transfer, are enabled).

    “The soft-phone is becoming a serious ‘IP phone option’ for businesses. They are easy to manage, save on electricity and administration,” said Nick Galea, 3CX CEO.

    “3CXPhone can easily be configured as a remote extension, allowing users away from the office to easily connect to the corporate phone system. The unique tunnel feature proxies all SIP & RTP traffic over a single port and makes firewall and NAT configuration a breeze. It is also possible to configure it as a remote extension using ‘direct SIP’,” he explained.

  • Skype Names David Gurle to Lead Skype for Business Team

    Skype today announced that it has hired David Gurle as the new General Manager and Vice President of its Skype for Business unit. He replaces Stefan Oberg, who will be leaving Skype in March 2010.

    David Gurle joins Skype from Thomson Reuters, where he served as its Global Head of Collaboration Services and Head of its largest business in Asia, the Sales & Trading Business Division.

    Prior to joining Thomson Reuters, he spent more than three years running Microsoft’s Real Time Communications business, a group that he founded. During this time, he oversaw the development of the company’s collaboration products including NetMeeting, Windows Messenger, Exchange IM, Exchange Conferencing Server, Live Communications Server and Office Communications Server, as well as Microsoft’s acquisition of PlaceWare.

    David Gurle

    Prior to Microsoft, he was Corporate Vice President, Business Alliances at VocalTec, the IP telephony pioneer, where he established and managed partnerships with a number of Tier I telecommunications service providers and hardware vendors, including Deutsche Telekom, France Telecom and Marconi.

    He also spent time at ETSI, France Telecom and started his career at Digital Equipment Corporation (DEC). David has also co-authored a number of books on VoIP and IP Telephony.

    He graduated ESIGETEL with a Masters of Science degree in Computer Science and Telecommunications.

    "Since its inception, Skype has been used by many entrepreneurs and small businesses to save money on their communications," said David Gurle.

    "Moving forward, our goal is to educate and attract larger organizations that can not only save money by using Skype to communicate, but also increase their organizational productivity and enhance the way they interact with customers around the globe."

  • Acrobits Provides Three New SIP VoIP Operators with the iPhone Apps

    Acrobits, a Czech Republic-based mobile software development company, has just released their latest white label clients for the iPhone: PLFon, TeleSIP and sipcall.

    This comes on the heels of their recent announcement to put renewed focus on creating white label softphones for the iPhone. These SIP VoIP providers are now on even footing with the VoIP giants that already have their own softphone applications on the iPhone.

    The white label clients give the providers access to iPhone customers they might not reach otherwise. Though Acrobits Softphone is compatible with virtually any SIP provider, some customers are more likely to use a provider that has their own softphone.

    “While techies love plaving with different softphones and comparing VoIP operators, your average VoIP user is going to do most of their calling through one provider. Having your own iPhone Softphone application brings you one step closer to convincing customers to give your service a try, rather than one of the other hundreds of VoIP operators that are out there,” says Acrobits.

    Acrobits Softphone, the SIP VoIP phone for the iPhone and iPod Touch, is the flagship product of our company, which is also planning to support Symbian and Windows Mobile phones in the near future.

    Acrobits is already working on Softphone clients for other VoIP operators, including Gizmo5.

    “VoIP service is a highly competitive industry and VoIP usage on mobile devices, especially the iPhone, will play a large part in deciding who tomorrow’s leading VoIP providers are,” says Acrobits.

  • Digium and Aumtech Certify a Multi-Language Speech Recognition Server Solution with Asterisk

    Digium and Aumtech, Speech and Computer Telephony Integration company, announced a partner relationship between the companies.

    According to them, Asterisk users now have a high-quality, low-cost Speech alternative, featuring server-based licenses that support 48+ ports of Automated Speech Recognition for less than the cost of one or two competitive ASR licenses.

    Aumtech’s solution provides 48+ ports of multi-language speech recognition for the Asterisk platform with support for $1,975 per port.

    The firm is introducing its Media Resource Control Protocol (MRCP) Connector utility, which enables Asterisk customers to access the high-quality Microsoft Speech Services 2007 ASR for a fraction of the cost of competitors’ ASR engines, as the company claims.

    This includes unlimited grammars and multiple languages of ASR. Aumtech says the list of 14 available languages and voices is constantly growing, and there are successful installations in North America, South America, Europe and the Far East.

    Asterisk users can access this speech solution on Asterisk and on a variety of Interactive Voice Response solutions, including Aumtech’s VoiceXML 2.1 Certified IVR platform, or other standards-based IVR platforms.

    The Aumtech MRCP Connector for the Microsoft Office Communications Server 2007 Speech Server allows companies using Linux to build speech-driven applications with the Microsoft ASR product.

    Since the MRCP Connector is based on the WC3 Forum’s open standard, applications running on open platforms such as Asterisk can access Microsoft’s Speech Recognition functions through Aumtech’s Connector. Existing Speech applications are also candidates for deployment on this solution, since industry standards ensure code portability.

    Bill Miller, Digium’s vice president of product management, said, “Our Asterisk customers have long recognized the value of Speech Recognition applications; however, most have delayed implementing these technologies due to the high entry costs. Aumtech’s solution is the lowest-cost at higher capacities for world-class Speech technology and may prove to be the ‘tipping point,’ finally bringing speech to the masses, and will benefit our customers worldwide.”

    Digium is the creator, sponsor and driving force behind Asterisk, the most widely used open source telephony software. The company’s software and hardware products enable businesses to implement turnkey unified communications solutions or to design their own VoIP systems.

    Aumtech is the Automatic Speech Recognition systems provider for large enterprises, governments, and global carriers. Its innovations include the first SIP-based IVR, Automated Outbound IVR Notification, and a standalone CTI product, called ESP (“Economical Screen Pop”).