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02/09/2010 - Skype Connect 1.0 Officially Launched

Skype on Monday announced the official launch of Skype Connect 1.0, formerly known as Skype for SIP. Previously available in beta, Skype Connect delivers a business solution that enables IP-enabled private branch exchange (PBX) or Unified Communications systems to connect to Skype.

30/08/2010 - Acrobits Promotion Rewards Loyal Customers of their iPhone SIP Client

Acrobits recently made a big splash in the VoIP world with their new business caliber SIP client for the iPhone, Groundwire. While this has legions of VoIP users on the iPhone very excited, it was also important to Acrobits to reward users of their already excellent SIP client Acrobits Softphone with an inexpensive upgrade option. Unfortunately, iTunes doesn’t offer a viable option for Acrobits to do this. It looked like existing users of Acrobits Softphone would need to purchase Groundwire at full price to benefit from the exciting new features of Groundwire. Not to mention the licensing agreement for the G.729 codec would require users to purchase a new license to use the codec in Groundwire (G.729 licensing requires a separate license for each client or device).

This was unacceptable to Acrobits. “We value our customers and want to reward loyal users of Acrobits Softphone,” says Acrobits. So for two days only (August 30th and 31st, 2010), Groundwire is on sale for only $2.99. And users who want the benefit of using the low bit rate, high quality G.729 codec will be able to purchase the license at half price, only $4.99. Not only is Acrobits leading the market in mobile SIP applications, but they are proving themselves adept at building and maintaining customer loyalty as well.

Though the promotion is targeted toward their existing customer base, new customers will benefit from the promotion as well. So if you were considering purchasing a SIP client for your iPhone, or were waiting for the right time to delve into the world of VoIP, now is the time to do it. You won’t be disappointed.

Acrobits will continue to bring new and exciting SIP clients to the mobile VoIP user in the coming months. Their Acrobits Softphone client for Android should be available on the Android Market sometime in September and you can expect an iPad specific version by the end of the year. Keep an eye out for this up and comer in the VoIP world.

26/08/2010 - AT&T Adds IP Voice Services to Virtual Private Network Services

AT&T announced that new and existing virtual private network (VPN) customers may add VoIP service to the network solution delivered over AT&T's global network cloud. This converged solution is said to enable customers to consolidate their separate voice and data networks, reduce equipment and maintenance costs, and simplify migrating these complimentary capabilities to a common, secure infrastructure.

24/08/2010 - Acrobits Releases Groundwire, the First Business Caliber SIP Client

Acrobits has released their new business caliber SIP Client, Groundwire for the iPhone. It supports transfer and attended transfer, call waiting and conference calling. It also adds a voicemail waiting indicator and a programmable voicemail dialer.

23/08/2010 - Acrobits Releases Groundwire, SIP Client Utilizing all the Capabilities of the iPhone

Mobile software developer Acrobits released their new business caliber SIP Client, Groundwire for the iPhone, this week. This news should delight both iPhone owners and the VoIP world in general. SIP users finally have a mobile client capable of meeting all their needs.

Groundwire has all the great features of Acrobits’ current softphone client, Acrobits Softphone, and adds some very important business caliber features. Groundwire supports transfer and attended transfer, call waiting and conference calling. It also adds a voicemail waiting indicator and a programmable voicemail dialer. But that’s just a taste, here’s the full list of features.

Multitasking background support for iOS4
Transfer and attended transfer
Push Notifications, a reliable way to receive calls when the Softphone is closed
Multi line
Call waiting, switch between calls with headset controls or just shake the iPhone
Conference calling
voicemail waiting indicator with programmable dialer
Customizable ringtones, choose from our selection or pick something from your media library
HD Wideband audio through G.722
excellent sound quality, includes the G.711, GSM and iLBC audio codecs. Make an in app purchase to add G.729 Annex A for great quality over 3G networks
Completely redesigned audio optimized for the best VoIP experience on the iPhone
Number rewriting, enabling you to utilize your existing contacts without having to create new entries to satisfy the dialing requirements of your PBX or SIP provider
Address Book Matching to automatically format your contacts into the proper international format
TLS support for encrypted SIP
Bluetooth support for iPhones with OS 3.1 and higher
audio codec manipulation, enabling you to prioritize the codecs used and disable ones you don’t want to use
call recorder and player, seamlessly integrated into the call history
comfortable and super-smooth user interface, fine-tuned specially for the iPhone
customizable background and colors
full localization (currently English, Chinese, Danish, Korean, Norwegian, Polish, Portueguese, Swedish, French, German, Italian, Russian, Spanish, Czech and Slovak)
very easy configuration
simultaneous registration of multiple SIP accounts, have multiple accounts registered to receive incoming calls and switch the account used for outgoing calls without leaving the keypad
iPhone contacts integration. Easy to call anyone in your contacts via SIP
Contacts search function, search your contacts by name or number
add new contacts directly from the softphone
Quickdial, your 12 favorite contacts are one touch away
ability to generate DTMF tones while in call, to control various PBX features or automated systems (use audio, rfc 2833 or SIP INFO)
speakerphone support
detailed call history, with intelligent call grouping for an easy overview
support for sip:username URLs in phonebook
configurable RTP port range
SIP Proxy support, VPN support
STUN server support, automatic service discovery using DNS SRV queries
quick import of accounts from major VoIP Providers, like Gizmo5, Voipcheap, TerraSIP and others
phone number resolution. We present the phone numbers in a convenient format with grouped digits, display the flag and country name and even region/city name for some countries, including the U.S.

Acrobits has once again proven their dedication to providing the best VoIP experience on the iPhone. And with their upcoming release of Acrobits Softphone for Android, SIP users on Android will soon be singing Acrobits’ praises as loudly as iPhone users do now.

18/08/2010 - LG-Ericsson and Accton to Deliver Unified Voice and Data Solutions for Businesses

LG-Ericsson USA officially launched its brand into the North American market offering a broad portfolio of end-to-end data and voice networking solutions for businesses ranging in size from small- and medium-sized businesses to large enterprises.

16/08/2010 - Sonus Enables Cable Customers to Make and Receive Calls on Their Smartphone Using Their Home Phone Number

Sonus Networks has introduced a new solution for cable operators to add value to their existing home line services. Dubbed "Fixed – Smartphone Convergence," cable operators can use new capabilities in the company’s ASX Telephony Application Server to allow cable subscribers with home phone service to combine their existing phone line with up to five additional SIP-enabled devices including 3G/Wi-Fi enabled smartphones.

04/08/2010 - VoicePulse Announces SIP Trunking Interoperability with IPitomy PBX Products

VoicePulse and IPitomy announced that they have successfully completed interoperability testing between SIP products and services. VoicePulse is now interoperable with IPitomy’s Pure IP PBX platform.

04/08/2010 - Sipgate offering low cost business VoIP

Sipgate, a German company targeting UK-based SME's has launched Sipgate Team, a business web portal that gives customers access to VoIP telephony technology. Companies can use the web interface to configure their telephone service.

The service will offer some basic features that we've come to expect from VoIP service including call holding, forwarding, conference calling, up to 2 hours of call recording, as well as personal user inbox management for voicemail, fax traffic and SMS sending and receiving. Initially targeted at SME's of 100 employees or less, the company claims any number of extra virtual lines can be set up.

For more:
- read the Techworld article

04/08/2010 - VoicePulse Announces SIP Trunking Interoperability with IPitomy IP PBX Products

VoicePulse announces interoperability with IPitomy's Pure IP PBX platform. IPitomy designs and manufactures a complete line of IP telephony equipment with an advanced feature set for businesses. IPitomy's PBX combined with VoicePulse's SIP origination and termination services create a complete phone solution for businesses of all sizes.

VoicePulse can now provide new customers using IPitomy's IP PBX with official configuration guides when setting up VoicePulse VoIP services on their PBX. Businesses using IP PBXs such as Digium's Asterisk, AsteriskNow, Fonality's PBXtra, trixbox, FreePBX, FreeSwitch, Switchvox, and now IPitomy can benefit from VoicePulse's competitive international rates, toll-free services, and failover features all on a "Tier 1" back bone network.

21/07/2010 - XO Communications Debuts XO Enterprise SIP Savings Estimator

XO Communications has released a new Savings Estimator tool that gives enterprises the ability to calculate an approximate cost-savings benefit of utilizing XO Enterprise SIP. According to the company, the XO Enterprise SIP enables customers to “simplify, streamline and achieve better cost savings.”

20/07/2010 - XO Communications Debuts XO Enterprise SIP Savings Estimator

XO Communication announces the availability of a new Savings Estimator tool that gives enterprises the ability to calculate an approximate cost-savings benefit of utilizing XO Enterprise SIP. Designed specifically for multi-location enterprises, the award-winning XO Enterprise SIP enables customers to simplify, streamline and achieve better cost savings by transforming their distributed voice network architecture to a more centralized and cost-effective VoIP solution.

The XO Enterprise SIP Savings Estimator tool provides a snapshot of potential savings by factoring in the number of employees, network locations and intra-company long distance calls. The tool also takes into consideration the multiple cost-savings benefits of XO Enterprise SIP, including increased network management efficiency and lowered operating costs as a result of reducing equipment, local voice trunks, long distance and multiple voice and data network charges.

To access the XO Enterprise SIP Savings Estimator, click here.

XO Enterprise SIP
XO Communications is an industry leader in VoIP and SIP trunking solutions for distributed enterprises. The award-winning XO Enterprise SIP enables customers to utilize a centralized IP-PBX architecture in key locations and deliver VoIP services to branch locations across an existing wide area network or using the XO MPLS IP-VPN solution. Moreover, utilizing XO Enterprise SIP customers can achieve a number of benefits including:

Lower Total Cost of Ownership by using a single or fewer IP-PBXs to support all locations;
Reduced Operating Costs by not having to maintain costly PRI facilities or local voice trunks at each location, and eliminating the operating expense of managing separate voice and data networks;
Greater Flexibility by allowing locations to burst above their normal call capacity and sharing idle voice trunk capacity from other locations across the enterprise;
Increased Efficiency in network management through simplified and converged network operations, significantly less effort to connect new locations to the public switched telephone network;
Business Continuity with redundant Enterprise SIP connections and the ability to automatically re-route calls to alternate locations;
Extensive Nationwide Availability of XO VoIP services in all 50 states and more than 2,700 cities.

15/07/2010 - IPsmarx Wants to Revolutionize the Calling Card Industry

IPsmarx released their newest innovation for the prepaid phone card industry that is said to allow Calling Card and Pinless Service Providers to expand their network of sales agents, increase their customer base, and reduce operating costs.

14/07/2010 - Broadvox Announces SMB IP Multimedia Communications

Broadvox has announced that Grandstream Networks' new GXV3140 is the first IP Multimedia CPE device to deploy with Broadvox GO! SIP Trunking. According to the company, the solution enables businesses to save up to 70% a month over their previous TDM systems.

07/07/2010 - Ingate Systems Presents the New SIP Trunk-Unified Communications Summit at ITEXPO West 2010

Unified Communications is coming of age and it's being driven by the urgent demand for SIP trunking, which lowers operating costs and delivers rapid ROI. To address the need for information on the what, why and how of Unified Communications and SIP trunking, Ingate Systems is partnering with TMC, thought-leaders and vendors in the space to present the new SIP Trunk-Unified Communications Summit at ITEXPO West 2010.

Free to all ITEXPO attendees, the Summit is a three-day seminar series providing in-depth educational information on SIP trunking and Unified Communications. The Summit will include general information panels and technical insight sessions from both the service provider and enterprise perspectives, and will feature visionaries driving the industry.

To date, confirmed speakers include Dialogic, Iwatsu, Mitel NetSolutions, ShoreTel, The SIP School and VOIPSA. David Yedwab, a Founding Partner in Market Strategy and Analytics Partners LLC., contributor to Unified Communications Strategies and TMCnet columnist, will discuss the future for service providers in a UC-enabled world. Additionally, service provider Telia will present their SIP trunk implementation with Intertex Data AB as a case study. Joel Maloff of Maloff NetResults will moderate the panel discussions.

The SIP Trunk-UC Summit will be held October 4-6, 2010 at the Los Angeles Convention Center in Los Angeles, California.

"This season will feature a strong focus on ROI. The economy has forced decision-makers to take a hard look at new technologies, to invest in solutions that lower costs, improve productivity and are easy to implement. SIP trunking fits the bill," said Steven Johnson, President, Ingate Systems. "SIP trunking is a secure, cost-effective way for enterprises to adopt SIP, and it's the first step toward UC adoption."

Sessions this season will include:

SIP Trunking Professional Development Program

Building for ROI
Case Studies
Service Provider Perspective - US and European Views
How to Sell SIP
Town Hall Meeting: SIP, UC and Security
Unified Communications Day

Fax-over-IP

Legacy PBX/PSTN and SIP Trunking
The Future for Service Providers in a UC-Enabled World
The ROI of SIP Trunking and UC
SIP Trunk Boot Camp

Deploying SIP Trunks - Getting it Right the First Time

SIP Trunk "Basic Training" with Ingate
Live demonstrations, during which participants will set up a SIP trunk in 20 minutes on-site, will also be part of the program.

Attendees can earn a SIP Trunking Professional Certificate by participating in the Professional Development Program on the first day of the Show.

"Ingate's SIP Trunk-UC Summit series is a unique opportunity for ITEXPO attendees to get a crash course on SIP trunking from the experts," said Rich Tehrani, CEO and group editor-in-chief for TMC, hosts of ITEXPO. "We are proud to once again work with Ingate Systems in providing these educational seminars at ITEXPO."

23/06/2010 - Yealink Release New Firmware for SIP-T2x Series IP phone

Yealink network, a manufacturer of IP voice and video phone, announced that it has released the latest firmware for its award winning IP phone series--SIP-T2x. They provide high quality audio, a broad range of voice codecs, security protection for privacy, and rich telephony features.

09/06/2010 - IPsmarx Partners with snom to Deliver VoIP Services for SMBs and Global Enterprises

snom announced that IPsmarx, maker of the IPsmarx’s Multi-Tenant IP-PBX Platform, has joined snom’s partner program as an Advanced Technology Partner. According to the companies, the combination of snom’s desktop phones with IPsmarx’s IP-PBX provides service providers who cater to small and medium sized businesses, as well as larger global enterprise businesses, “a complete and affordable advanced IP telephony system.”

08/06/2010 - Acrobits Launches New Free Service for SIP Providers

Acrobits has launched a new free service which allows SIP providers to appear on the list of pre-configured providers in Acrobits Softphone. The company also informed that it has released the new 3.2.3 version of its Softphone.

26/05/2010 - Virtual PBX Opens VoIP Peering Service to All SIP Phones

Virtual PBX has expanded its Open VoIP Peering service to include support of any SIP-compliant phones. Users of any SIP-compliant softphone or desktop phone can now tap into the benefits of the Virtual PBX Open VoIP Peering service. This service allows small businesses to take advantage of the robustness and features of traditional hosted PBX services and leverage the cost-effectiveness of VoIP technology. Today’s announcement allows customers more freedom to mix and match SIP-compliant phones with Virtual PBX’s hosted PBX services.

Breaking Down Proprietary Barriers for Users

Hosted IP-PBX providers that support native VoIP traffic usually have closed proprietary implementations. Customers have usually been locked into VoIP, softphone and even telephone handset offerings from the same company that provides the PBX service. Virtual PBX is giving its users more choice by supporting open standards. Rather than being limited to the features and capabilities of just a single vendor, clients can choose the offering that best fits their needs. Virtual PBX believes this will ultimately provide richer services for clients, since each player can bring unique content and features to the table. At the same time, total costs can be cut by sending more traffic through less expensive technologies. There is no extra cost from Virtual PBX for accessing softphones as opposed to other players who charge extra for proprietary digital-enabled phone extensions.

Up until this announcement, Virtual PBX’s VoIP peering implementation worked only with VoIP service that used the North America Numbering Plan to address phones. Now, customers can choose any phone that complies with established SIP standards, including products from top vendors like Truphone, Callcentric, Ekiga and Gizmo5, giving full freedom of choice in price, features and vendors.

Reducing Costs

Hosted IP-PBX services provide businesses with phone systems without the high costs of installing and maintaining PBX equipment. Typical prices range from $25 to $60 per user per month, averaging about $40 per user. Virtual PBX’s VoIP Peering product, the iVPBX plan, reduces that price to just $10 per user. The iVPBX product includes all standard Virtual PBX features, such as a virtual receptionist, ACD queuing, voicemail, conferencing and follow-me forwarding.

21/05/2010 - Mobile VoIP Becomes a Threat to Tradicional Voice Revenues

Mobile VoIP is no longer just hype, but has become a credible threat to traditional voice revenues, says Frost & Sullivan. According to the research group, an ambitious group of mobile VoIP start-up companies are creating a paradigm shift in the way users communicate with each other, with voice services moving to a true internet era of Telco 2.0.

20/05/2010 - Sipera raises another $10M to advance UC security

UC sercurity company, Sipera Systems raised another $10M to advance its enterprise unified communications (UC) security systems. Their offering provides security for smartphone, VoIP, IM and video applications.

A new investor, S3 Ventures, led this particular round of funding while previous investors Austin Ventures, Duchossois Technology Partners, Sequoia Capital, and STAR Ventures all chipped in some more cash. Sipera's funraising total now stands at $48 million. The new cash will allow the company to advance its smartphone UC security solutions and UC-Sec enterprise UC security product family--which will help executives use iPhone and other smartphones securely. Sipera is known for demoing how easy it is to hack those devices at various trade shows.

"Sipera is at the forefront of Unified Communications security solutions, including most recently protecting smartphones to enable safe, enterprise-class VoIP and UC for employees on the go. Given the explosive enterprise adoption of smartphones and UC applications including VoIP, IM and video, the timing is ideal for Sipera to capitalize on its leadership in comprehensive UC security," said Brian R. Smith, Managing Director of S3 Ventures--one of the newest investors in the company.

Back in April, Sipera announced that they were securing over a million devices. In addition, they achieved 300 percent year-on-year growth in demand as of the first quarter of 2010, marking 2,400 percent growth in demand over 24 months.

For more:
- read the release

10/05/2010 - VoX Communications Completes Development of Its Symbian Mobile VoIP App

VoX Communications has completed development of a downloadable mobile VoIP application for use on Symbian mobile phones.

According to Gartner, 80 million Symbian devices were sold worldwide in 2009, or approximately 47% of the entire global smartphone market. Symbian devices are so dominant in the marketplace today, that it would take the combined sales of the iPhone, RIM, Windows Mobile and Android to equal the same volume opportunity. VoX estimates that its app can be downloaded to more than 100 million Symbian mobile phones that are currently in use. Gartner also projects that Nokia will sell in excess of 100 million Symbian phones in each of the next 3 years.

07/05/2010 - Verizon Boosts Functionality and Security for Selected VoIP Services

Verizon Global Wholesale is now offering additional functionality for three of its product platforms -- SIP Gateway Service, Advanced Toll Free IP Termination and Carrier IP Termination Transport. This new functionality enables the product platforms to share a single, secure virtual private network connection called an IP Security (or IPSec) tunnel.

06/05/2010 - Winning VoIP Philosophy: Interview with Arash Vahidnia, CEO of IPsmarx

“VoIP penetration among businesses is increasing rapidly, but I believe that reliability and redundancy can still be improved. Many service providers have not yet implemented redundancy to improve reliability,” said Arash Vahidnia, President and CEO of IPsmarx, in an interview with VoIP.Biz-News.com.

05/05/2010 - UM Labs Announces Skype for SIP Support

UM Labs announces support for Skype for SIP. This means that Organisations can now use the UM Labs SIP Controller to simplify the task of linking their corporate PBX to the Skype for SIP service. Skype for SIP offers a number of compelling benefits, including free inbound calls from any Skype user and low cost local numbers for inbound and outbound calls in over 25 countries. A Skype for SIP link requires that the corporate PBX accepts inbound SIP calls over the Internet. The UM Labs SIP Security controller monitors these calls protecting the corporate PBX from a wide range of security threats, many of which are not blocked by a standard Firewall.

UM Labs SIP Controller's current product release 1.4 has been successfully tested - inbound and outbound calls - with Skype's SIP beta program. Current customers can download a Skype for SIP application note from the UM Labs website.

The UM Labs SIP Controller provides significant benefits to any enterprise looking to Skype-enable their PBX. These include:

The UM Labs controller simplifies the process of setting up a Skype for SIP connection. Its service enablement features ensure that SIP requests are presented in exactly the format that Skype needs.
The UM Labs controller means that a company with any SIP capable PBX is now able to connect to Skype for SIP even if that PBX cannot itself meet Skype's connection requirements.
Skype for SIP requires that the company's PBX is connected to the internet, so the SIP specific security provided by the UM Labs product is essential. Standard firewalls cannot block the wide range of SIP security threats that can compromise a corporate PBX.

UM Labs is a pioneer and leader in Voice over IP and Unified Messaging security. The company markets a family of cost effective SIP Security Controllers that easily and securely connect VoIP systems used by enterprises, government, and communications service providers to the public internet. Confidentiality, integrity, and authenticity of VoIP are critical considerations for most businesses. Driven by lower bandwidth costs and the promise of increased flexibility, VoIP is quickly becoming a critical tool in the business-to-business landscape. Significant growth in SIP

Trunking and consolidation of voice and data traffic over the public internet are raising new security and interoperability concerns that were previously overlooked. To solve these problems, UM Labs has developed a family of cost effective SIP Security Controllers which can be easily plugged into existing networks to enable SIP connectivity, security and voice encryption.

05/05/2010 - XO, TMC Net and Broadsoft to Host Webinar: ''Beyond SIP Trunking - Unify the Enterprise''

XO Communications in collaboration with TMCNet and BroadSoft, will host an interactive webinar on May 12th entitled, "Beyond SIP Trunking - Unify the Enterprise." The event will bring together several SIP trunking experts to discuss the latest industry trends and insights on how distributed enterprises can streamline their managed voice services.

Specifically, the hour-long session will cover topics such as the benefits of SIP trunking services and future trends, how to evaluate the return-on-investment of SIP, as well as things to consider when evaluating SIP trunk service providers. It will also offer an inside glimpse of various enterprise networks before and after SIP trunking was implemented.

The webinar will feature the insights and perspective of leading SIP trunking experts, including:

Lisa Pierce, founder and president of Strategic Networks Group, a consultancy dedicated to improving the quality of emerging telecommunications and IT services, and the service experience that business customers receive from key suppliers. She brings a unique expertise in a wide range of network technology including SIP trunking, unified communications, broadband access, managed network services, among other things.
Steve Carter, senior product manager at XO Communications, led the launch of the company's SIP trunking services, including XO Enterprise SIP, which won the INTERNET TELEPHONY Product of the Year Award in 2009.
Alex Doyle, senior director of global solutions marketing at BroadSoft, has been intricately involved in the development of the BroadWorks application platform and has been responsible for product solution management of BroadSoft's business and consumer applications.
Erin Harrison, senior editor of TMCnet, is a seasoned reporter and editor and covers IP communications, information technology and other related topics for TMCnet.

Webinar Details: Beyond SIP Trunking - Unify the Enterprise When: May 12th at 2:00 p.m. EDT Who Should Attend: Both technical and non-technical audiences from U.S.-based businesses Registrations: Complimentary Webinar - Click here to register for the webinar.

XO Enterprise SIP XO Communications is an industry leader in SIP trunking solutions for distributed enterprises. Its newest solution, XO Enterprise SIP, enables customers to utilize a centralized IP-PBX architecture in key locations to deliver VoIP services to branch locations across an existing wide area network or using the XO MPLS IP-VPN solution. Moreover, utilizing XO Enterprise SIP customers can achieve a number of benefits including:

Lower Total Cost of Ownership by using a single or fewer IP-PBXs to support all locations;
Reduced Operating Costs by not having to maintain costly PRI facilities or local voice trunks at each location, and eliminating the operating expense of managing separate voice and data networks;
Greater Flexibility by allowing locations to burst above their normal call capacity and sharing idle voice trunk capacity from other locations across the enterprise;
Increased Efficiency in network management through simplified and converged network operations, significantly less effort to connect new locations to the public switched telephone network;
Business Continuity with redundant Enterprise SIP connections and the ability to automatically re-route calls to alternate locations;
Extensive Nationwide Availability of XO VoIP services in all 50 states and more than 2,700 cities.

04/05/2010 - Panasonic Announces Digium Asterisk Certification for Its New TGP500 Series SIP Cordless Phone System

Panasonic announces Digium has certified the Panasonic line of SIP Cordless Phone Systems, the KX-TGP500 and KX-TGP550, for use with the Asterisk telephony platform. By leveraging Panasonic's market leadership in DECT cordless telephones and Digium's innovative open source PBX, this alliance is a winning combination.

The Panasonic TGP500 series of SIP Cordless Phone Systems boasts outstanding voice quality and a range of productivity-boosting options in a feature-rich 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.

Digium is the creator, sponsor and driving force behind Asterisk, the most widely used open source telephony software. The company's product lines include a wide range of software and hardware that enable businesses to implement turnkey unified communications solutions or to design their own VoIP systems. Resellers, telecom professionals and software developers choose Digium's products because only Digium delivers the technical superiority, security and flexibility associated with Asterisk.

With flexible configuration options, it has never been easier to deploy and expand a SIP-based phone system. Handsets can be easily added wherever needed; all that is required is an electrical outlet nearby. 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 TGP500 series SIP phone system, it is quick and easy to add up to 6 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.

TGP500 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, 10 days Standby. Its elegant design features a white backlit large LCD on the handset and a Handset locator button on the base unit. It also has a handset speakerphone, 2.5mm headset jack and belt clip.

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, 10 days Standby, plus a Hands-Free Speaker phone, Handset Call Button on the base unit, and one-touch call transfer with Busy Lamp Indication.

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

The TGP500 Series will be available in May 2010 through the following Panasonic authorized distribution partners: Jenne Distributors, NETXUSA and D&H Distributing.

03/05/2010 - Top Trends in Videoconferencing

No Jitter has an interesting roundup of the top trends they see in Videoconferencing. As you might have noted in today's issue, SIP seems to be one of them. Read the rest here...

21/04/2010 - CounterPath Releases Network-Based Mobile Mashup Application

CounterPath announced the launch of NomadicPBX, claimed to be the world’s first turnkey platform for enabling converged mobile and broadband SIP voice, messaging and presence services.

The application is a presence-based, fixed and mobile voice, and instant messaging/short message service (IM/SMS) technology mashup with a select set of enterprise-ready features.

13/04/2010 - SIP Forum’s SIPit 26 to Take Place in Kista, Sweden, May 17-21, 2010

The SIP Forum has announced that its next SIP Interoperability Testing event, SIPit 26, will be held May 17-21, 2010 in Kista, Sweden. The event will be hosted by Edvina and TANDBERG, and sponsored by Intertex, Ingate, .se, and the IPv6 Forum as an association sponsor.

Conducted by the SIP Forum twice a year, SIPits are the world’s premier interoperability testing events for the SIP, bringing together the leading SIP application developers, service providers and IP communications equipment manufacturers to ensure their SIP implementations work seamlessly together in an IP network testing environment. An important goal of the SIPit events is to help refine both the SIP protocol and its implementations in order to further establish SIP as a global interoperable standard for real-time Internet communication services.

"SIPit 26 brings together equipment vendors and service providers across the global IP communications industry to test and validate their SIP implementations in a live, real-world IP network setting in Kista, Sweden, a center of telecom technology in the suburbs of Stockholm,” said Marc Robins, SIP Forum President and Managing Director. "In addition, this spring event will also give companies the chance to test specifications from standards bodies such as the IETF, as well as recommendations formulated by SIP Forum task groups. A goal is to reinforce the progress that these groups are making to further establish the SIP protocol within the service provider, consumer and enterprise network environments.”

SIPit is organized by the SIP Forum’s Test Event Working Group and serves as a “plugfest” for participating companies to perform SIP interoperability testing with other participants in a live network environment. Conducted twice a year, with events rotating in the United States, Europe, and Asia, the SIP Forum has hosted 25 plugfest events around the globe. The previous event, SIPit 25, was hosted by the University of New Hampshire Interoperability Lab in Durham, New Hampshire in September 2009.

“The SIPit events are extremely effective testbeds, both for implementations and for specifications,” said Robert Sparks, chair of the SIP Forum’s Test Event Working Group. “We frequently have participants indicate that a week spent at SIPit provides results that would have taken months to achieve with individual pair-wise testing. Information about the state of implementation and interoperability of the specifications has been very useful in informing ongoing standards work.”

"Interoperability is the foundation for the TCP/IP protocol suite. As we move to realtime communications, it's important to test interoperability in all the platforms that are developed. For us in the Asterisk.org Open Source development group, it's been really important to participate in SIPit, where we can test, learn and improve our solution in a very open and friendly setting. SIPit is an important part of the success of the IETF Session Initiation Protocol," says Olle E. Johansson, founder of Edvina and developer of Asterisk. "We're proud to host this event in partnership with TANDBERG and thus become part of the success story of SIPit."

30/03/2010 - ABP Ships Patton SmartNode VoIP SBCs, Pre-Configured for SIP-Trunking Providers

ABP Technology is now delivering pre-configured units of the SmartNode 5200 Enterprise Session Border Router from Patton. Leveraging Patton’s SN5200 ESBR, ABP delivers fully-provisioned, tested, and 100% quality-assured SIP-trunking solutions that help Internet Telephony Service Providers reduce start-up costs and focus on core revenue-generating activities such as subscriber acquisition.

By pairing state of the art provisioning services with carrier-grade SmartNode Session Border Controller equipment, ABP and Patton enable ITSPs to attract new business subscribers and increase profits by delivering survivable, value-added SIP-trunking services.

In the recently announced strategic alliance, ABP ensures timely, hassle-free delivery of ready-to-deploy Patton VoIP equipment throughout the Americas and Caribbean by providing order fulfillment, pre-configuration and provisioning services.

Patton’s SmartNode 5200 SBC lowers overall equipment costs by including an advanced IP router, QoS, VoIP-VPN security, least-cost call routing and IP-link redundancy—with no added licensing or support fees.

With trained Patton Certified SmartNode Specialists on staff, ABP offers pre and post-sales technical support for assured interoperability with third-party network elements.

The VoIP industry recognizes—and businesses and carriers demand--the US-made quality, maturity, and technical innovation in award-winning SmartNode VoIP equipment.

23/03/2010 - Grandstream Now Skype for SIP Interoperable

Grandstream, a manufacturer of IP voice/video telephony and video surveillance solutions and Skype announced that Skype for SIP interoperability testing and certification of Grandstream's IP PBX and gateway has been completed successfully.

22/03/2010 - Skype Selects tekVizion Labs to Manage Skype for SIP Interoperability Certification Program

Skype announces it has selected tekVizion Labs as the independent and objective verification and testing service to run the Skype for SIP Interoperability Certification Program. In addition, Skype and tekVizion Labs together announced a number of third-party IP PBX and gateway solutions to have been newly awarded Skype for SIP interoperability certification.

The following are some of the recent IP PBXs and gateways for small- and medium-sized businesses to have passed tekVizion Labs’ testing and been certified as interoperable with Skype for SIP:

AudioCodes Mediant 1000 Multi-Service Business Gateway: This gateway offers the ultimate flexibility and modularity, supporting the use of Skype for SIP with many types of TDM PBXs, Key Systems and IP PBXs. It can be integrated with a company’s routing, firewalling, enterprise session border controllers and application hosting capabilities..
Grandstream GXE502x IP PBX: This all-in-one IP PBX delivers a converged business communication solution to SMBs, especially those companies with less than 30 seats per location.
Grandstream GXW4xxx Analog Gateway: Designed for compatibility with traditional analog PBX/key systems, as well as for full interoperability with leading IP-PBXs, softswitches and most SIP-based environments, this gateway will enable SMBs to integrate traditional communications systems with Skype for SIP and efficiently manage communication costs.
VoSKY SSG: This PBX-to-Skype for SIP gateway will enable SMBs to connect any legacy analog PBX or key system to Skype.

SMBs can now take advantage of the cost savings provided by Skype’s low-cost global calling rates when employees call landlines and mobiles around the world through their IP PBX, legacy PBX or key system or UC solution that has been certified as Skype for SIP interoperable. A company can also receive inbound voice calls from any of the more than 521 million registered Skype users around the world via a global click-to-call button on its Web site. These Skype calls are received by the existing PBX or UC system and are handled or directed in the same way as any other inbound caller. If a company buys and associates online Skype numbers with their interoperable PBX or UC system, it can also receive inbound calls through Skype from business contacts and customers calling from landline and mobile phones.

Hardware or software vendors interested in scheduling Skype for SIP interoperability testing for their IP PBXs and gateways or UC systems should visit the tekVizion Labs Web site. Businesses interested in learning more about Skype for SIP should visit www.skype.com/business/products/pbx-systems/sip/.

22/03/2010 - Vonage wins long distance calling virtual phone number patent

Vonage has been awarded patent No. 7,680,262 aka the "Method and Apparatus for Placing a Long Distance Call Based on a Virtual Phone Number." What does that mean in non-patent speak? Basically they have patented a method of making long distance phone calls using SIP and virtual phone numbers for the price of local calls.

According to the release: "Virtual phone numbers allow people outside a Vonage customer's local calling area to call for the price of a local call. Vonage customers can choose a virtual phone number in the calling area of a parent, relative or friend or wherever they want to establish a "virtual" presence."

VoIP Watch did a quick hit post to explain the move saying that they've basically figured out how to do what the Foriegn Exchange number did for PSTN calling, but for a far lower price. You have a local number in one place, but you are able to get calls on it far away.

For more:
- read the release
- read Andy's quick take on VoIP Watch

17/03/2010 - IPsmarx Introduces IP-PBX And SIP Trunking To Its “All In One” Solution

IPsmarx has just announced the addition of a Multi-level IP-PBX and SIP trunking to its integrated VoIP softswitch/billing platform.

According to the company, it’s a “complete solution for competitive service providers” to offer the portfolio of services demanded by the Small Mediums Business market.

16/03/2010 - AudioCodes' SIP Phone Support or Microsoft Unified Communications

AudioCodes is extending its product range for customers seeking access to the Microsoft Unified Communications environment with the introduction of AudioCodes' SIP Phone Support for Microsoft Unified Communications. AudioCodes' SPS allows economically affordable connections to Microsoft Office Communications Server via standard SIP IP Phones. A direct connection to Office Communications Server is supported both for currently installed phones and newer models, as well as connecting mobile phones using AudioCodes' Mobile Clients, utilizing AudioCodes' gateway technology.

Enterprises migrating into the Microsoft Unified Communications environment have been showing great interest in enabling cost-effective IP Phones to connect to Office Communications Server. According to Synergy Research, the current installed base of IP Phones in the market is approximately 65 million. By 2014, around 20 million new IP Phones will be sold annually. Most of these phones have not had direct access to Microsoft Office Communications Server. Gartner Magic Quadrant for Unified Communications 2009 stated that "Enterprises looking into UC, particularly those with Microsoft applications already in place, should understand the Microsoft portfolio, because it represents a new paradigm for communication by a market leader. Microsoft's solution, while comprehensive, is also the basis for a range of partner offerings." Based on the Microsoft leadership and customers' demand for partner solutions, customers can now protect their existing investment in third-party IP-Phones, while enjoying the full benefits of the Microsoft infrastructure, including unified call control, integration with their Microsoft Office Communicator, presence information and more.

The IP phones supported by SPS include AudioCodes' 300HD family of High Definition IP Phones, as well as other third-party IP Phones, such as Cisco, Avaya, Aastra, Polycom and other standards-based SIP phones. The mobile smart phones support is enabled using AudioCodes' Mobile SIP clients, supporting all major smart phones' operating systems, including Windows Mobile, iPhone OS, Symbian and Android.

AudioCodes' 300HD family of IP Phones offer integrated, DSP-based support of Microsoft Real Time Audio codec and the support of Secured Real Time Protocol and SIP over TLS, enabling maximum security, high definition calls between IP Phones and Office Communicator clients. In addition, AudioCodes' phones provide enhanced presence and Active Directory support in the phone user interface, offering enhanced user experience for the Microsoft clients.

Existing customers of AudioCodes' Mediant gateways can now upgrade their existing installed base to support SPS. The same gateway can support the SPS functionality while providing a Media Gateway function between Microsoft Office Communications Server and/or Microsoft Exchange Server and the enterprise TDM PBX and/or the PSTN. SPS can scale from small demos of a few users all the way up to supporting enterprise installations of thousands of IP Phones and SIP Mobile Clients. By optionally integrating it with the different AudioCodes' Mediant Media Gateways it can scale from 120 concurrent RTP to SRTP calls on the Mediant 1000 all the way up to 884 calls on the fault-tolerant Mediant 3000.

15/03/2010 - IPsmarx SIP-Based Calling Card Platform Wins “Best VoIP Product of the Year”

March 10th, 2010 – IPsmarx, a leading provider of VoIP application and switching solutions for service providers, has just been awarded ‘Best VoIP Product of the Year’ for their SIP-based calling card platform. This is the second time in as many years that IPsmarx was selected a winner by the readers of VoIP.Biz-News.com for their experiences with the product, as well as the vendor.

05/03/2010 - MWC 2010: Interview with Johan Lantz of Genaker

Genaker focuses on development of state-of-the-art solutions based on SIP, they are R&D Company that aims to replace the legacy of the walki-talki devices with modern technology communicating over cellular networks.

04/03/2010 - Winner of the Biz-News.com "Product of the Year Award 2009” Announced

Our polls for the Biz-News.com “Product of the Year Award 2009” closed on the 15th of February. The winner is a result of the amount of votes they were awarded by readers, all readers where invited to vote for their favourite products or service in the Smartphone, HDTV, Storage and VoIP categories.

03/03/2010 - REDCOM Successfully Completes SIP Interoperability Testing with Polycom IP Phones

REDCOM has completed SIP interoperability testing for REDCOM’s HDX and SLICE 2100 Carrier-Class 4/5 softswitches with a wide range of SoundPoint IP phones from Polycom.

REDCOM tested several Polycom SoundPoint IP models (320/321/330/450/550/560/650/670), as well as the VVX 1500 Business Media Phone for verification in REDCOM’s lab to ensure that the company’s HDX and SLICE 2100 softswitch solutions are fully interoperable. REDCOM successfully provided rigorous and comprehensive SIP interoperability testing in its company lab in Victor, which verified 100 percent integration and functionality between the Polycom VoIP phones and REDCOM’s HDX and SLICE 2100 SIP call control platforms.

Based on this verified SIP interoperability between the Polycom IP handsets and SIP-based REDCOM HDX and SLICE 2100 softswitches, commercial customers will be able to deploy these best-of-breed VoIP solutions with confidence and without compromising functionality. REDCOM’s HDX and SLICE 2100 are powerful softswitches and SIP call managers that retain complete Class 4/5 capabilities in a single platform solution.

TRANSip, the breakthrough technology suite behind REDCOM’s HDX and SLICE 2100, is designed to provide a complete, integrated VoIP-TDM solution, whether your business strategy requires new market development, a VoIP network overlay or a migration path to Next Generation services. By coupling enhanced VoIP capabilities with the reliable functionality of your existing TDM network, TRANSip helps service providers to respond to consumer demands for Next Generation services while they manage their capital investments.

23/02/2010 - Interact Incorporated Announces Interoperability with the BroadvoxGO! SIP Trunking Product Line

Interact Incorporated are pleased to announce that Broadvox has certified version 6.14 of Interact’s SPOT Media Platform for interoperability with the BroadvoxGO! SIP Trunking product line. The SPOT VoiceXML/CCXML Media Platform completed extensive testing and proved to be successful in all aspects.

Deployable in VoIP, SIP, TDM or PSTN environments or on a variety of hardware platforms, Interact’s VoiceXML/CCXML Media Platform, SPOT, provides feature rich and high density media processing with call control signaling, IP and PSTN connectivity. This allows operators worldwide to introduce and deliver new and innovative voice and data service offerings, including interactive voice response, voice portals, conferencing services, voicemail, unified messaging platforms and prepaid services to meet ever changing market demands.

Broadvox performs the certification process under rigorous conditions encompassing key elements of interoperability to ensure “real world” system operation. The core SIP curriculum testing comprises over sixty separate elements involving key aspects of call processing, both inbound and outbound.

22/02/2010 - Patton Taking Orders for the SmartNode 5200 Enterprise Session Border Router

Patton is now taking orders for the SmartNode 5200 Enterprise Session Border Router. Providing "any-to-any" protocol mediation for secure connectivity between IP-PBX systems and Internet Telephony Service Provider, the newest SmartNode resolves security and interoperability challenges businesses face when implementing SIP trunking.

The SmartNode ESBR lowers overall equipment costs by including an advanced IP router, QoS, VoIP-VPN security, least-cost call routing and IP-link redundancy—with no added licensing or support fees.

While most SBC vendors tack on per-feature licensing fees and charge for support contracts, Patton includes all SBC features and functions—along with free tech support—in the product base price.

Infonetics' predicts 89 percent revenue growth for SIP trunking services by 2013.

Although minor variances between vendor's SIP implementations can disable enterprise VoIP systems, protocol mediation in the SN5200 ensures operability among all SIP "flavors" for fast, easy SIP trunking setup.

With a built-in virtual firewall that combines ACLs with policy-based routing, the ESBR provides an ideal security solution for enterprises with up to 50 people.

For business with 50 to 250 workers, the SN5200 delivers secure VoIP communications by processing forwarded SIP signaling messages to "SIP-enable" an external stateful firewall.

Patton's white paper library offers in-depth coverage of SIP trunking, VoIP-VPN security, and VoIP QoS.

During Q2, watch for Patton's SN5400 ESBR, featuring a transcoder that maximizes voice quality by converting low-bandwidth ITSP CODECs to high-bandwidth CODECs within the LAN.

16/02/2010 - OnSIP to Leverage Both SIP and XMPP for Complete Unified Communications Offering

Onsip - an affordable small business phone system

Junction Networks announces that it supports Extensible Messaging and Presence on any software or phone, enabling OnSIP customers to take advantage of a complete Unified Communications offering, regardless of the device they use to access it.

OnSIP customers now have access to a complete Unified Communications experience, with no additional cost on any client (phone or software application) supporting these open protocols. For example, the entire OnSIP Unified Communications experience is accessible using Bria, by Counterpath, a leading software application which supports both XMPP and SIP.

"Everyone knows what Skype does; Voice, Video, IM and Presence over the Internet on a Skype software download," said Rob Wolpov, President, Junction Networks. "For businesses, OnSIP goes further, adding a comprehensive package of business services with extensions, auto-attendants, voice mailboxes, conference bridges and much more. And now, our support for these services extends beyond any proprietary software or phone."

"OnSIP now is a comprehensive business communications suite with hosted Voice, IM, Video and Presence services delivered reliably over the Internet to any single or multi-location organization," said Andy Abramson, author of VoIPWatch (http://www.andyabramson.com). "As someone who has been using the service since its launch, the quality and consistency can't be beaten."

"CounterPath's market approach has always been to provide telephony solutions based on SIP and open standards. This allows our technology to be interoperable with a large number of platforms and devices", said Todd Carothers, VP Product Management for CounterPath. "Enabling service solutions like OnSIP builds on the ecosystem of open Application Programming Interfaces (APIs) and plug-and-play options for both business and consumer users, ultimately providing them with more options to enrich their UC experience."

OnSIP recently added HD Voice calling and conference calling features to its existing suite of services. Plans start as low as $39.95 per month.

28/01/2010 - Acrobits Softphone Enables SIP VoIP Calling Over 3G

Acrobits adds the ability to make calls over 3G or Edge networks to Acrobits Softphone, the leading SIP Softphone for the iPhone and iPod Touch. Users will now be able to use the softphone to make or receive calls even when no Wi-Fi connection is available. Combined with Acrobits’ recent addition of universal support for Push Notifications, this is great news for the ever-expanding world of VoIP.

Now that Acrobits Softphone works over 3G, SIP users with an iPhone have a truly portable softphone. You can now make calls with your VoIP account anywhere you have a 3G or Edge connection. And since Acrobits’ Push Notification service allows you to receive calls when the softphone is closed, you can receive calls anywhere you have a 3G or Edge connection as well. “We believe adding 3G capability puts Acrobits Softphone at the forefront of the integral mobile VoIP market,” says Acrobits.

Acrobits has consistently improved Acrobits Softphone to keep up with customer’s needs and the iPhone’s capabilities. They use this same dedication on their white label softphone clients. This has put them at the top of the iPhone VoIP market, and may eventually make them a name in the larger VoIP world.

25/01/2010 - snom Makes Broadcasts Possible From VoIP Phone

snom, a developer and manufacturer of IP phones, has developed a new audio device that will allow SIP-based VoIP telephones to be used as an extension of any public address system.

20/01/2010 - 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.

19/01/2010 - 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.

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.

15/01/2010 - IMG Border0 Hspace6 Altsipforumjpg Alignright Srchttpwwwvoipmonitornetcontentbinarysipforumjpg Width233 Hei

The SIP Forum announces the recent re-election of industry veteran and VoIP pioneer Richard Shockey to the Board of Directors, and the election of Mr. 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.

Meanwhile, the SIP Forum announced the reappointment of Marc Robins as Managing Director and President.

The elections were held during the SIP Forum’s Annual General Meeting in San Francisco, on Nov. 3, 2009.

The elections of Shockey, Johnston and Burger, all industry trailblazers, complement the SIP Forum’s already prestigious Board of Directors, which also includes Chris Gatch, CTO of Cbeyond; Steven Johnson, CEO of Ingate Systems; Glenn Russell, Director of Business Services at Cable Television Laboratories Inc.; Rene Sotola, a Vice President in the Global Telecommunications practice at CGI Group; and Robert Sparks, Principal Engineer at Tekelec. Each member of the Board of Directors serves a two-year term.

Chairman Richard Shockey, founder of Shockey Consulting, is a widely respected 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. He is a founder and was co-Chair of the IETF ENUM Work Group and is author of several IETF RFCs. He has also authored numerous technical articles on SIP-based Next Generation Network technologies for a plethora of publications. Additionally, Mr. Shockey 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. In addition, Mr. Shockey was a Distinguished Member of the technical staff at NSR.

“I am honored to have been elected the new Chairman of the Board,” said Richard Shockey. “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.”

Rejoining the SIP Forum Board of Directors, Dr. Alan Johnston brings nearly two decades of valuable industry experience. Dr. Johnston 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. He served as an architect on the first enterprise SIP VoIP product in the U.S. as a Distinguished Technical Member at MCI. Dr. Johnston is currently a Consulting Member of the Technical Staff of Avaya Inc., a global leader in business communications applications, systems and services. 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. Along with his protocol and technical work, Dr. Johnston is the author of four books discussing SIP, including the best-selling SIP: Understanding the Session Initiation Protocol, Understanding Voice over IP Security, SIP Beyond VoIP, and Internet Communications Using SIP. Dr. Johnston has also served as co-chair of the IETF (Internet Engineering Task Force) Centralized Conferencing Work Group.

“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 Alan 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.”

Dr. Eric Burger, CTO of Neustar Inc., brings a wealth of knowledge and experience to the SIP Forum Board of Directors. Dr. Burger was re-elected to the SIP Forum Board of Directors after two successful terms as Chairman, and recently named SIP Forum Chairman Emeritus. Dr. Burger has been an active contributor on a host of IETF protocols, including SIP, SIPPING, SIMPLE, LEMONADE, SPEECHSC and VPIM. He currently chairs various IETF working groups involving Speech Services Control, Mobile and Unified Messaging and Media Server Control. He is also active within the Institute of Electrical and Electronics Engineers and the Association of Computer Machinery. Dr. Burger also sits on the Board of Advisors for Mobera Systems, AGNITY, Sigma Systems and Dexrex LLC. Dr. Burger holds 17 published U.S. patents, and has been published in myriad esteemed technical and standards publications.

“I look forward to my continued participation in the SIP Forum Board of Directors, and in supporting the vital mission of the organization,” said Dr. Eric Burger. “The SIP Forum’s role as a champion of technical interoperability for the IP communication industry is more important today than ever before and I am very excited to stand on the front lines bringing the charge forward!”

Along with electing board members, the SIP Forum also reappointed Marc Robins as its Managing Director. As Managing Director and President of SIP Forum LLC, Robins brings more than 27 years of relevant industry experience, including positions as a reporter, analyst, editor, author, trade show producer and magazine publisher. Robins has also served the telecommunications industry as a marketing executive and a consultant and is founder and Chief Technology Evangelism Officer of Robins Consulting Group, an IP communications industry consultancy.

“I am honored by the continued vote of confidence by the SIP Forum’s Board of Directors, and I look forward to continuing to serve the organization, and by extension the entire IP communications industry,” said Marc Robins. “There is no shortage of important work before us, and I look forward to working with Richard Shockey and the entire Board of Directors in expanding our activities and membership ranks over the next year.”

Together, Robins, Shockey, Johnston and Burger, along with other members of the SIP Forum and its Board of Directors, will leverage their years of experience and expertise to advocate and champion new IP communications applications and technologies and evangelize the use of the SIP standard to advance communications capabilities.

14/01/2010 - Skype for Business brings on new talent

As Skype gets more serious about its business VoIP offerings, it is bringing on new talent. David Gurle will be taking over for Stefan Oberg as the new General Manager and Vice President of the Skype for Business unit.

Gurle came to Skype from Thomson Reuters where he served as Global Head of Collaboration Services and Head of its largest business in Asia, the Sales & Trading Business Division. He was responsible for the company's business and product development in the Collaboration Services business and financial services community's preferred collaboration tool Reuters Messaging. Before Thomson Reuters, he ran Microsoft's Real Time Communications business for over three years. Prior to that he worked at IP telephony pioneer VocalTec.

For more:
- read this article

14/01/2010 - Windstream launches new VoIP/SIP service

Windstream is launching a new VoIP and SIP Trunking solution called Dynamic Office-SIP.

The tier 2 telco's business services solution will combine voice, data, high-speed Internet with IP communications using SIP to provide cost-savings over legacy phone systems. The new IP communications offering provides customers with access to Windstream's private IP network allowing customers to exchange voice traffic over an Internet connection without having to purchase a Primary Rate Interface (PRI). The service also provides special support for remote workers.

Launching new services comes as no surprise as Windstream has continued to scale its company through many small acquisitions over the last year. Recently, they snapped up smaller telcos D&E Communications, Lexcom and Iowa Telecommunications as well as one CLEC called NuVox.

For more:
- read this article
- read the release

13/01/2010 - 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.

12/01/2010 - 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.

11/01/2010 - Acrobits Brings Three New VoIP Competitors to the iPhone App Store

Acrobits releases three new 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.

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. As the VoIP market grows, Acrobits remains dedicated to providing both VoIP users and providers with the best Softphone on the iPhone.

06/01/2010 - ONSIP to Support HD Voice for All On Network And Conference Bridge Calls

Junction Networks announces that its OnSIP Virtual PBX is now the first Hosted PBX Service to support HD voice on all on-network and conference bridge calls. OnSIP is now wideband-ready, using SIP compliant, wideband telephony endpoints and conveys the kind of high-definition, in-the-room sound quality that is the next best thing to being there. Further, these high-quality HD calls will continue to be free on-net, thus further reducing telephone costs, while at the same time, offering additional features to users.

A growing number of companies have begun to offer high-definition endpoints -- IP phones and soft phones -- where the audio quality far exceeds that of traditional landline handsets. While all telephone audio quality has been measured against PSTN landlines until now, the fact is that the traditional PSTN conveys only about one fifth the range of frequencies the human ear can hear - a span of about 3500 Hz out of 20 kHz. With wideband-enhanced, high definition telephony, what was once hard to distinguish is now easy to hear.

Wideband or "high-definition" audio spans a range twice that of traditional PSTN voice, and has arrived with the proliferation of broadband VoIP. Further, the sampling rate of wideband audio calls is twice as high as traditional PSTN voice.

When it detects a connection between wideband extensions, OnSIP allows the two endpoints to negotiate a call using the standard wideband G.722 codec. Continuing the commitment to supporting any SIP compliant endpoints, OnSIP HD has been successfully tested with and between Polycom, Cisco, Linksys, Aastra and SNOM wideband phones, which customers can obtain and easily set up themselves as OnSIP extensions. Customers then enjoy in-the-room sound quality on extension-to-extension and on conference calls, as long as the person speaking is similarly equipped with a wideband endpoint.

04/01/2010 - IP Comm's Hot or Not in 2010

Yankee Group's Zeus Kerravala has a cool feature on what the hot trends will be in 2010... Cloud Computing possibilities, SIP Trunking, Consolidation all make the hot list. Article

15/12/2009 - Dialogic Becomes Digium Interoperability Partner

Dialogic has met the program requirements to become a Digium Interoperability Partner by completing the certification of the Dialogic 1000 Media Gateway Series and the Dialogic 2000 Media Gateway Series for use by the Asterisk community. Digium's Interoperability Partners have products that are complementary to and interface with the open source Asterisk telephony platform. These products interact with Asterisk through a SIP standards-based interface and are now certified by Digium for interoperability with Asterisk Business Edition.

Dialogic Media Gateways, including DMG1000 Gateways and DMG2000 Gateways, are widely used to provide PBX integration between applications deployed on SIP-based media servers and the installed base of TDM and hybrid IP-PBX systems. Open source software such as Asterisk has emerged as a viable SIP service creation platform used to create innovative communication applications that can be integrated with existing PBX infrastructures. In the absence of direct SIP to SIP integrations between an Asterisk-based solution and an existing PBX, Dialogic Media Gateways can provide the signaling and media translation necessary to make the solution work.

11/12/2009 - Requestec Provides Bell Mobility with 3G Mobile Video Calling App for Facebook

Requestec, an Adobe Flash-to-SIP telephony provider, announced their key involvement in the release of Bell Mobility’s, Bell Video Call application built on the Facebook platform.

The application allows Facebook users to visit the profile page of a Bell subscriber that has added the application and click on their Bell Video Call tab. From here, calls can be made from anywhere in the world to the Bell subscriber’s HSPA Video Calling handset; all at no cost to the caller.

The company claims it’s the first video calling application in North America that is fully integrated into Facebook.

09/12/2009 - Broadvox and Xorcom Certify Interoperability of Broadvox SIP Trunking with Xorcom IP PBX

Xorcom and Broadvox announces interoperability certification between the Xorcom series of Asterisk-based IP-PBX solutions and Broadvox SIP Trunking services. The bundled offering delivers significant cost savings over traditional telecommunication services for calling in the United States (excluding Alaska), Canada, Puerto Rico and the U.S. Virgin Islands while leveraging the flexibility and reliability of Xorcom's IP-PBX line.

Xorcom's all-in-one appliances allow seamless communication using both VoIP and traditional telephony protocols, such as FXS, FXO, BRI, T1/E1 PRI, T1 CAS and E1 R2. Members of the series differ in the number of supported users, starting from the basic XR1000, which is suitable for SOHO, up to the robust XR3000, which supports up to eight PRI connections along with hundreds of analog and IP extensions. The award-winning TwinStar feature enables dual-server redundancy for the complete PBX, including all telephony trunks and interfaces, as well as IP phones.

Broadvox SIP Trunking services enable efficient bandwidth usage for both voice and data applications plus an overall lower service cost for telephony, including unlimited local calling and reduced fees for long distance and international toll calls. Broadvox SIP Trunking services are delivered on a state-of-the-art IP communications network monitored 24x7 and supported by industry leading SIP engineers and customer service organizations.

Both Broadvox SIP Trunking and the Xorcom IP-PBX products are scalable and flexible, enabling customers to grow and enhance their system functionality as needed. The bundled solution is currently available both via the Broadvox and Xorcom reseller channels.

07/12/2009 - Acrobits Creates iPhone White Label Softphone Solutions for SIP VoIP Providers

Acrobits announces a new focus on providing SIP VoIP providers with their own custom-made iPhone Softphone. Acrobits can quickly provide a VoIP provider with their own custom iPhone application, allowing their existing customers to make calls via the iPhone. It also opens the door to new subscribers that already have an iPhone.

Acrobits has taken it to the next step. “By designing custom Softphone applications for them, we have found a way for VoIP providers to increase their market share in the lucrative market that the iPhone created,” says Acrobits. Peoplefone may be one of the first, but it is certain that many other VoIP providers will be interested in having their own custom designed application for the iPhone and Acrobits is ready and able to make them and their customers happy.

Though they have a renewed dedication to creating white label versions of their popular Softphone, don’t think that Acrobits will be neglecting their flagship product. Future versions of Acrobits Softphone will add support for multiple calls, call forwarding, voicemail, and many other features that VoIP users crave. Both Acrobits Softphone users and users of their white label clients will benefit from these added features.

04/12/2009 - Skype For SIP Now Available in Beta

Skype announced that it is opening up the Skype for SIP beta program. It allows businesses to utilize Skype for SIP with their existing SIP-based PBX or Unified Communications systems.

Skype for SIP beta enables businesses to receive and manage inbound calls from Skype users worldwide on SIP-enabled PBXs by either connecting the company Web site to the PBX via Skype click-to-call buttons or purchasing online Skype numbers.

03/12/2009 - Skype for SIP now open Beta

More SIP news this week--Skype for SIP, which until now has been in a closed Beta for only certain businesses since March, is now open to anyone who wants to try it out.

According to Skype Journal, anyone with a business control panel, a corporate Skype name subject to the business terms of service, and a business SkypeIn phone number can now use Skype for SIP. This new 'Skype Trunking' connects a company's phone switch to Skype with calls coming in being handled by Skype-to-Skype and Skype-In and all outgoing calls being made through Skype-Out.

Skype for SIP requires a Skype number, it costs $6.95 per line and international calls to 36 countries is 2.1 cents a minute.

For more:
- read this blog post

02/12/2009 - XO Unveils New Enterprise SIP

BroadSoft and XO Communications announced a new enterprise-class, multi-site SIP trunking solution offered by XO.

XO Enterprise SIP is an enhanced SIP trunking service offering for large enterprises, that enables them to transform their distributed PBX/PSTN interconnection to a “more centralized and cost-effective” VoIP solution.

16/11/2009 - Zultys Empowers Voice Solutions Through Internet Telephony Service Providers

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!

09/11/2009 - Doddle Makes VoIP Easy as Pulling up a Website

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.

04/11/2009 - InCharge Systems Announces Reference System for SIP Security Interoperability Testing

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.

02/11/2009 - SIP to save Skype?

GigaOM has a long-form report on how SIP might go far to replace the underlying software at issue between Skype and its founders. Report

29/10/2009 - Report: SIP Trunking catching on

A new report, SIP Trunking Deployment Strategies: North American Enterprise Survey, by Infonetics shows that purchase decision-makers at medium and large companies are either already spending money on SIP Trunking or they are planning to head in that direction. Many companies have deployed VoIP within their organizations, but they are still using legacy TDM to connect to the PSTN. The report finds that as technology upgrades start up again, SIP trunking will come to replace the legacy TDM technology. Due to the economy many tech upgrades are on hold for the time being.

Polling these medium and large companies, the survey asked about their use of PBX manufacturers, trunking services, providers, and expenditures. They found that 39 percent of respondents have already deployed SIP trunking and the majority are deploying it across their companies--not just in small trials. By 2010 it will be the second most commonly deployed trunking type.

The typical respondent to the survey spends between $100 thousand and $500 thousand per year on trunking services.

For more:
- read the release

28/10/2009 - Sipera SLiC Makes Smartphone VoIP and UC Secure and “Business Ready”

After demonstrating how easy it was to eavesdrop and record VoIP calls made over an unsecured WiFi network on the iPhone using open source software called UCSniff, Sipera Systems, which offers real-time Unified Communications (UC) security, released the Sipera Secure Live Communications (SLiC) mobility solution.

The company claims Sipera SLiC is the industry’s first security solution enabling enterprises to “tame” the smartphone, permitting employees to use VoIP, UC, cloud telephony, and other low-cost and feature-rich communications applications on mobile devices with complete security and privacy.

27/10/2009 - Panasonic Announces Interoperability of Its SIP Cordless Phones with Broadsoft's Broadworks Platform

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

23/10/2009 - Acme Packet’s SBC's Selected to Ease Interoperability with IP PBXes for Skype for SIP Beta

Skype announced the selection of Net-Net OS-Enterprise session border controllers from Acme Packet for its Skype for SIP beta offering. Acme Packet is the leader in session border contro solutions and the sole SBC partner for the Skype for SIP beta program.

Skype’s deployment of Acme Packet’s SBC simplifies the interoperability and feature compatibility of the Skype for SIP beta offering with enterprise IP-PBX equipment and next-generation unified communications platforms which utilize the SIP standard. As a result, Skype for SIP will allow those enterprises with an on-premise IP PBX to take advantage of an innovative IP-enabled communications tool and to benefit from end-to-end interoperability. The implementation of an SBC by Skype as part of the Skype for SIP beta program also enables the delivery of high-quality, real-time interactive communications, while minimizing the exposure to risks for those companies who sign up for the trial.

Skype for SIP will allow many companies to reduce their costs by making outbound calls to landlines and mobiles worldwide at low Skype rates from devices connected to their existing SIP-enabled PBX systems. It will also allow organizations to receive inbound voice calls to their PBX from the more than 400 million registered Skype users around the world via a global click-to-call button on their Web site. In addition, if they buy and associate local online numbers with their PBX, they can receive inbound calls to the PBX from landline and mobile phones via Skype.

22/10/2009 - VoicePulse Expands Executive Management Team with Christopher Y. Silk

VoicePulse announced, today, that it has hired Christopher Y. Silk as its new Chief Executive Officer. Silk will report to the company's Board of Directors. Founder and former President/CEO, Ravi Sakaria, will serve as Chairman of the Board.

This transition allows Sakaria to focus on the company's overall direction and strategy. Sakaria sees Chris as the right person to continue and grow VoicePulse's success as an industry leader in quality VoIP products and unmatched customer service.

"I am confident that, by hiring Chris, I have strengthened VoicePulse's ability to deliver innovative services based on VoIP technology and capitalize on the company's core competencies,” Sakaria explains. “Chris' combination of management, financial and sales skills are a clear match for VoicePulse and I'm extremely excited about the company's future under his capable leadership."

Mr. Silk was brought on to accelerate the growth of the company and to build upon its core competencies. He brings with him a wealth of experience in operations management, sales strategy and business development.

“I am very excited about the opportunity to join VoicePulse in this leadership role,” says Silk. “This is a company I have known well for the last 5 years and I believe we are ideally positioned for continued success. Ravi, Ketan and the rest of the VoicePulse team have built an incredible foundation which will support a world class telecommunications company.” Silk adds, “It is our goal to take this foundation and move aggressively forward as an industry leader in VoIP technology.”

Silk's career has spanned over 15 years in the telecommunications industry, including leadership roles with Verizon, UUNET and SBC. He was also the CEO of a private cable company where he worked on several acquisitions as the "Mom & Pop" Private Cable industry evolved into consolidation. Silk's experience and expertise will be the driving force behind VoicePulse's continued success.

Source: PR.com

05/10/2009 - Sprint SIP Trunking Goes "General Availability" for Customers

Sprint now is reportedly selling SIP trunking service on a "general availability" basis, allowing business users to buy a single IP connection for voice, data and video communications while reducing local, long-distance and calling feature expenses.

The company was one of the first U.S.-based providers of SIP trunking services qualified for use with Office Communications Server 2007 R2, and has been selling the service since early 2009 on a limited basis.

SIP trunking eliminates the use of primary rate interface (T1) local trunks and allows businesses to share capacity over one IP connection for multiple locations and applications.

Sprint SIP trunking is available through the company's newly formed Business Markets Group. Composed of more than 4,000 sales, support, marketing and operations personnel, BMG is solely dedicated to enterprise, general business and public sector customers.

Click Here for the Complete Article

05/10/2009 - Vonage Goes Mobile: Wi-Fi and Cellular Networks Low Rates Calls Available

Vonage has launched Vonage Mobile, its first mobile calling application for smartphones. This free downloadable application provides seamless, low-cost international calling while on Wi-Fi or cellular networks.

It’s available for download on the iPhone, BlackBerry and iPod touch.

The app works with the existing mobile plans, what lets you keep your number, mobile device, existing contacts and mobile service provider.

05/10/2009 - Sprint's SIP trunking now generally available to OCS clients

Sprint announced in a press release the general availability of its integrated Global MPLS network and SIP trunking for customers using Microsoft Office Communications Servers 2007 R2. As a cost saving measure, SIP trunking allows companies to use a single IP connection for the convergence of voice, data and video communications. With the general availability release of SIP trunking, Sprint has positioned itself as a one-stop-shop vendor of unified communications for Office Communications Servers customers, according to the company.

Sprint was one of the first U.S. companies to offer SIP trunking to companies using Microsoft Office Communications Servers 2007 R2, beginning a limited offering in early 2009. In February, Sprint combined its Global MPLS network with Office Communications Servers SIP trunking to facilitate companies shifting to Unified Communications. "The combination of Office Communications Server 2007 Release 2 and Sprint SIP Trunking provides a powerful new way for people to collaborate and offers customers a rich and integrated communications experience," said Eric Swift, senior director of the Microsoft Unified Communications Group at Microsoft Corp, back in February.

For more:
- read the release

Related articles
Analyst: SIP trunking could drive SBC market
Sprint announces new unified communications offerings
Sprint enhances VoIP services for cable companies

24/09/2009 - Via Browser, Junction Networks' 'my.OnSIP' Adds Presence, Availability, IM Missing Pieces to Hosted PBX Service

Junction Networks is broadening the range of its popular OnSIP hosted PBX service beyond voice, adding the unified communications features of instant messaging, presence, and phone status. All four components are presented to OnSIP end users through my.OnSIP, an IM-style interface available from a browser.

Through one window, my.OnSIP shows all users which contacts on their hosted PBX are "present," which are on the phone, and which are free to call with a click on their names. It lets users send calls only to those who can answer, avoiding voice mail and phone tag. And unlike consumer IM services that are often banned from the workplace as note-passing distractions, my.OnSIP's chat is limited to those on the hosted system, even though those coworkers and their extensions may be physically located at multiple sites on different continents or at temporary locations.

Capex and Opex that SMBs Love to Avoid

"In adding these unified communications features to OnSIP, and by making them Web-accessible, we're closing the last feature gaps between hosted and on-premise phone systems," said Junction Networks President Robert Wolpov. "True, some on-premise PBX vendors offer chat, presence, and maybe even phone status - but they often charge considerably extra for these non-voice media, and they often require proprietary phone sets. Most importantly, as customer-premise equipment, they always require the capital and operating expenditures that small-to-medium sized businesses love to avoid."

The hosted alternative to an installed PBX enables coworkers in one or many sites to share the same company phone number, greetings, auto attendant, extension dialing and transfer, hunt groups, voice mail system, and advanced features enjoyed by workers whose extensions are wired to the traditional PBX system in the traditional company phone closet. The difference is that all this phone switch functionality is outsourced to a hosting provider in a remote center, leaving the business with nothing to buy but the phone sets themselves.

Different phones, a uniform control display

"While some hosted providers - ourselves included -- have long supplied Web tools for administrators, few if any have extended Web access down to the level of end-user phone controls," Wolpov said. "In the process, we let our customers' employees all share the same 'deluxe' phone toolset, with the same display and the same clickable ways of making, taking, and transferring calls - even if they all have different SIP phones with different buttons."

My.OnSIP controls any SIP-compliant phone or softphone and runs on the browser of any desktop or laptop running Windows, Macintosh, or Linux. Available to OnSIP users now, it can be accessed for free by logging in at my.onsip.com.

The UC tool and interface add no cost to the OnSIP hosted PBX service, which starts at $39.95 per month for a full suite of PBX features and an unlimited number of users and extensions. Unlike typical hosted services that require long-term contracts and add $30 to $50 per seat per month to a base PBX charge, OnSIP requires no contract and only adds charges per usage. "Usage" includes off-network voice minutes (on-network calls are free); additional voice mail boxes; or advanced applications, such as next-available-agent-style call distribution.

Junction Networks also is making my.OnSIP's application programming interface available to developers, who may hook the interface's calling, instant messaging, presence and availability information to any application whose users need real-time communication.

23/09/2009 - Skype for SIP Now Interoperable with Cisco Unified Communications 500 Series

Skype announces the beta version of Skype for SIP has been certified as interoperable with the Cisco Unified Communications 500 Series for Small Business. This will enable small and medium-sized businesses who manage their networking and communications needs with this affordable UC solution to communicate more efficiently by directing their outbound calls to mobiles and landlines via Skype, while also allowing them to receive inbound calls from Skype users.

Interoperability with Skype for SIP means that small businesses can take advantage of the cost savings provided by Skype’s low-cost global calling rates when their employees call landlines and mobiles around the world. A company can also receive inbound voice calls from any of the more than 480 million registered Skype users around the world via a global click-to-call button on its Web site. These Skype calls are received in the Cisco Unified Communications 500 Series solution and can be handled or directed in the same way as any other inbound caller. In addition, if a company buys and associates online Skype numbers with their Cisco Unified Communications 500 Series solution, it can then receive inbound calls via Skype from business contacts and customers calling from landline and mobile phones.

The Cisco Unified Communications 500 Series platform is part of Cisco’s Smart Business Communications System which continues to expand having just added a new set of IP phones with high definition audio, a unified threat management device as well as support for third party application integration, including products from healthcare, automotive and insurance industries.

Certification testing of Skype for SIP with the Cisco Unified Communications 500 Series for Small Business was conducted by tekVizion Labs, an independent test facility in Richardson, Texas, which specializes in IP communications interoperability testing.

Cisco VARs will need to register for the Skype Service Partner Program and pass an online certification exam to qualify to configure the Cisco solution to support Skype for SIP, as well as to support those business customers who may already be using the Cisco Unified Communications 500 Series for Small Business and want to integrate Skype for SIP into their present communications solution.

26/08/2009 - JAJAH Brings SIP Trunking Services to the Enterprise

JAJAH, the IP communications company, is working with Microsoft to provide SIP Trunking services to Microsoft enterprise customers globally. According to the firm this will allow companies to make high quality voice calls over JAJAH's IP Platform in the cloud, without requiring an infrastructure upgrade.

14/08/2009 - InterAct First to Validate Next Generation 9-1-1 Architecture

InterAct, a provider of software for enterprises and government agencies, announced the successful integration with proposed Next Generation 9-1-1 architecture. The company is the only provider to completely process end-to-end NG9-1-1 calls from the caller to the Computer-Aided Dispatch (CAD) and Geographic Information mapping systems (GIS) using nothing but IP connections.

12/08/2009 - Junction Networks Decides to Test and Review SIP Phones

Small and medium-sized businesses often turn to hosted IP PBX services for rich features, multi-site convenience, zero maintenance and lower phone bills. Now they’ll have help picking the right phones. Junction Networks, makers of Hosted VoIP service OnSIP, has begun a review site where their engineering team will regularly post evaluations and the results of their independent, vigorous tests on SIP phones, soft phones and other user agents.

“Once customers plug in and register their phones as extensions to a SIP PBX, whether hosted or on-premise, they should have no further worries about that phone’s capabilities,” says Robert Wolpov, Junction Networks president. “We put each phone through a multi-step interoperability test of 32 basic functions, as outlined in the SIP specification: ring, go on hold, transfer, and so on. If a phone fails any one of them, we’ll let you know.” They also judge models by subjective criteria such as voice quality and ease and comfort of use.

Wolpov points out that Junction Networks, unlike most other hosted IP PBX companies, does not resell any particular vendor’s phone, allowing it to make unbiased judgments (and allowing OnSIP customers to use any SIP-compliant device they may already own). “At the same time,” he notes, “our experience with a wide range of customers allows us to fit a review to the user scenario. We’ll advise you, for example, that high-definition audio quality is a worthwhile splurge, but if you’re choosing a conference room phone that won’t be used much, it’s ok to save money with traditional audio quality.”

The site is kicking off with Linksys SPA942, Polycom 331, and Snom 320 VoIP phone reviews. Junction Networks intends to review more of the 20 phones whose configuration details are already listed on the OnSIP Knowledgebase, as well as new models as they’re released. They’ll use Twitter (www.twitter.com/onsip) to tell followers when new results are posted. They also hope to receive the same requests for evaluation that vendors commonly send the testing labs of trade and consumer media. Readers are welcome to talk back with their own comments on reviews and phones.

07/08/2009 - Paradial to Deliver Firewall NAT Traversal Solution to Major Asian Telecom Operator

Paradial, an IP-communications software developer, has signed an agreement with a major Asian telecom operator, a comprehensive provider of communications services in the region.
The licensing agreement covers Paradial's RealTunnel standards-based firewall and NAT traversal product, which includes STUN, TURN and ICE support.

22/07/2009 - Zultys & Broadvox Make SIP the Right Choice for Connectivity

Zultys and Broadvox announce interoperability certification between the Zultys MX family of IP PBX communication systems and Broadvox GO! SIP Trunking to provide feature-rich Unified Communications services to small and medium businesses and enterprises, while saving them money and providing high-quality, reliable voice services.

Broadvox provides SIP trunking services to Small and Medium Business and Enterprises as well as many Carriers. As well as the "SMBs SIP solution" with their Broadvox GO! SIP Trunking offering, the pre-tested interoperability of Zultys' PBX products and Broadvox VoIP network enables businesses to deploy customized Unified Communications solutions and achieve the full benefits and cost-savings of business VoIP. Broadvox's nationwide network coverage delivers multi-site customers with the benefit of a single coast-to-coast trusted SIP provider.

Zultys' eight years of experience producing pure SIP IP-PBX products, combined with Broadvox's dedication and commitment to the IP voice network benefits customers by reducing costs while increasing productivity.

The Zultys family of advanced SIP Open Standards-based IP-PBX products offer enterprises and small and medium businesses a feature-rich energy-efficient server that does more in one box than any other IP PBX on the market. The MX250 IP PBX and MX30 IP PBX servers function as full PSTN/IP gateways and are loaded with business-enhancing features right out of the box, such as Soft Phone, Find-me/Follow-me, Presence, Secure Chat, Teleworker support, SIP open standard desktop and cordless phones, and much, much more.

19/06/2009 - Media5 SIP Softphone App Turns iPhone into IP-PBX Extension

Media5 has released a SIP client application that allows the Apple iPhone and iPod Touch to be used as a IP-PBX extension. The company says the full-featured softphone enables the Apple devices to be used to access the same phone services and features as if they were in the office.

That includes remote workers being able to contact other offices or employees.

Pascal Doré, Media5's mobility product line manager, said the new release of the Media5-Fone extends its mobile portfolio to iPhone users on the go. "It offers them the key features needed to integrate an easy-to-use SIP IP-PBX extension within the iPhone," she said.

Doré said in addition to the Lite version, Media5's engineers are working to bring the next fully featured Enterprise version of the Media5-Fone. She said that will embed strong Voice security encryption among the key features.

VoIP service providers who offer calling plan can also benefit from the same SIP connectivity extension for their customers who own an iPhone.

Enterprise users can also leverage the cost-saving benefits of VoIP by enabling their users with high quality phone calls wherever there is a broadband connection.

Media5-Fone is now available in the Apple App Store.

Other features of the Media5-Fone include:

Voice Mail Integration
Loudspeaker
VoIP over Wi-Fi
Native Contacts List
Hold
Easy Configuration
Call History
Mute

Source: VoIP Biz

17/06/2009 - Media5 SIP Softphone App Turns iPhone into IP-PBX Extension

Media5 has released a SIP client application that allows the Apple iPhone and iPod Touch to be used as a IP-PBX extension.
The company says the full-featured softphone enables the Apple devices to be used to access the same phone services and features as if they were in the office.

03/06/2009 - SIP Security Book Gives a Detailed Overview of SIP Security Issues

Research and Markets has announced the addition of John Wiley and Sons Ltd's new report "SIP Security" to their offering.

This book gives a detailed overview of SIP specific security issues and how to solve them. While the standards and products for VoIP and SIP services have reached market maturity, security and regulatory aspects of such services are still being discussed. SIP itself specifies only a basic set of security mechanisms that cover a subset of possible security issues. In this book, the authors survey important aspects of securing SIP-based services. This encompasses a description of the problems themselves and the standards-based solutions for such problems. Where a standards-based solution has not been defined, the alternatives are discussed and the benefits and constraints of the different solutions are highlighted.

Key Features:

Will help the readers to understand the actual problems of using and developing VoIP services, and to distinguish between real problems and the general hype of VoIP security
Discusses key aspects of SIP security including authentication, integrity, confidentiality, non-repudiation and signalling
Assesses the real security issues facing users of SIP, and details the latest theoretical and practical solutions to SIP Security issues
Covers secure SIP access, inter-provider secure communication, media security, security of the IMS infrastructures as well as VoIP services

Vulnerabilities and countermeasures against Denial-of-Service attacks and VoIP spam this book will be of interest to IT staff involved in deploying and developing VoIP, service users of SIP, network engineers, designers and managers. Advanced undergraduate and graduate students studying data/voice/multimedia communications as well as researchers in academia and industry will also find this book valuable.

26/05/2009 - Ingate and Dialogic Announce Secure SIP Trunking for Legacy PBX

Ingate Systems and Dialogic Corporation have announced a partnership that will allow enterprises using legacy PBX and Contact Center systems to adopt SIP trunks as a replacement for traditional PSTN voice services.
The companies said they have completed the necessary testing to validate the Dialogic 2000 Media Gateway Series (DMG2000) as interoperable with Ingate SIParator and Ingate Firewall products.

13/05/2009 - Acme Packet and BroadSoft Enhancement Their Joint SIP Trunking Solution

Acme Packet and BroadSoft announce new enhancements to their joint SIP trunking solution that enables service providers to ensure business continuity for large enterprise customers. The solution, which integrates BroadSoft's BroadWorks VoIP application platform with Acme Packet?s Net-Net session border controller, delivers advanced IP-based communications services that were previously not available with premise-based PBX solutions. These services include Microsoft?s Hosted Messaging and Collaboration version 4.5, video-enabled SIP trunking and fixed-mobile convergence.

IP-based Services Increase Collaboration and Productivity for PBX/IP-PBX Customers

While cost savings continue to be a major driver, the demand for unified communicationsand other value-added services is perpetuating the rise in SIP trunking. The BroadSoft/Acme Packet solution enables service providers to increase average revenue per user and reduce churn by bundling applications with their connectivity offers. The solution allows an enterprise with an on-premise PBX or IP PBX to take advantage of innovative IP-enabled communications tools from the service provider ?cloud? including:

Microsoft?s Hosted Messaging and Collaboration version 4.5 ? integrates premise-based PBX phones with IT tools such as e-mail, presence and instant messaging;
Video-enabled SIP Trunking ? delivers ?personal telepresence? to PBX customers by adding video stations to selected employees and meeting rooms; and
FMC ? extends PBX features to mobile devices independent of the network, using BroadSoft?s award-winning BroadWorks Anywhere functionality.

New Enhancements Increase Security and Resiliency of SIP Trunking Connection

New features and functionality in BroadSoft?s latest release of BroadWorks, 14.sp9, further strengthen the BroadSoft/Acme Packet SIP trunking solution, enabling service providers to meet the stringent business continuity requirements of global enterprises with large IP PBX deployments. New trunking features support:

Fully Redundant IP Networks ? eliminates any single point of failure for an enterprise;
Multiple Trunk Groups per IP PBX ? enables an enterprise to apply sophisticated routing policies for delivery of calls across the trunk groups; and
Dynamic Multi-site Enterprise Support ? allows an enterprise to purchase a fixed amount of call capacity and apply that across any number of locations. This is particularly useful for multi-site call center deployments where capacity moves between sites based on the time of day.

In this solution with BroadSoft, Acme Packet?s Net-Net family of session border controllers enhances both the service providers? and enterprises? SIP trunking functionality. For service providers, it provides controls for session admission and overload controls for the IP network transport to assure SLAs, maximize revenues and minimize costs. For enterprises, Acme Packet Net-Net SBCs provide similar control functions relative to its IP PBX, Unified Communications platform and network.

End-to-End Interoperability Testing Improves SIP Trunking Go-to-Market

SIP trunking presents a new set of go-to-market challenges relating to IP PBX interoperability for service providers. IP PBX vendors have different degrees of maturity in their SIP trunking implementations and may not be SIPconnect compliant. Unlike competitive offerings, Acme Packet and BroadSoft have addressed these challenges by ensuring end-to-end interoperability testing with over 40 different IP PBX vendors and variants, including Avaya, Cisco, Siemens and Microsoft?s Office Communications Server 2007 R2. In addition, the BroadSoft/Acme Packet solution has attained SIPconnect compliance and supports current 3GPP specification for deployment of SIP trunking in IP Multimedia Subsystem networks.

07/05/2009 - Cost Savings Drive SMBs To IP Telephony

Small to medium-sized businesses primarily shift to VoIP services because of the cost savings they offer.
That's the conclusion of a new report from Infonetics Research, which also points to powerful features as a secondary motive for SMBs to switch to IP telephony.

29/04/2009 - XConnect Appoints IP Expert Shockey To Board

Richard Shockey has joined the advisory board of XConnect, the VoIP and Next Generation Network (NGN) interconnection service provider.
A pioneer in ENUM (Electronic NUMbering) and expert in VoIP, Shockey is a founder and has been co-chair since 2002 of the IETF (Internet Engineering Task Force) ENUM Working Group.

04/04/2009 - Speakeasy Certifies Digium's Asterisk PBX for SIP Trunking

Speakeasy, a Best Buy company, has added Digium, the Asterisk company, to its growing portfolio of partners certified interoperable with Speakeasy’s expanded SIP Trunking integrated voice and data services.

Speakeasy further expands its SIP Trunking integration of voice and data services to reach an even larger SMB market with the certification of Digium’s Switchvox SMB and Switchvox SOHO IP PBXs and AsteriskNOW open-source telephony platform.

“We are excited to certify Digium for our expanded SIP trunking services,” said Bruce Chatterley, Speakeasy president and CEO. “By certifying Digium’s Switchvox and AsteriskNOW offerings, we are working together to provide solutions for small business customers to upgrade their telecom infrastructures regardless of their legacy voice and data systems.”

Digium is one of the first IP PBX manufacturers to be certified with Speakeasy. Speakeasy can deliver its SIP trunking service directly to Digium’s Asterisk telephony hardware products.

Source: Phone+ Mag

30/03/2009 - Avaya Announces SIP Architecture That Connects Users, Applications and Systems

Avaya today announced the launch of a new SIP-based architecture that integrates communications across multi-vendor, multi-location and multi-modal businesses.
Called Aura, the company said it is centered on the new open standards Aura Session Manager, which centralizes communications control and application integration.

30/03/2009 - snom Introduces New VoIP Conference Speaker Phone

snom introduces the snom MeetingPoint a SIP-based conference phone for the North American enterprise and small and medium-sized business markets at VoiceCon Orlando 2009. Designed for both medium to large size meeting rooms, the snom MeetingPoint features Konftel?s OmniSound 2.0 audio conferencing technology with advanced noise suppression and high fidelity wideband audio and offers a host of powerful conferencing capabilities. Featuring snom?s fourth generation SIP technology, the snom MeetingPoint provides enterprise and SMBs businesses with the same advanced SIP calling features and broad compatibility with standards-based IP PBX, hosted VoIP and unified communications solutions as snom?s 820 and 3XX series desktop VoIP phones.

The snom MeetingPoint is generally available today in the U.S. and Mexico through snom?s network of North American distributors and resellers and has an MSRP of US $899.

25/03/2009 - Gizmo5 CEO Challenges Skype For SIP

The CEO of Gizmo5 Michael Robertson has responded to last week's announcement of Skype for SIP by posting a comparison (see below) of the new service and his own company's OpenSky.
While welcoming Skype's initiative, he described it as a "vaporware announcement" with "murky pricing details".

23/03/2009 - eBay Bets on Skype's Entry Into SIP-based PBX To Boost Revenue

Skype has launched Skype for SIP, a beta program that allows companies to make domestic and international VoIP calls from an office PBX rather than PC.
The move comes the week after eBay announced that it expects Skype to more than double its revenue to over USD $1 billion by 2011 - with hopes high that the new business service will be a compelling proposition.

23/03/2009 - Beta Version of Skype Comes to SIP-based PBX Systems

Skype announces the beta version of Skype For SIP for Business users. SIP, short for Session Initiation Protocol, is an open standard and the leading voice over Internet protocol used in businesses telephony networks at millions of locations globally. According to IDC, 438,000 IP PBXes were shipped worldwide in 2008.

Skype For SIP allows SIP PBX owners to benefit from Skype?s low cost calls to fixed phones and mobiles around the world, and to receive calls from Skype users directly into their PBX system.

Businesses can now be reached by the community of over 405 million Skype registered users through click-to-call from their business Web sites. The calls will be received through their existing office system at no cost to the customer. At the same time, businesses can benefit from Skype?s low-cost global calling rates when placing calls to landlines and mobiles worldwide from devices connected to their PBX systems. In addition, they can choose to purchase online Skype numbers available in over 20 countries to receive calls from business contacts and customers who are using traditional fixed lines or mobile phones.

Key Features

The beta version of Skype For SIP will enable business users to:

Receive and manage inbound calls from Skype users worldwide on SIP-enabled PBX systems; connecting the company Web site to the PBX system via click-to-call
Place calls with Skype to landlines and mobile phones worldwide from any connected SIP-enabled PBX; reducing costs with Skype?s low-cost global rates
Purchase Skype?s online numbers, to receive calls to the corporate PBX from landlines or mobile phones
Manage Skype calls using their existing hardware and system applications such as call routing, conferencing, phone menus and voicemail; no additional downloads or training are required

How to participate

The Skype For SIP beta program for business users opens today. SIP users, phone system administrators, developers and service partners are invited to apply at www.skypeforsip.com. Applicants will need to be businesses, have an installed SIP based IP-PBX system, as well as a level of technical competency to configure their own SIP-enabled PBX. The initial beta is available to a limited number of participants.

During the beta period all calls will be charged at standard Skype rates. Further pricing details will be announced when the product is fully launched later this year.

19/03/2009 - Nimbuzz Bridges iPhone 3G VoIP Gap

Nimbuzz has today released what it describes as the most comprehensive VoIP application for the iPhone after "quite a few" rejections from Apple.
Building on its iPhone app launched in November, Nimbuzz users can now make international calls to mobiles and landlines at domestic rates by dialing a local access number available in over 50 countries.

03/03/2009 - INTERVIEW: Carrie Hartford Fedders From IPsmarx Technology

IPsmarx was named as joint winner of the 2008 voip-biz.news Product of the Year Award last week for its SIP-based calling card platform.
Carrie Hartford Fedders, account manager with IPsmarx, spoke to voip-biz.news about the solution, which eliminates the need for a VoIP gateway and PSTN lines using DID (Direct Inward Dialing)
technology.

03/03/2009 - Speakeasy Launches Direct SIP for Integrated Voice

Speakeasy expanded its SIP IP voice and data integration capability as part of its comprehensive 'Integrated Voice' service. Speakeasy can now provide combined voice and data service over direct SIP trunks, in addition to its existing service over analog and digital lines. Speakeasy's Integrated Voice connects to a traditional key system or PBX, and now with direct SIP, to an IP PBX, making it easier than ever to take advantage of the cost savings and flexibility of Speakeasy Voice.

To support direct SIP, Speakeasy plans to certify eight of the major IP PBX hardware manufacturers. Speakeasy has certified the ADTRAN Netvanta 7100 and Fonality trixbox Community Edition (CE).

With Speakeasy's Integrated Voice, available bandwidth is automatically allocated to data transmission, such as e-mail and general Web use, when calls are not in use. This optimizes the customer's network utilization. In addition, by using an optional VoIP codex, G.729, customers can free up even more data bandwidth by further compressing voice traffic.

02/03/2009 - AireSpring Recognized as Leading SIP Trunking Provider

AireSpring was awarded the coveted Members' Choice Award as the top "SIP Trunking" VOIP provider from the Telecom Association, a national professional membership organization of over 3,200 Telecom industry professionals. In addition, AireSpring was awarded 2nd place in the CLEC/LEC and Reseller categories and finished in the top 10 for Carrier, Internet/Data, and Multi-Location provider. The total of 6 major awards reinforces the effort that AireSpring has put into the creation of its Local, Long Distance, and Data products as well as the revolutionary next generation IP network which supports AireSpring's SIP products.

"Winning TA's annual Members' Choice award is a significant tribute to each winner's channel partner and customer service Programs," stated TA Founder Dan Baldwin. "This is the fourth consecutive year for our annual Members Choice award and we set a new record in ballots cast over the past several months."

"AireSpring is thrilled to be recognized by the TA for our achievements in IP communications and as a Carrier and Reseller of innovative, aggressively priced, voice and data products," stated Daniel Lonstein AireSpring COO. "IP Communication is the direction that the entire industry is moving; it is our privilege and honor to be chosen by fellow telecom professionals as the best SIP and VOIP provider in the industry. We continue to be inspired by the accolades we receive for our product line and look forward to releasing even more cutting-edge products in the coming year."

AireSpring's Voice, Data, and Integrated products are continually recognized by customers and agents as robust, flexible, and affordable. Over the past several years, AireSpring has been awarded Top Reseller, Top Channel Program, Top SIP Trunking Provider, and Product of the Year by various magazines and organizations. AireSpring currently offers lowest cost High Speed Internet, Voice, and SIP Trunking services as well as innovative hybrid products which deliver many of the advantages of SIP to customers with legacy TDM equipment. AireSpring continues to innovate and expand the reach and features of products offered through its groundbreaking enhanced IP network.

27/02/2009 - MyGlobalTalk and IPsmarx's SIP-based Calling Card Platform Share voip-biz.news Product of the Year Award

Two innovative products dominated voting to share the honours in voip-biz.news' Product of the Year 2008 competition.
With 33 per cent of the nominations, MyGlobalTalk's VoIP calling solution earned praise for its sound quality and call rates, as well as features such as no contract being required, no connection fees and no minimums.

20/02/2009 - Swiss GSM Carrier in&phone Buys Blueslice's SDM Platform

Blueslice Networks has sold a SIP-enabled ngHLR, HSS and AAA, bundled into one fully integrated solution, to Unify Mobile.
The SDM platform is to be used by in&phone, one of its mobile operations in Switzerland.
Montreal-based Blueslice's CSP 3000 includes the ngHLR and its Advanced Low Cost Roaming solutions - giving in&phone the ability to offer subscribers new roaming features.

06/02/2009 - SIP Print Enters UK With FSA-Compliant VoIP Call Recording Solutions

SIP Print has announced the availability of its voice recording appliances for the UK financial services market.
The move marks the preliminary entry into the UK market for SIP Print.

16/01/2009 - XO Communications Names Wagner As New Head of Business Services

Daniel Wagner has been appointed head of XO Communications' Business Services unit.
The appointment, which is effective immediately, will see Wagner focus on accelerating the division's profitability and revenue growth.

19/12/2008 - Gizmo5 Introduces Browser-Based VoIP Application

Gizmo5 has launched a web-based VoIP app that allows users to call 800 numbers and SIP addresses for free.
GizmoCall is Flash-based, so it only requires a browser to use the service rather than having to download a software client.
Users go to the Web site, sign up for a username and password, and start making calls.

18/12/2008 - OnePhone VoIP Client Coming To Blackberry

Devoteam is to release a Blackberry version of its VoIP client OnePhone that runs on mobile platforms enabling voice calls over an IP network.
It is expected to be available for the RIM handset in the first quarter of 2009.
OnePhone is a SIP-based, dual mode GSM-WiFi solution that is able to interwork with public and private WiFi hot spots, and with mobile networks.

04/12/2008 - Ingate Partners with Codima for VoIP Installations

Ingate has entered a partnership agreement with Codima. Ingate and Codima will create a vertical offering to provide SIP-based VoIP pre-assessment and post-deployment tools from Codima Toolbox combined with Ingate Firewall and Ingate SIParator products, to enable successful VoIP installations.

The market for Unified Communications is expanding rapidly and the partnership leverages the need for robust SIP-based VoIP solutions. Organizations today are looking for new opportunities to communicate such as Instant Messaging, presence, conferencing capabilities, the ability to share applications and more. At the same time, they are relying on their business-critical communications systems to perform 24/7. Opening up new ways to communicate cost-efficiently, SIP-based VoIP installations benefit from the innovative technologies developed by Codima and Ingate.

The partnership enables Ingate to ensure VoIP readiness to its customers by having an accurate overview of the network and an in-depth network analysis. The Codima flagship product autoMonitor identifies devices and software installed on a network and draws maps of the topology directly in Microsoft Office Visio in addition to monitoring the network finding the weakest links. With this crucial information available, an Ingate installation offers a new cost-efficient way to ensure that a SIP-based unified communications system is fully functional from the start, avoiding project delays and customer dissatisfaction.

The beginning-to-end solution for managing VoIP networks, Codima Toolbox adds value to the offering. For example, autoVoIP? Traffic Simulator tests if an existing network can carry the new converged technology and the monitoring and troubleshooting tools ensure call quality in daily operation.

In a SIP trunking scenario, the Ingate Firewall and SIParator assure smooth communication between the service provider and the enterprise IP-PBX by mutually adapting the used SIP flavors (SIP normalization for interoperability). Ingate products utilize a frequently updated Startup Tool with pre-configurations for approved SIP trunking service providers on one side, and for all leading IP-PBXs on the other. The installation of fully secure SIP trunks is thus reduced to a plug-and-play procedure.

The Ingate Firewall or SIParator, installed as the demarcation point of the enterprise toward the Internet, which often is the bandwidth bottleneck, are at an ideal location to provide priority of voice over data. This is performed by the advanced Ingate Quality of Service function, effective both for in- and outbound traffic.

25/11/2008 - snom Teams with Sangoma to Accelerate Delivery of Open Source VoIP

www.snom.com has integrated of its SIP-based phone portfolio with Sangoma Technologies. After rigorous testing and evaluation, snom's 3xx series VoIP phones and snom m3 wireless IP DECT phone received full certification by Sangoma's lab as interoperable and deployable with its AFT series of cards: A101, A102, A104 and A108 PCI and PCI Express cards.

With the technology partnership, enterprises with legacy PBX platforms can realize the benefits of IP telephony by utilizing Sangoma's open source VoIP components with snom handsets.

snom's 3xx series (snom 300, 320, 360 and 370) are the industry's premier, business-class, open, standards-based SIP VoIP phones. They feature a global executive design and styling, with a large, high-resolution display screen, programmable function keys, and advanced business calling features.

The snom m3 IP DECT is the company's first cordless offering and provides an optimal VoIP communications system for the home office, SMB or enterprise. The new mobile VoIP phone features an elegant design and advanced mobility without compromising audio quality. Since the company's inception in 1996, snom has been a leading proponent of open standards and is interoperable with the broadest array of IP telephony platforms.

snom recently announced the debut of the snom 820, a powerful new business VoIP phone for the North American enterprise and small and medium-sized business markets. The new snom 820 SIP-based business phone sets a new standard for design innovation and business-class performance blending a sleek, cutting-edge look with a highly intuitive user interface and a rich set of business communications features.

snom phones offer the most comprehensive VoIP security including support for TLS and SRTP protocols and VPN capabilities. The SIP telephones can support several audio devices simultaneously, such as the handset & headset, hands-free and power over Ethernet.

21/11/2008 - Security tool for VoIP solutions released

A new tool which allows enterprises to assess if their VoIP solutions are vulnerable to targeted eavesdropping has been released.
UCSniff, from Sipera Systems' VIPER Lab, is a free application which allows network managers find out how easy it is to imitate an enterprise VoIP phone, download a directory and then listen in on confidential calls.

18/11/2008 - Acme Packet Certified in GSMA PathFinder Partner Programme

Acme Packet obtains product certification for the GSM Association?s PathFinder Industry Partner Programme. PathFinder is a GSMA-managed service operated by NeuStar that enhances IPX (IP eXchange) services with destination discovery capabilities for voice and other sessions. The IPX is a managed interconnect service defined by the GSMA, offering secure, high-quality transit services to both mobile and fixed network operators for SIP and other IP-based services.

The PathFinder Industry Partner Programme is designed to foster interoperability and support relationships with companies offering products and services that are complementary to the PathFinder service. Acme Packet?s Net-Net Session Director and Session Router configurations have proven interoperability with the ENUM-based number and route resolution service managed by NeuStar. This service demonstrates Acme Packet?s Open Session Routing architecture for delivering trusted, first-class SIP-based interactive communications within and between mobile, fixed-line and transit networks.

GSMA PathFinder and Acme Packet

The PathFinder service facilitates IP interoperability by translating telephone numbers to IP-based addresses for SIP-based services such as voice, messaging, presence and video sharing. Based on Carrier ENUM, PathFinder is a centralized routing database for dynamic route selection and is available to mobile and fixed network operators.

Acme Packet?s Net-Net SD and Net-Net SR query the PathFinder databases using the industry-standard ENUM protocol. Using these databases, Acme Packet products make dynamic routing decisions within the core IP network and to the PSTN and other IP networks using a wide selection of parameters. Acme Packet?s Net-Net SD and SR support the various PathFinder service types, including SIP or H.323-based VoIP, presence and messaging.

The GSMA PathFinder service demonstrates the principals of Acme Packet?s OSR architecture, which features Acme Packet?s session routing proxy or session border controller working in conjunction with best-of breed routing database products and services. Acme Packet?s OSR architecture addresses scaling problems when SIP session routing decisions become much more complex, requiring a dynamic, real-time routing decision for each individual session for multiple sources and destinations within a network. Acme Packet?s OSR architecture is deployed in several tier-one fixed, cable and wireless operators? networks around the world.

Acme Packet SBCs have been widely used in GSMA IPX trials since 2007?both by IPX carriers or connecting service providers. The Net-Net SBCs were most recently used at IPX trials conducted by Telecom New Zealand International. This builds upon Acme Packet's IPX trial experience with customers including Belgacom International Carrier Services, Telefónica International Wholesale, Telenor Global Services and others.

13/11/2008 - Junction Networks Picked as Voice Provider for edgeBOX

Editor's Note: We personally use Junction Networks for our SIP termination and we have been satisfied with their service. I have also chatted with their CEO and he was very knowledgeable and courteous.

New Jersey-based Critical Links has chosen VoIP service provider Junction Networks as the preferred provider for its edgeBOX all-in-one, voice-and-data appliance.

edgeBOX is an integrated device aimed at the SMB market. It includes a full-fledged IP-PBX, wireless access point, router, file, e-mail and VPN server. The edgeBOX will come preconfigured with a free trial of Junction Networks’ SIP or IAX VoIP trunking and PSTN gateway.

“Junction Networks’ VoIP service is part and parcel of our no-headache, low-maintenance, low-cost approach to office communications,” said Abdul Kasim, Critical Links’ vice president of worldwide marketing. “edgeBOX customers can use Junction Networks’ service as soon as they activate their telephone numbers. Their voice service comes in at a fraction of the cost of traditional PSTN carriers, but meets our tests of reliability, voice quality and customer service.”

Junction Networks’ free trial for edgeBOX waives the $9.95 monthly service charge and supplies $10 of inbound or outbound calling services for the first 30 days. After the trial period, the service is provided on a pay-as-you-go basis; there are no long-term contracts or penalties for cancellation at any time. Calls between edgeBOX extensions, whether on-premises or around the world across Internet connections, will incur half-cent-per-minute charges. Web-based user portals show real-time usage records.

Junction Networks President Rob Wolpov said, “While our service is easy to connect to any SIP- or IAX-based PBX, it’ll be even easier to use on an edgeBOX, with configuration built in. We cater to SMB customers, and edgeBOX customers are SMBs who want everything in one device — including the know-how.”

Source: Phone Plus Mag

05/11/2008 - DIGITALK Now Certified "XConnect-Ready"

XConnect, the world’s largest provider of VoIP federation peering services, has announced that the DIGITALK SIP Application Server has been certified XConnect-Ready.
Eli Katz, XConnect CEO, said the impact for customers would be to make VoIP federation-based routing quick, simple and easy.

03/11/2008 - DIGITALK SIP Application Server Now Certified ''XConnect-Ready''

DIGITALK SIP Application Server has been certified XConnect-Ready having completed interoperability testing based on SIP signaling and ENUM queries with XConnect Federations.

XConnect?s certification ensures DIGITALK customers, such as Telfort, BT, and Cable & Wireless, will be able to rapidly connect to XConnect Peering Federations to reduce the costs of terminating VoIP calls to millions of telephone numbers in the XConnect registry, protect their networks from spam-over-Internet-telephony attacks, and reliably deliver new IP communications services across disparate and often separate mobile, wireline and IP based telephony networks.

XConnect enables feature-rich multi-media communication, reduces capex and opex and enhances call quality for service providers via its multi-lateral XConnect Alliance, DirectRoute and Private Federations services. XConnect Federations are carrier-neutral peering environments that deliver complete signaling interoperability, intelligent ENUM Registry services, and VoIP security for the interconnection of XConnect Members, which include voice over broadband providers, MSOs, and PTTs. The XConnect Ready Partner Program is an ecosystem of vendors and solution providers dedicated to facilitate service provider peering.

30/09/2008 - RakSIP Service Connect Any Mobile, SIP Client or IP Phone to Raketu VoIP

Raketu releases its RakSIP service. The new service allows users to connect any third party mobile or desktop SIP software client, or IP Phone, or SIP hardware device to Raketu so that they can take advantage of Raketu's RakOut calling rates. From any third party SIP device, Raketu users will be able to login to Raketu and make phone calls no matter what device they are using or where they are in the world.

Users can use the built-in SIP client on Nokia or BlackBerry mobile device, or download a SIP client for their iPhone or WinMobile mobile device, to connect to Raketu and start making calls immediately. Users can also connect IP Phones, ATA devices and SIP desktop software to the new RakSIP service. Raketu users that already have Raketu's RakIn service can immediately begin taking advantage of the new RakSIP service. Raketu users that are not RakIn customers, can simply signup and subscribe to the RakSIP service for $.99 per month.

Raketu's RakSIP Service Features

With the new RakSIP service users are able to make outbound RakOut calls at Raketu's free or incredibly low rates. Raketu's RakIn service provides the lowest cost for the inbound calling numbers and includes the RakSIP service, voicemail, callerID, forwarding, and more. RakSIP is in addition to Raketu's existing communications, information, entertainment and social networking features, where users can make international calls computer-to-computer, computer-to-phone, or phone-to-phone, send sms-text messages, instant message, and email totally free or at Raketu's ultra-low rates -- all from any device, anywhere in the world, mobile or desktop/laptop.

Raketu's new RakSIP service is available from the Raketu website and is accessible from the Raketu download client. Users simply login, make a payment, activate their RakSIP service, configure their preferred SIP mobile, desktop or device, and begin making phone calls. Current Raketu users can start using the new services immediately, and new users can sign up at http://www.Raketu.com.

26/09/2008 - SecureLogix Offers Free VoIP Security Tool

SecureLogix Corporation has announced that its releasing a free suite of custom Voice-over-IP (VoIP) security assessment tools.
Downloadable from the company's Web site, the tools can be used to assess susceptibility to a wide variety of SIP threats, including Denial-of-Service (DoS) and Man-in-the-Middle attacks, eavesdropping, audio insertion and deletion, and even call teardown.

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