VoIP Monitor

netTALK Offering Unlimited SMS Texting to the U.S. & Canada Using Your netTALK DUO Home Phone Number

VoIP Monitor — Wed, 12 Dec 2012 22:31:52 GMT

netTALK.com announces the release of its Unlimited SMS texting service using your DUO home phone number. netTALK customers can now send and receive SMS text messages from the netTALK Text Android app by logging in with their DUO credentials for SMS texting. Even while traveling internationally customers can still send and receive SMS text to and from the U.S. & Canada (where Wi-Fi or 3G/ 4G access is available).

“The addition of this app further enhances the entire netTALK experience and our innovation underscores why we are the fastest growing company in the nation’s fastest growing industry,” said Cedric Guillory, netTALK’s Senior Director of Marketing. We’re always looking to add feature packed products at an affordable price. With our new netTALK Text Android app with unlimited SMS texting, no contracts and no monthly bills, we are doing just that!” Guillory did emphasize that at this time the app is for SMS texting only and does not work with images or attachments, but it comes with the following features and that an iOS (Apple) version will be released soon.

netTALK Text Android App Features:

Works on 3G, 4G, or Wi-Fi
Supports Android tablets
Push Notification for new messages
Sound Notifications for new messages
Easily call any contact from a conversation
Easily search specific words in conversations
Sync with native contacts
Voice-to-Text Message Creation (if supported by device)
Saved conversations so you can sign in to multiple devices

“Even when your device is off you will receive offline messages; the app automatically saves them for you,“ added Guillory. “To take advantage of this outstanding app you must have a netTALK DUO account."

Download the free netTALK Text app at Google Play, Amazon App Store (Kindle Fire Edition) and sign up to start making Unlimited SMS text messages today!

The Company also recently announced its “Season to Celebrate” promotion offering a $10 Free International Credit with the purchase of any DUO device direct or from any local or online retailer. The promotion runs through Christmas Day (Dec. 25, 2012).

Security Researcher Demonstrates Enterprise VoIP Phone Hack at Recent Amphion Forum

VoIP Monitor — Wed, 12 Dec 2012 22:24:43 GMT

During the recent Amphion Forum, a conference where device and mobile security experts from different disciplines gather, Ang Cui, a fifth-year grad student from the Columbia University Intrusion Detection Systems Lab, demonstrated how connected devices such as networked printers and VoIP phones can be easily hijacked to give intruders virtually unlimited remote access to extremely sensitive information and allow them to eavesdrop on private conversations. The Amphion Forum is hosted by Mocana, a leader in device and mobile security.

Using a common Cisco-branded VoIP phone, Cui inserted and then removed a small external circuit board from the phone’s Ethernet port—something Cui asserted could be easily accomplished by a company visitor left unattended for a few seconds—and starting using his own smartphone to capture every word spoken near the VoIP phone, even though it was still ‘on-hook.’ While he did not specify the precise vulnerability, Cui said it allowed him to patch the phone’s software with arbitrary pieces of code, and that this allowed him to turn the Off-Hook Switch into what he called a “funtenna.” According to Cui, once one phone is compromised, the entire network of phones is vulnerable. Cui later said he could also perform a similar exploit remotely, without the need to insert a circuit board at all.

The vulnerability Cui demonstrated was based on work he did over the last year on what he called ‘Project Gunman v2’, where a laser printer firmware update could be compromised to include additional, and potentially malicious, code. With this, it becomes possible to remotely compromise a printer located within the organization’s firewall and eavesdrop on documents being printed or stored, without ever setting foot on the premises. The compromised printer could then be used to launch other attacks on the internal network. The demonstration at the Amphion Forum in San Francisco took such an attack even further.

Cui pointed out that current security solutions don’t work with embedded systems like VoIP phones and printers and code signing isn’t enough. “Signing files doesn’t make the files secure,” Cui said.

He also said that routers, printers and phones are general-purpose computers without host-based intrusion systems or antivirus protection built in, so they make attractive targets. Further, they often lack encryption for data in motion or at rest.

Cui’s research was carried out as part of a DARPA CRASH (from the I2O office) and IARPA Stonesoup Program, and he recently briefed agencies of the U.S. federal government about the potential for a serious attack on all its Cisco Unified VoIP phones.

“The VoIP phone vulnerability demonstrated at the Amphion Forum was a stark reminder of the need to address the device security mess. The sad fact is that most devices connected to corporate networks, like printers and VoIP phones, are almost totally unsecured,” said Kurt Stammberger, CISSP, vice president of market development at Mocana and chair of the Amphion Forum. “The Amphion Forum is a unique event where thought leaders from academia, business, government and technology can gather to discuss the threats and opportunities presented by the unprecedented proliferation of mobile and connected devices that are creating the Internet of Things.”

The Amphion Forum was founded to provide a medium for stakeholders in the smart device economy to share solutions and forge a clear direction for the future of the Internet of Things. The most recent event was held in San Francisco on December 5 and attracted more than 350 participants and thought-leader presenters, making it the largest and most successful Amphion event since it was founded in 2011. Event organizers believe that by fostering a World Economic Forum-type environment, where big thinkers can share ideas for some of the most pressing issues facing the global device infrastructure, safer medical electronics, increased energy security and more secure industrial automation.

Rebtel Launches Voice Platform for iOS and Android Developers

VoIP Monitor — Wed, 12 Dec 2012 22:14:30 GMT

Rebtel announces the private beta launch of its new SDK for iOS and Android. The Rebtel Voice Platform introduces a powerful yet simple tool that will now allow developers to integrate voice capabilities into any app in 15 minutes.

Now any app developer looking to add a live calling feature has the ability to stream voice to their customer base. Gaming apps like Poker and apps that offer social networking features such as picture sharing comprise the bulk of apps developed today, and are prime examples of specific use case scenarios where smartphone users could benefit from free app to app calls.

Using Rebtel's Voice Platform for in-app calling means game developers can offer players the opportunity to phone a friend for strategy tips without ever having to leave the game environment, and dating apps can now offer the ability for real-time voice calls for better matchmaking possibilities.

The launch of the platform comes at a critical time for developers who are looking into new ways to monetize their user base, while making their apps more social to create and extend value to consumers. With voice no longer being a stand-alone service provided by operators, the Rebtel Voice Platform also addresses the emerging trend of an ever-growing number of smartphone users who are turning to OTT (Over The Top) services for free or cheap calling alternatives at the expense of traditional carriers.

“Voice is a feature that can significantly augment the user experience in a variety of verticals and in the long term we see the Rebtel Voice Platform as a game-changing tool for developers,” said Rebtel CEO Andreas Bernstrom. “With the launch of this free service, thousands of content publishers and smartphone developers will have the opportunity to enrich their applications with a voice layer thereby increasing usage and appeal, or they can even distribute their own branded mobile VoIP app without any large back-end commitment.”

Rebtel’s software development kit makes for an intriguing proposition for developers in most app categories and the company has already attracted inaugural launch partners. They include Palo Alto-based, VIVfone a consolidated mobile CRM app that fundamentally improves the business calling, information gathering and transaction services for consumers and Stockholm-based MobisleApps that develops applications in the lifestyle, dating, and productivity categories.

“Rebtel’s SDK allows us to bring a simple, yet powerful and thoroughly road- tested calling functionality into our app. It connects our users in an intuitive but elegant and reliable way,” said Fredrik Olofsson, CEO, MobisleApps. “There is a tremendous value-add for us in being able to extend the utility and interactivity over a wide range of applications, which we feel will give us a lift in downloads, as well as in driving longer and more frequent user interactions.”

“We chose to work with Rebtel because starting with superior quality HD calls lays the foundation for a robust service that consumers will love,” said Ratesh Dhir, Co-Founder and VP Business Development, VIVfone.

Rebtel’s Voice Platform SDK tool-kit for iOS and Android includes the following:

Free SDK tool for any developer of any size
Swift integration that can be completed in 15 minutes
Synchronous free app to app calling over Wi-Fi or 3G
Crystal clear quality with HD voice technology based on bandwidth adaptable iSAC codec
All signaling and codec handling wrapped into one easy package, with no back-end support or infrastructure commitment needed

On the heels of the news, Rebtel will be delivering a dedicated panel session in promotion of their new SDK launch during the Appnation conference in San Francisco on Wednesday, December 12th from 3:10pm to 4:00pm. The session titled, “Give Your App a Voice - How to Add a Social Layer to Your App,” will focus on how developers can successfully integrate this powerful new add-on service to drive customer acquisition and retention.

App developers interested in more finding out more about Rebtel’s Voice Platform SDK can also sign up from the website at www.rebtel.com/sdk or see a video.

3CX and snom Team Up to Deliver Advanced Unified Communications Solution

VoIP Monitor — Wed, 28 Nov 2012 21:08:01 GMT

3CX announces a strategic partnership to provide increased flexibility for organizations using snom phones with 3CX Phone System.

This partnership between two prominent European technology companies merges the best of breed PBX and IP phone technologies to provide diverse businesses and institutions with a complete phone system solution. snom phones are easy to deploy, as they are fully supported for plug and play provisioning, and all snom phone features, such as direct dial and call recording, are available with 3CX Phone System.

"We are extremely pleased to announce this strategic new partnership, which allows us to offer our joint customers a more advanced unified communications solution that combines best of breed IP Phone and PBX technology," said Nick Galea, 3CX CEO. "Both snom and 3CX can now innovate faster, and offer a more advanced and better integrated telephony platform. This will benefit our customers and partners, and allow 3CX to accelerate its growth."

3CX will continue to focus on developing software aspects in tandem with snom, which will focus on hardware, to provide a superior Unified Communications solution with innovative new features in the near future. International sales and partner training will also be made available.

"Our snom desktop IP phone portfolio's tight integration with the 3CX IP PBX platform delivers advanced, feature-rich VoIP solutions and offers tremendous value to businesses," said Dr. Michael Knieling, snom technology AG Chief Operating Officer. "This partnership highlights snom's commitment to SIP interoperability by blending snom's 'best-of-breed' SIP endpoints - the snom 3xx, snom 7xx and snom 8xx series IP phones - with 3CX's award-winning Windows-based IP PBX software."

VoIP Innovations Launches Private Label Service for VoIP Resellers

VoIP Monitor — Tue, 27 Nov 2012 21:28:44 GMT

The VoIP industry has become very popular over the past few years. However the opportunity to sell VoIP services has been limited to those who can run a switch and have the technical knowledge to do so. VoIP Innovations is hoping to change that by providing an entry-level platform designed for entrepreneurs with limited technical knowledge.

VoIP Innovations is announcing the official launch of Titanium III Private Label service for VoIP resellers. This solution enables resellers to quickly and inexpensively capitalize on the booming VoIP business. Titanium Private Label allows resellers to brand, market, and sell residential and business VoIP services as their own. Resellers can choose to sell VoIP Lines, Sip Trunks, or Hosted PBX service through a web-based platform. This service permits resellers to fully manage their business through the cloud and avoid the need to run their own switch.

“We looked at several platforms out in the market and found that they were too expensive and cumbersome.” said Jason Tapolci, President of VoIP Innovations.

The company’s proprietary and innovative software solution was built from the ground up and is fully hosted in the cloud. This means no large capital expenditures or overhead expenses are required to own, operate, maintain, and troubleshoot a switch. Resellers can now have a fully-enabled VoIP switch with all the functionality and features that residential and business customers demand.

Titanium III Private Label was designed to make running your VoIP business as painless as possible. For this reason, we have built unprecedented automation into every business function. Daily tasks such as account sign-ups, support, number porting, account notifications, and more have been 100% automated.

“A reseller can quickly provision their first device within 15 minutes of getting their log-in information” explained Corey Potts, VP of Sales for VoIP Innovations.

Additionally, VoIP Innovations bundles all services like e911, CNAM, and Caller ID into one price point that’s easy to understand. All services are flat rate which makes billing customers extremely easy. Best of all, resellers can use any IP Phone, ATA or softphone of their choice.

PCN Introduces UltraEdge VoIP System for Any Grade Legacy Copper

VoIP Monitor — Mon, 26 Nov 2012 19:02:41 GMT

PCN announces the introduction and availability of its UltraEdge system for rapid cost effective deployment of VoIP and Unified Communication data networks.

With a single Ethernet connection to access the Internet or outside data networks, the UltraEdge system requires limited IT knowledge for installation and enables access and transport for IP device connectivity at every edge. Using any grade copper the UltraEdge VoIP system easily interfaces to existing junction boxes, legacy PBX consoles, or other wiring infrastructure already in place.

Providing maximum capabilities and feature rich applications, the system is a true IP networked solution leveraging PCN's high-performance 19" Rack Mount Multi-Channel Server (PCN3485-MCS4) for 10/100 IP transport to every UltraEdge module (PCN3485-SCC1) which enables access for VoIP enabled phones or other devices. Unlike media converters, each UltraEdge module is a true Ethernet switch where each end point has access for up to three Ethernet RJ45 ports.

Unlike DSL extenders and converter technologies, the PCN UltraEdge solution is built on PCN's patented "Dynamic Adaptive Channeling" data prioritization algorithms which run over 500 times per second in real time scanning, monitoring and managing the physical layer of legacy copper looking for anomalies and moving IP data to its most optimum spot on the wire. This not only provides 3X the noise immunity of DSL, but enables distance stretching. When combined together this not only allows more reliability and robustness of your VoIP system, but provides access to IP at distances up to 1500 feet and far greater than other solutions.

Across the world there is an abundance of legacy copper that is unusable for IP enabled Ethernet systems and where running new structured cabling is either too expensive, or requires significant shut down and disruption to business operations. This is true for government buildings, hospitals, historic buildings, factories, college campuses and office buildings that have Cat-3 or older wiring.

Without having to run any new structured cabling, the UltraEdge system allows owners and integrators to instantly leverage and take advantage of legacy wiring as if their facility was just wired with new Cat 5/6 cabling.

The UltraEdge VoIP system supports high-speed 10/100Base-T Ethernet-on-Demand across any grade copper wiring topology. This includes multi-drop, daisy chain, point to point and others allowing access in the far reaches of a building or installation.

Davis also commented, "Customers love the fact that they can easily install the multi-channel router at their old PBX locations or at other junction wiring; and then simply drop a plug-and-play UltraEdge module anywhere to have immediate access to high-bandwidth VoIP networks."

The UltraEdge VoIP system supports third party standard Layer 2 and Layer 3 Ethernet switching, QoS and a variety of network security features including port security, multi-level passwords, DoD protections, SSH and SSL for encryption. When combined with PCN's physical layer technologies the result is a comprehensive VoIP and Unified Communications system that is more reliable, more robust and one that goes further.

Inflow Communications Expands VoIP Business Phone System and HD Video Conferencing Coverage in San Francisco

VoIP Monitor — Mon, 26 Nov 2012 18:58:35 GMT

Inflow Communications has expanded its consultation, installation and follow-up support services with the acquisition of San Francisco VoIP phone system provider, Ketchum Integration.

Inflow’s expanded service offering builds on the core competencies of both companies. Ketchum Integration provides single and multi-site VoIP installation. Inflow Communications has been a provider of strategic consultative services and installation of ShoreTel VoIP business phone systems and HD video conferencing with Unified Communications integration since 1997. The new San Francisco location joins Inflow Communications’ other offices in Oregon and Washington.

“Our new partnership with Ketchum is very exciting for us,” says Inflow Communications President Travis Dillard. “With this expansion, we have improved our ability to service customers in and around San Francisco and ensure they receive the same level of support our customers in Portland and Seattle enjoy.”

Inflow Communications was recently awarded ShoreTel’s Outstanding Customer Achievement Award for 2012, and is a Top 30 ShoreTel partner based on customer satisfaction reporting. Inflow’s unique combination of consultation and post-installation service have set the company apart from competitors in the business phone system and video conferencing market.

The acquisition took effect November 1st.

Fraunhofer IIS Makes Full-HD Voice Available for VoIP Apps

VoIP Monitor — Tue, 13 Nov 2012 23:51:22 GMT

Fraunhofer IIS will showcase how its Full-HD Voice codec, Enhanced Low Delay AAC, supercharges the communication quality of any VoIP app, be it for smartphones, tablets or televisions at CES 2013.

Natively supported in iOS and Android, the AAC-ELD codec is now readily available to app developers and service providers to offer superior VoIP applications that run at the Full-HD Voice quality level. This delivers a completely new same-room communication experience to the end-user.

"From mobile phones to TVs, consumers expect the highest audio quality from their entertainment experiences. There is no reason the expectations for VoIP communication should be any different," said Harald Popp, head of the Multimedia Realtime Systems department at Fraunhofer IIS. "By providing developers direct, native access to the best codecs to deliver Full-HD Voice VoIP apps, consumers can expect the best quality possible."

Fraunhofer's Full-HD Voice technology is already widely used to provide CD-like audio quality for professional video conferencing systems, as well as consumer video telephony applications, such as apple FaceTime. With AAC-ELD now available in iOS and Android, this same level of professional communication quality can be readily deployed in consumer VoIP apps.

App developers can access the natively-integrated AAC-ELD codec through the Fraunhofer FDK Codec Library in Android since version 4.1 and in iOS since version 4.0 as well as in OSX, allowing cross-platform deployment of Full-HD Voice.

Background Information on Audio Quality in Communication Systems

Historically, the audio quality of VoIP services has been similar to legacy telephony, with a maximum audio bandwidth of 3.4 kHz. Recently, some services have offered a higher audio bandwidth of 7 kHz, which is still quite limited compared to 20 kHz which is the maximum bandwidth humans are able to hear. Another reason for the limited audio quality of phone calls is usage of speech codecs. As their algorithms emulate a single human vocal tract, they are only optimized for a single speaker in a quiet environment. Speech codecs do not deliver other signals clearly, such as ambience sounds, music and background noise. The result is compromised audio quality in conversations and highly distorted background sounds or music.

With the state-of-the-art MPEG-4 communication codec AAC-ELD, app developers can easily offer a level of performance that removes the limitations of speech codecs and offers the full audio bandwidth of up to 20 kHz, giving voice, music, and ambience sounds equal treatment.

Voxbone Global DID Numbers Enhance Aculab Cloud Platform

VoIP Monitor — Mon, 12 Nov 2012 21:28:47 GMT

Voxbone announces it is providing international direct inward dial numbers to Aculab for Aculab Cloud, a true cloud-based platform that makes it easy for developers to create on-demand automated telephony applications that make, receive and interact with calls.

Aculab selected Voxbone because the company’s VoxDID service enables Aculab to easily enhance its services with instant provisioning through Voxbone’s convenient application programming interface solution. Voxbone maintains a supply of geographic and national phone numbers for incoming communications from more than 50 countries and 4,000 cities.

Aculab Cloud enables functions including transferring calls, playing messages, mixing calls together in a conference, playing text-to-speech prompts/messages, analyzing call progress, detecting answering machines, queuing calls, and making thousands of simultaneous outbound calls.

“We specialize in enabling solution providers to create and deliver high-performance inbound and outbound contact center applications, speech-enabled IVR, unified communications and other services to enterprise customers,” said Faye McClenahan, Head of Strategic Marketing at Aculab. “Voxbone’s DID numbers and high-quality network services enable us to enhance the resources developers can use to meet their customers’ needs for cloud-based communications.”

DIDs are another example of the ongoing industry migration to VoIP and cloud services. Like many service providers, Aculab has followed the trend from on-premises, hardware-based solutions to cloud-based offerings.

“As more service providers create services and applications based on the cloud, we are seeing increasing demand for our inbound communications services,” said Voxbone CEO Rod Ullens. “By using our VoxDID service, Aculab will enable its customers to build better apps more easily and rapidly.”

VoX Communications Mobile VoIP App to be Available on the Samsung App Store

VoIP Monitor — Thu, 08 Nov 2012 22:49:04 GMT

VoX Mobile VoIP calling application is now available on the Samsung app store for Android devices, which are projected to exceed one billion in service in a short period of time. Users of Samsung devices are now able to download the VoX "Pay and Go" subscription-free calling plan, or other VoX calling plans directly to their Samsung devices.

The subscription-free VoX plan called "Pay and Go" can be topped up with amounts as little as one dollar and enables calling anywhere in the world without paying a recurring monthly fee.

VoX's Chief Information Officer, Mark Richards, noted, "We are highly encouraged by the increased conversions brought to us by our newly-launched Pay and Go plan, so we decided to broaden the reach of our app by making it available on Samsungapps.com. We note that the Samsung store is ranked in the top 5,000 sites in the world and the app store icon is now installed on every Samsung smartphone and tablet that is shipped globally."

"We believe that if more people see our app, more people will use it," continued Richards. "We are very pleased that we are one of only a very few mobile VoIP apps that have been accepted in the Samsung App Store."

One in five smart phones sold in the U.S. is from the Samsung Galaxy series. Samsung has sold over 50 million Galaxy SI and SII phones and has sold over 20 million Galaxy SIII phones in the first 100 days since its launch. The VoX app went through rigorous device testing by Samsung and the VoX app is approved to run on 71 Samsung devices, including all popular smart phones and tablets, and was made available to over 50 countries.

A VoX user can subscribe entirely on the phone or tablet and choose a U.S. phone number without leaving the interface, which makes the sign-up process more elegant than many competitors. The mobile VoIP paid plans are a low cost of entry for a high quality and reliable mobile VoIP service offering. New subscribers can download the app for a free 60 minute trial and make calls to any country on the unlimited calling list or sign up for the Pay and Go plan for as little as a dollar, literally a "pay as you go" plan with no monthly subscription fees.

Mobotix Adds VoIP Supply to Global Partner Network

VoIP Monitor — Fri, 26 Oct 2012 20:51:02 GMT

Shopping around for IP surveillance cameras is one thing but, deciding which one of those thousands of cameras is the right fit and will actually perform as advertised is quite another thing.

That's why, for those customers that need trustworthy help to make the best decision, we're happy to announce that the Mobotix Global Partner Network now includes VoIP Supply as a Registered Mobotix Partner.

Not just a fancy title, what a Registered Mobotix Partner means for customers is:

Customers receive combined attention because as a Partner, VoIP Supply receives additional high-performance support from Mobotix to design a solution.
Customers are assured that VoIP Supply is fully trained to support them because Partner level status requires that we attend certification seminars.
To offer the highest level of help to customers, VoIP Supply has to help itself by investing a high level of commitment to the Partner Program. For example, VoIP Supply is required to have demonstration products on hand so that our people know how to use Mobotix equipment.

"We want the international channel network program to provide our partners with the best support possible and to improve their business transactions," explains Dr. Magnus Ekerot, Chief Sales Officer of Mobotix.

Accessing VoIP Supply for Mobotix IP cameras is almost like having access to Mobotix themselves. As a Partner VoIP Supply can tap into another level of support from Mobotix from the beginning of a project right through to the end

"Mobotix supplies some of the toughest, most versatile IP cameras on the market," said Garrett Smith, Chief Marketing Officer at VoIP Supply. "Now as a Mobotix Registered Partner, VoIP Supply can provide the best tech support when delivering rugged Mobotix quality to any surveillance project."

Grandstream Announces 1-Line IP Telephone with LCD Display

VoIP Monitor — Tue, 23 Oct 2012 21:22:58 GMT

Grandstream Networks introduces the GXP1160/1165. Along with a graphical LCD display that supports multiple languages, this low cost, single-line IP phone features high quality audio, integrated Power over Ethernet (GXP1165 model only), 2 call appearances, rich telephony functionalities, and Electronic Hook Switch capability with Plantronics headsets. The GXP1160/1165 is fully interoperable with leading IP Telephony platforms (such as Asterisk, Broadsoft, Metaswitch, 3CX, etc.) and various Internet Telephony Service Provider networks, and can be easily mass deployed and securely managed from a central location.

GXP1160/1165 Feature Highlights

128x40 pixel graphical LCD display with support for 11 languages
Single SIP account, up to 2 call appearances, 3 XML programmable context-sensitive soft keys, 3-way conference
Phonebook with up to 500 contacts and call history with up to 200 records
Automated personal information service (e.g., local weather), music ring tone/ring back tone
Dual switched auto-sensing 10/100Mbps network ports, integrated PoE (GXP1165 only)
Automated provisioning using TR-069 or AES encrypted XML configuration file, SRTP and TLS for advanced security protection, 802.1x for media access control
Electronic Hook Switch capability with a number of Plantronics headsets

Pricing and Availability

The GXP1160/GXP1165 will be generally available by the end of November 2012 through Grandstream's worldwide distribution channels at a list price of US$55 and US$59 respectively.

snom m9 Cordless DECT Phone is Now Interoperable with BroadSoft's BroadWorks

VoIP Monitor — Mon, 22 Oct 2012 21:09:57 GMT

snom announces the completion of interoperability between the snom m9 cordless DECT phone and BroadSoft's BroadWorks hosted Unified Communications platform. The snom m9 has completed interoperability testing with BroadWorks Release 18.SP1.

snom's m9 is now available to BroadSoft's more than 500 global telecommunications service provider customers in order to deliver enhanced, personalized communications and entertainment services to their businesses and residential subscribers. BroadSoft serves 18 of the top 25 largest telecommunications service providers, based on revenue, in more than 65 countries. With the completion of interoperability testing, BroadWorks powered end-users can benefit from the latest features and functionality when choosing a snom m9 DECT phone for their business.

"snom is committed to interoperability with a broad array of SIP-based platforms and is pleased to have completed the recent testing with BroadWorks," said Michael Knieling, chief operating officer of snom technology AG. "BroadSoft is at the forefront of the IP telephony industry and offers a mature, reliable platform, so we are extremely pleased to make our snom m9 cordless DECT phone widely available to BroadSoft customers around the globe."

The snom m9 combines snom's highly-interoperable IP communications firmware with an advanced DECT handset, giving business users an affordable, cordless VoIP phone. The m9 offers interference-free communication through a dedicated DECT frequency band, provides 100+ hours of stand-by battery time, features wideband audio, supports up to nine handsets and four simultaneous calls per base station and delivers superior speech quality, demanded by small business and residential users alike.

Standard features include hands-free mode and calling line identification, as well as typical mobile phone features, such as an on-board address book, calendar, calculator and alarm functions. Advanced features include IPv6 support, voice encryption, pre-installed security and color picture caller ID. The snom m9 is also fully functional in any open standard SIP environment.

Additional products in the snom portfolio will complete interoperability testing with BroadWorks in the coming months.

The SIP Forum Announces Dates and Opens Registration for Third Annual SIPNOC

VoIP Monitor — Mon, 22 Oct 2012 21:05:54 GMT

The SIP Forum announces it will host the third annual SIP Network Operators Conference (SIPNOC 2013), an educational conference focused on the challenges and opportunities of deploying SIP-based carrier services worldwide. Building on the success of past conferences, this year the SIP Forum will add an extra day of special workshops and sessions, designed to educate service provider technical staff on best practices and strategies for the successful implementation of SIP-based services and applications.

SIPNOC 2013 will be held at the Hyatt Dulles Hotel in Herndon, Va., from April 22-25, 2013, and will continue its practice of providing leading technical and operations personnel from the global carrier community with educational, non-commercial and technical content focused on the real-world challenges of deploying SIP services in global IP networks.

Early bird registration is available at http://www.regonline.com/sipnoc2013. More details about SIPNOC 2013 are available on its conference website at http://www.sipnoc.org.

"The growth and acceleration of SIP services within the international service provider community - and the accompanying technological, logistical and businesses challenges - was the original impetus behind SIPNOC and remains our guiding motivation," SIP Forum Chairman Richard Shockey said. "We're pleased to bring this unique gathering for the network operator community back for a third year, once again providing a non-commercial, technically-oriented setting for discussing and sharing ideas and knowledge about SIP implementation."

SIPNOC 2013 will once again bring together communications industry leaders to learn, discuss and formulate new ideas and strategies to address the challenges and opportunities for SIP-based carrier services in fixed and mobile IP network environments. The conference, which is designed specifically for SIP network operations personnel and engineering staff, will feature well-known industry speakers and a number of highly technical educational and instructional panels and sessions. The SIP Forum will also host networking events at the conference and offer a series of informal "Birds of a Feather" sessions, which encourage discussion on a variety of topics in hallways, available meeting rooms and break-out areas.

"SIPNOC demonstrates the SIP Forum's ongoing commitment to serving as the industry's non-commercial think tank for international SIP interoperability and deployment," said Marc Robins, President and Managing Director of the SIP Forum. "Our third annual conference will expand upon key issues identified in previous discussions and through our SIP Forum Task Groups, with the aim of delivering high-level technical support and guidance to the broad service provider community worldwide."

SIPNOC 2013 will build on critical themes discussed at past events, including: application development and testing; SIP trunking interoperability and the SIP Forum's SIPconnect 1.1 technical specification; Fax over Internet Protocol (FoIP) interworking; implementing SIP over IPv6; user-agent configuration; emergency services; policy servers; security; operational issues; call routing and peering; troubleshooting and monitoring; SIP Interconnection; HD-Voice Deployment Challenges; and Video interop issues.

"SIPNOC is an international gathering where communications engineers and network professionals can discuss and troubleshoot the real-world intricacies of working with SIP every day," Robins added. "We expect this year's conference to build on our past successes, and explore new complexities as we strive together, as a community, to deploy SIP-based services in diverse network environments worldwide."

Attendees at SIPNOC 2013 will include telecommunications providers, major backbone operators, interconnect and wholesale solution providers, ISPs, ITSPs (Internet Telephony Service Providers), cable operators and wireless network operators, as well as large enterprises deploying major SIP initiatives. Industry stakeholders - such as network equipment vendors, government agency representatives and academic research organizations - are also encouraged to attend.

The SIP Forum enjoys an international reputation for developing key educational events around SIP deployment. The organization's SIPit series of interoperability testing events regularly provides a test bed for SIP-based applications and equipment and has been heralded as critical for the development of new products and services in the industry. The SIP Forum is also home to committees and task groups comprised of industry experts examining a myriad of SIP-related topics, including the use of SIP with FoIP, video, SIP over IPv6 and user-agent configuration. SIPNOC 2013 Corporate Sponsorship Opportunities