VoIP Monitor

Voipswitch Now Interoperable with Sangoma's NetBorder Express Gateway

VoIP Monitor — Thu, 09 Jul 2009 16:38:18 GMT

Voiceserve and Sangoma announce the completion of successful interoperability tests between the companies' product lines. Voipswitch's award-winning softswitch, and Sangoma's NetBorder Express Gateway are both based on the SIP communication protocol for VoIP.

This successful validation of interoperability which included comprehensive functionality and stability tests ensures that users of the Voipswitch software can now select Sangoma's NetBorder Express Gateway as a gateway device to create cost-effective switching applications.

This additional facility now available for Voipswitch clientele, the use of Sangoma hardware cards, enables its customers to deploy PSTN trunks with extremely modest costs. The Sangoma board is simply inserted into the Voipswitch server. Clients will not need any expensive external gateways. The calling card, wholesale termination and IP PBX scenarios will be enhanced dramatically with the NetBorder Express Gateway installed. Also, having a Sangoma board will act as a secure backup in the event that the internet link is severed, thus permitting traffic to pass through PSTN trunks.

Sipera Systems Uses Empirix’s Hammer VoIP Testing Solution

VoIP Monitor — Wed, 08 Jul 2009 16:47:55 GMT

Sipera Systems now employs Empirix’s Hammer VoIP testing solution, enabling them to develop superior VoIP and UC security solutions. With the Hammer solution, Sipera is able to discover, prioritize and correct potential issues during the pre-deployment phase, reduce the cost of bringing its products to market and ensure that its UC-Sec security appliances can simplify and protect VoIP and UC implementations with comprehensive security.

During its evaluation, Sipera confirmed that the Hammer solution provided unrivaled performance testing for high capacity throughput, platform stability and reliability. To ensure the quality and reliability of their products, Sipera’s development and testing team now relies on Hammer’s extensive media support, security test functionality, high load handling capabilities and ability to provide detailed QoS metrics. Hammer also provides broad signaling and security protocol test coverage (including SIP, SCCP, TLS and SRTP).

Hammer testing solutions emulate the realistic behavior of users and devices interacting with voice-centric applications. It makes calls, verifies features, measures voice quality and determines performance, as well as proving system and multi-vendor interoperability for VoIP, NGN and IMS architectures.

Digium Broadens the Focus of AstriCon 2009

VoIP Monitor — Tue, 07 Jul 2009 17:32:22 GMT

Digium is expanding the focus of the AstriCon Open Source Telephony Conference and Exhibition, to be held October 13-15, 2009, to reach the broad group of professionals who design and manage large enterprise, government and carrier networks. Celebrating the tenth anniversary of Asterisk’s release as an open source project, the conference will be rich with content for Asterisk software developers, Digium’s channel and technology partners, and business users from small, medium and large companies. This year’s conference also features a dedicated track that focuses on Asterisk’s move into mainstream large enterprise and government markets.

The event, now in its sixth year, attracts hundreds of people who work with phone systems, unified communications and VoIP. Business, technical, carrier and advanced Asterisk tracks offer detailed information for a variety of attendees. AstriCon will take place at the Renaissance Glendale Hotel and Spa near Phoenix, Arizona. Registration is now open at http://www.astricon.net.

Digium is working with Technology Marketing Corporation to promote the event to a broader group of people than have attended AstriCon in the past. TMC has helped support other Digium events, including Digium|Asterisk World, with training sessions, video production, attendee registration and exhibit management.

“Asterisk’s increased uptake by bigger companies and carriers spurred our decision to broaden the focus of AstriCon this year. We’re excited to offer a dedicated track for developers and to offer equally strong programs for other audiences,” said Mark Spencer, Digium’s founder and chief technology officer. “This year we received a number of outstanding proposals from speakers who have done great things in enterprise and government telephony environments, and we have a lot to offer attendees in that segment of the Asterisk market. Finally, we’re sure all of our audience members will be excited to hear from our keynote speakers, who we’ll be unveiling mid-month.”

Telefónica O2 Germany Selects Acme Packet to Deliver IMS-based Residential VoIP Services

VoIP Monitor — Tue, 07 Jul 2009 17:20:09 GMT

Telefónica O2 Germany has selected Acme Packet’s session border controllers to provide key IP Multimedia Subsystem functionality at its access and interconnect borders. The Acme Packet Net-Net Session Director enables Telefónica O2 Germany to deploy a secure and interoperable IMS network supporting residential retail and wholesale VoIP services and IP interconnects with other service providers.

Directly, and through its wholesale partners, Telefónica O2 Germany currently provides fixed telephony and DSL services to approximately 1.4 million customers. The Net-Net SBCs are part of an IMS next-generation network that will replace an older VoIP platform and enable Telefónica O2 Germany to offer VoIP services to its residential customers and wholesale hosted residential VoIP services—Telefónica Voice4Carrier and Telefónica Voice4SIPOwner—to other service providers. Telefónica O2 Germany also uses the Net-Net SBCs to securely interconnect with other tier 1 service providers for off-net call origination and termination.

Telefónica O2 Germany selected Acme Packet’s Net-Net 4500 SD to meet critical 3GPP IMS functional requirements at the access and interconnect borders of its network. At the subscriber access border, the Net-Net 4500 SD integrates the Proxy-Call Session Control Function, the initial SIP signaling contact point for subscriber endpoints, and the Access Gateway Function for controlling subscriber media flows. At the interconnect border to other service provider networks, the Net-Net 4500 SD integrates the Interconnect Border Control Function, the SIP signaling point for sessions leaving or coming into the network, and the Transition Gateway for controlling associated media flows. Both the access and interconnect SBCs include a Signaling Firewall, a vital function that protects the SBC itself and core IMS service elements, which is missing from the IMS architecture.

Net-SAFE, Acme Packet’s security framework, provides denial of service attack prevention, topology hiding and access control to protect Telefónica O2 Germany’s IMS network and ensure service availability. In addition, the Net-Net 4500 SD also provides multiple encryption capabilities, including IPsec using IMS-AKA and Transport Layer Security, for secured sessions between subscriber endpoints and Telefónica O2 Germany’s IMS services network. Acme Packet SBCs maximize the service reach of Telefónica O2 Germany using hosted NAT traversal, SIP signaling normalization as well as interworking between its next-generation IMS and older VoIP platforms. Telefónica O2 Germany uses Acme Packet’s flexible session admission control policies to prevent core network overloads and assure customer SLAs. To help ensure regulatory compliance, the Net-Net SD provides signaling and media replication for lawful intercept and a location server interface to obtain and append location information for emergency calls.

Report: VoIP Market in Poland 2009 - Development Forecasts for 2009-2012

VoIP Monitor — Wed, 01 Jul 2009 17:10:16 GMT

Research and Markets has announced the addition of the "VoIP Market in Poland 2009 - Development Forecasts for 2009-2012" report to their offering. VoIP market in Poland 2009 draws on a wide variety of trusted sources to create a comprehensive reference tool. Proprietary data and projections are added by skilled analysts with years of expertise in their fields. Readers benefit from the field experience provided via interviews with seasoned telecom industry professionals.

The market analysts at PMR have gathered all of the latest market intelligence and assembled it into this single clear, concise document - ready for delivery to you in convenient electronic format.

If you are interested in complete coverage of:

The current status of the VoIP telephony market in Poland, with the overview on market value, dynamics and trends, VoIP market structure, popularity of VoIP in Poland and legal environment
VoIP integration into the mobile telephony market as well as value added services that are popular in the world of VoIP telephony
Key players on the VoIP market, including: Internet communicators, Cable TV operators, Telecommunications, operators, and VoIP operators
Forecast values of market segments into the year 2012??

MyGlobalTalk Now Available on the iPhone

VoIP Monitor — Wed, 01 Jul 2009 17:06:54 GMT

i2Telecom announces its popular international calling solution MyGlobalTalk is now available as a free iPhone application on the iTunes online store. The MyGlobalTalk application for the iPhone enables US-based users to make international phone calls for less by routing calls through the MyGlobalTalk network. It works wherever mobile service is available, providing high-quality service without the need for a Wi-Fi connection. There are no contracts, connection fees or call minimums. It is strictly pay as you go.

Some of the key features the application provides are:

Easy-to-use interface allows users to dial as they normally would
Integration with native iPhone contacts list
“Faves” and “History” lists for calls made only through MyGlobalTalk
Easy access to account information and balance

Raketu Releases Mobile Videocalling With No Download

VoIP Monitor — Tue, 30 Jun 2009 13:58:22 GMT

Raketu announces the release of Raketu Mobile Video Calling. With Raketu's new mobile video calling, mobile users on WiFi or data services will now be able to make video calls completely free. Mobile users will also be able to make voice to voice calls, mobile to mobile or mobile to desktop, over WiFi or data, completely free. Mobile and desktop users can also take advantage of Raketu's low dialout calling rates at up to 65% lower than Skype's. Raketu provides the most flexibility of calling with computer to computer, computer to phone, and phone to phone calls. Raketu users can even make calls directly from any browser without a download. And it's available using Raketu's calling plans or at Raketu's ultra-low rates -- all from any device, anywhere in the world, mobile or desktop/laptop.

Digium Releases Hardware-based Voice Compression PCI Express Card for Asterisk Systems

VoIP Monitor — Tue, 30 Jun 2009 13:48:20 GMT

Digium releases the TCE400B PCI Express card for use with voice applications based on the open source Asterisk telephony platform. The new card provides hardware-based voice compression and decompression (codec) capabilities to shift transcoding from software to hardware. Using the TCE400B in place of a software-only solution places fewer demands on servers and frees up Asterisk to more efficiently process calls and to provide functionality for phone systems such as call recording, conference calling and interactive voice response.

Asterisk is the most widely used open source telephony engine and tool kit. By offering flexibility to access and alter the software code, Asterisk empowers developers and systems integrators to create VoIP communication solutions that match the specific needs of a business. Many businesses use Asterisk as a PBX to manage phone calls, but it’s also commonly used as a gateway between IP and PSTN networks, a telephony feature server and as the basis for call center applications.

In addition to its support for G.729a transcoding, the TCE400B gives Asterisk the ability to convert G.723.1 compressed audio into other formats, a capability not otherwise possible with Asterisk or software-only solutions. The card’s capabilities allow transcoding between simple, G.711 u-law and a-law, and complex, G.729a 8.0 kbit/s and G.723.1 5.3/6.3 kbit/s, codecs. When running in G.729a mode, the TCE400B can support 120 simultaneous transformations; in a mixed G.729a and G.723.1 mode, it supports 92 simultaneous transformations.

ManageEngine Upgrades VoIP Quality Monitoring Software

VoIP Monitor — Mon, 29 Jun 2009 19:04:07 GMT

ManageEngine announces an update to its VoIP quality monitoring solution, VQManager, focused on meeting the management requirements of VoIP service providers. The latest release, VQManager 6.2, greatly improves on the application's existing VoIP packet processing capability, making it possible to support larger VoIP environments, monitoring up to 1000 simultaneous VoIP calls. And because larger environments are apt to have multiple operating systems, the new VQManager is designed to perform equally well on different versions and flavors of Windows and Linux operating systems.

To assist growing VoIP service providers, ManageEngine is now also making VQManager available in a licensing model where pricing is based upon the number of simultaneous calls to be monitored. Service providers leasing VoIP trunks on an 'x' dollar per month per trunk basis can now affordably purchase a VoIP monitoring solution to manage present traffic volumes instead of having to acquire expensive, huge-capacity software so early in the business' growth phase. As their VoIP network grows, service providers can upgrade VQManager to higher concurrent call licenses, paying only the difference between the existing and the new number of licenses.

VQManager is a VoIP quality monitoring software that detects and analyzes SIP, H.323 & Skinny protocols along with RTP and RTCP media. The solution monitors each and every call in real-time, throughout the day, to provide the VoIP engineer a sound troubleshooting platform to diagnose and analyze VoIP performance degradations. VQManager's deep, packet-level analysis of a VoIP call allows a wide range of critical information to be reported, including: Protocol type, codec used, octets transferred, call initiation time, call set-up time, call ringing time, duration of call, call participants' names, URL and IP address, and call participant-specific QoS values viz. MOS, delay, jitter, packet loss and R Factor. Over 30 built-in reports, including concurrent call reports, carrying months of call data, help service providers make better capacity planning decisions.

Key feature additions found in VQManager 6.2 include:

Improved performance to support environments with up to 1000 simultaneous VoIP calls
An exclusive licensing and pricing model for VoIP Service Providers based on concurrent calls
Full support for Cisco CallManager 6.x and 7.1 environments
Support for processing Avaya S8700 CDRs, Cisco Unified Communications Manager 6.1 and 7.0 CDRs
Ability to create 'Schedulers' for predefined reports and e-mail them as PDF files.
Enhanced multi-day reporting to provide day-specific averages on ASR, ACD, MOS etc.

IPtimize Acquires National Telecom & Broadband Services

VoIP Monitor — Fri, 26 Jun 2009 19:31:56 GMT

IPtimize has acquired the assets and operating network of National Telecom & Broadband Services, a Florida corporation including Teliphonex, their business-class VoIP hosted PBX solution. Based in West Palm Beach, Florida, the commercial VoIP business assets acquired by IPtimize serve small business and enterprise customers in multiple metropolitan areas including West Palm Beach, Miami, Orlando and Tampa, Florida. The acquisition adds approximately 1,000 customer lines served by IPtimize and adds both expanded geographic reach and an established commercial customer base to the growing VoIP business at IPtimize.

IPtimize offers a complete suite of advanced commercial voice services, including its SIP Trunking and Hosted VoIP Services to client companies and their customers. IPtimize also offers provisioning and hosted communications packages, alleviating upfront customer costs by offering the solution on a hosted basis - reducing the need for capital expenditures and making high-end provisioning options available to smaller or budget-constrained customers.

IPtimize will be offering a complete suite of advanced commercial voice services to the acquired National Telecom customers. The transition is expected to be seamless for customers. Additional enhancements and features from IPtimize will be made available to customers in the future.

Additional details of the agreement are to be included as exhibits to the Company’s Form 8-K being filed with SEC.

Truphone Adds 11 New Nokia Devices

VoIP Monitor — Thu, 25 Jun 2009 19:08:06 GMT

Truphone has added 11 additional Nokia handsets to the list of devices compatible with Truphone and whose owners can now ring up big savings by taking advantage of Truphone’s broad range of calling plans and services. With the addition of the 11 new handsets, Truphone is now compatible on 26 Nokia devices. Of those, software for 14 of the Truphone-compatible devices can now be downloaded from Nokia’s new Ovi store, with the software for 11 of the remaining 12 new devices to be added to the Ovi store soon.

In addition to the Ovi Store the Truphone application is also available for all 26 Nokia devices as an SMS over the air download from the Truphone web site and will always parallel the applications available via the Ovi store. The Truphone Web site will also include applications for other Nokia devices not currently available via Ovi, as well as the other supported mobile platforms.

The new Truphone-enabled Nokia devices are:

N96
N78
N85 (also VoIP enabled)
N79 (also VoIP enabled)
5630 (also VoIP enabled)
5800
5320
6210
6220
6650
E63

After downloading and installing the Truphone application to their handsets, Nokia device owners will be able to take advantage of two calling plan options:

Truphone Anywhere: Calls made when customers are connected to local wireless networks are handled as local calls and then passed on to the Truphone network, from where they are routed at Truphone’s fantastic rates to landline and mobile phones worldwide. This service is intended for use in the user’s home country as savings cannot be guaranteed abroad.
Truphone Wi-Fi calling: Calls made when customers are in accessible Wi-Fi hotspots anywhere in the world, can be made for free to other Truphone users and at Truphone’s fantastic rates to landline and mobile phones worldwide.

All of the new handsets are Truphone Anywhere capable, and in addition, three of the handsets, the N85, N79 and 5630, are also compatible with the original Truphone Wi-Fi calling service, hailing back to Truphone’s origin as the first Mobile VoIP service available on Nokia N- and E-series devices.

This means that when customers are connected to a Wi-Fi network, a call to another Truphone customer is completely free and also means they can use a Nokia handset as an alternative, inexpensive roaming solution if they choose to only make calls when connected to a Wi-Fi network when abroad or anywhere.

With Truphone Anywhere, everyone can save money when making international calls. It’s simple: International calls dialled on a Truphone Anywhere enabled handset are connected as local calls to the Truphone network, then routed to their destinations at Truphone’s low rates. This means that for the price of local calls, which are often bundled in an operator’s monthly package, customers can make calls to other Truphone users for free or to landlines and mobiles around the world for as little as 1.5p per minute.

DynaSis Partners with Fonality to Deliver Affordable Atlanta VoIP Solutions

VoIP Monitor — Thu, 25 Jun 2009 19:01:25 GMT

DynaSis announces its new partnership with IP phone system provider Fonality to offer small to medium sized businesses access to affordable Atlanta VoIP solutions comparable to large business phone systems. The barrier to entry for VoIP in Atlanta for most small to midsized businesses has long been the cost involved with installing and maintaining a full scale VoIP solution that can provide the sound quality and features enterprise solutions offer. Enterprise solutions are often costly to maintain because they require a skilled IT professional for minor issues such as simple additions or deletions of extensions, and most of the cost savings realized from long distance savings are eaten up by maintenance and initial installation costs.

Fonality from DynaSis creates affordable SMB Atlanta voice over IP solutions that are easy to use, easy to manage, scalable, feature-rich, and reliable. Through this partnership, DynaSis is now able to offer small to mid-sized businesses the ability to sound and operate like a large business.

With Atlanta VoIP solutions from DynaSis advanced features help even the smallest businesses sound like multi-million dollar operations. Big business features such as auto-attendant, dial-by-name directory, call queues, call parking, drag-and-drop call transfer, CRM integration, and advanced reporting are finally available at affordable prices.

JAJAH Connects One Billionth Call

VoIP Monitor — Tue, 23 Jun 2009 18:36:01 GMT

JAJAH has connected one billion calls since it launched its platform. The company, which began as a web-activated-telephony service, has evolved to become the world's leading communications platform, partnering with telecommunications and technology companies such as Intel, Microsoft and Yahoo!

The billionth call

The historic billionth call on the JAJAH platform was made by Ila Vasudev Zeff, a 29-year-old resident of San Francisco, CA. Mrs Zeff made the call on Yahoo! Voice. She was calling her mother in India, who she calls at least weekly since moving to the United States from India.

JAJAH has been recognized for its leadership in transforming global communications. Recently named a TiE50 winner, JAJAH has also won the Alexander Graham Bell Award for 'Best Communication Solution,' as well the Always On and the Global Red Herring award.

The company has also achieved unprecedented business success. Its innovative platform has been deployed by dozens of companies across the globe, making it possible for these companies to roll out new, proven communications services to their customers.

Voxbone Selected by Nimbuzz to Provide Mobile VoIP Function Without Wi-Fi

VoIP Monitor — Tue, 23 Jun 2009 18:15:08 GMT

Voxbone has been chosen by Nimbuzz to provide local access numbers. Nimbuzz is using Voxbone to enable its users to make voice calls – without benefit of 3G or Wi-Fi connectivity – to their contacts on any of all popular IM and VoIP services, including Gizmo5 and Skype.

Nimbuzz, running on all Internet-capable mobile phones, detects when the handset is out of Wi-Fi or 3G range and conveniently steps in, requesting permission to automatically dial a local access number and route the call over the Internet. Such calls are free except for the low charge (if any) to the local access number. The Nimbuzz software client determines the correct access number to dial from the user’s Nimbuzz profile.

Currently growing at a rate of over 750,000 sign-ups per month and operating in 200 countries, Nimbuzz combines the services of mobile VoIP, chat, location, file sharing and MMS under one application. Equally important, it works across popular communities and social networks, including Skype, Windows Live Messenger, Yahoo! Messenger, ICQ, GoogleTalk, AIM, Facebook and MySpace. Nimbuzz VoIP also reaches PSTN phones through Skype-Out, using any of 10 VoIP third-party VoIP services (including Skype). A full phone keypad is part of its interface, along with the multi-service buddy list.

FreedomVOICE Connects with Customers Launching New Blog for FreedomIQ Hosted VoIP PBX

VoIP Monitor — Tue, 23 Jun 2009 18:06:42 GMT

FreedomVOICE announces the launch of the brand new FreedomIQ Hosted VoIP Blog. The blog will act as a resource for the VoIP industry. FreedomVOICE feels that the FreedomIQ Hosted VoIP PBX is the perfect product to be represented by a blog. Blogs are a tool for free expression, and this is something the company values as an integral part of their corporate culture.

Proving to be very knowledgeable in their online efforts, FreedomVOICE has also connected with their customers though LinkedIn, Twitter, and Facebook.

In an effort to make the blog as interactive as possible, FreedomVOICE is allowing guest bloggers to supply content for the site. Guest bloggers should be familiar with the FreedomIQ product and have sufficient experience in the VoIP industry. If you are interested in becoming a guest blogger, contact the media contact listed at the bottom of the page.

XConnect Wins Global Telecoms Business Innovation Award for Korean NGN/VoIP-Peering Federation

VoIP Monitor — Mon, 22 Jun 2009 18:28:53 GMT

XConnect has received a 2009 Innovation Award from Global Telecoms Business magazine. The award was presented to XConnect and the Korea Internet Neutral eXchange in the IP Services Innovation category, for creating a national Korean NGN/VoIP-peering federation, which was announced in December and began operating in March.

The Global Telecoms Business Innovation Awards are designed to honor innovative projects involving telecoms operators and service providers around the world, in association with their vendors and suppliers.

The 2009 awards were presented at a dinner ceremony June 16 in London, attended by about 350 industry leaders from around the world.

IPEVO Equips USO Pacific Region with Skype Phones

VoIP Monitor — Mon, 22 Jun 2009 18:08:14 GMT

IPEVO announces that 18 USO centers across the Pacific Region will deploy its S0-10 Skype Desktop Phone and S0-20 Wi-Fi Phone for Skype devices. Using Skype and the IPEVO phones, more than 890,000 troops and their families that use the centers each year will now have a free communications link back home, without the need for a computer to make calls.

The S0-10 is a fully certified desktop Skype phone for the home and office. Connecting directly through an existing network, the S0-10 eliminates the need to use a computer for IP calls. The S0-20 is a Wi-Fi phone that gives users the flexibility and mobility to use Skype wirelessly in a home or office Wi-Fi network. People traditionally think that you need a computer to use Skype, but with IPEVO's stand-alone phones, calls can be made using a Skype account directly on the device.