Source: www.voip-news.com
Love this! VoIP and Gadgets Blog traces rudimentary computer technology that the writer used as a kid to the VoIP service that he now bases his career writing about.
The VoIP Weblog explores Sipgate in its competition with Skype.
Truphone 3.0 has launched for the iPod Touch. Read about it on VoIP News of the UK.
Source: www.voip-news.com
A new phone service from sipgate claims to be a landline replacement that has all the functions of a phone service including telephone number but it carries no cost for the set-up, the number itself or any monthly charges or minimums. Sipgate one works with any phone (including standard analog ones, mobile phones and VoIP phones). It operates over the internet.
Calls cost 1.9 cents per minute, but calls to other sipgate users and received calls are free as well, giving it an advantage over Skype where you pay for incoming calls.
“There is simply no barrier to people disconnecting their old phone lines anymore. Phone and cable companies have long been pushing voice plans in the region of $25 to $40 per month–which end up being as much as $60 or more with extra charges–and that’s just ridiculous,” said Thilo Salmon, CEO of sipgate. “Even with calls to other landlines and mobile phones, most users will spend less than $5 a month using sipgate one. And for those people only receiving incoming calls on their VoIP phone, the service is completely free.”
I have to disagree with Salmon on there being no barrier to disconnecting old phone lines — if the internet goes down or the power goes out, you will lose your phone service, which is a problem for some people (like parents who want the security of having an analog backup phone with a traditional line so that you can phone anytime in an emergency). Nonetheless, as far as VoIP goes, this sounds like a good deal.
Here’s the benefits:
At no additional cost, sipgate offers a rich feature set designed to make users’ lives easier and more convenient:
– Need to write down a number while driving? Record any phone
conversation by pressing *6
– Want to customize your voicemail? Upload an mp3
– Want your calls to follow you? Have your home, office and mobile
phone ring in parallel
– Left your mobile phone at home? Simply go online and divert calls to
your office line
– Want to talk to friends in an ad-hoc conference? Bridge them with the
touch of a button
– Missed calls while on a flight? A list is waiting for you in your
Source: www.voip-news.com
Technology Marketing Corporation has award Radware’s SIP Director the 2008 Communications Product of the Year Award. SIP Director is a fully SIP-aware application delivery controller.
“We are pleased to be the recipient of this distinguished award from Technology Media Corporation,” said David Aviv, Vice President, Advanced Services, Radware. “At Radware, we strive to offer the finest solutions and provide the most dedicated customer support team, and we believe this recognition affirms our passion to deliver business smart networking to carriers. By addressing growing demand for SIP service delivery by both carriers and operators, SIP Director provides the high availability, scalability, resilience and interoperability that our global customers require in order to address their growing network requirements.”
More information about the winners of the award can be found at www.tmcnet.com/voip.
“SIP Director has been recognized with a 2008 Product of the Year Award for their excellence in the advancement of voice, data and video communications,” said Rich Tehrani, TMC President and Group Editor-in-Chief, “Radware has proven they are committed to quality and excellence in solutions that benefit the customer experience as well as ROI for the companies that use them. I am pleased to honor their hard work and accomplishments and look forward to more innovative solutions from them in the future.”
Source: www.voip-news.com
Dialogic 2000 Media Gateway Series is now certified as interoperable with Ingate SIParator and Ingate Firewall products, according to Dialogic Corporation and Ingate Systems. The media gateway went through testing to validate the interoperability.
With customers at the SMB and enterprise level clammoring to upgrade to VoIP services to cut operating expenses, SIP trunking has really gained in popularity.
“To date, most SIP trunking providers and resellers have focused on the opportunity to migrate SMB class customers from PSTN service in concert with an upgrade to a SIP-ready IP-PBX system. While this opportunity focus remains logical and compelling, there is also a significant opportunity to migrate Enterprise class customers who maintain legacy (non-SIP) PBX and contact center systems” stated Steve Johnson, President of Ingate Systems. “The combined security features of the Ingate SIParator with the SIP to PBX trunk conversion capability of a DMG2000 gateway enable a cost effective, secure and reliable SIP trunk interface for these customers.”
According to the companies:
SIP trunks are delivered to business customers over broadband IP data networks. With this interoperability solution, both Ingate and Dialogic products are deployed on the customer premises to support the SIP trunking service. The Ingate products are deployed at the network edge between a wide area IP network and the corporate LAN, securely passing SIP signaling and VoIP media streams to and from the corporate LAN. The Dialogic gateway resides on the corporate LAN and is connected to the legacy PBX or contact center via traditional T1/E1 trunk ports. The DMG2000 gateway passes the SIP Trunk signaling and media from the Ingate SIParator to the PBX by emulating traditional PSTN trunk services.
“We are very pleased to announce this interoperability with Ingate today,” said Franz-Josef Eberle, General Manager, Dialogic Enterprise Gateway Product Group. “Ingate is a recognized leader in securing SIP trunking for business customers, which comprise a critical piece that can be overlooked by some device vendors. We believe this solution set truly opens up a whole new Enterprise market opportunity for SIP Trunking providers and their resellers.”
Source: www.voip-news.com
With workers being spread out more far and wide, it can be a challenge to keep everyone connected without spending gobs of money. After all, the key to being a successful, spread-out organization is ensuring accountability through regular communication and check points.
That’s why Sipera Systems’s new workforce communications solution is so interesting. It combines secure VoIP, video messaging, collaboration and other unified communication features to connect workforces that are spread out geographically. The Sipera Business Continuity Communications Solution (SBCCS) is a solution that can connect to any worker in anywhere with IP network connectivity, using full encryption, access control, policy enforcement and other security.
“Disruptive events like floods, pandemics, and workplace security breaches may be relatively rare, but in a global market, the odds increase that businesses large and small will eventually be adversely affected, if only for a brief time. Over the past 24 months, we’ve been working with several major banks, hospitals and school campuses to plan ahead for potential disruptions, primarily by enabling remote and home office communications,” said Andy Asava, Vice President of Worldwide Sales at Sipera. “Our UC-Sec solutions enable and secure complete UC services at employee’s homes and remote offices, without requiring anything more than existing office VoIP phones and laptop PCs. This enables ongoing business operations outside enterprise headquarters, to remote workers wherever they might be and whenever circumstances dictate staff should not commute to their usual workplaces.”
Here’s some detes of this solution:
Simplicity and security are the two highlights of Sipera’s teleworking solutions, which require no configuration changes on IP-PBX or unified communication servers when employees are directed to work from home or other offices. UC-Sec appliances provide threat protection from Internet attacks through deep packet inspection, stateful communications proxying and management, and full application awareness. Privacy of all communications is ensured with TLS/SRTP standards-based encryption, for signaling, media and other services. Sipera customers are ensured of granular control of enterprise security policies for teleworkers, with flexible unified communication policies. In addition, Sipera provides advanced access control security features, including 2-factor authentication for especially security conscious businesses.
The Sipera UC-Sec products are comprehensive, real-time UC security solutions that simplify and protect next-generation communications over any network to any device. Dozens of customers around the world rely on the UC-Sec appliances to safeguard their communications and boost compliance with privacy laws, regulatory mandates and mission-critical confidentiality requirements. Backed by the extensive vulnerability research of VIPER Lab, the UC-Sec solutions provide threat protection, policy enforcement, access control and privacy in a single, real-time appliance.
Source: www.voip-news.com
Is mobile voice traffic going to transition to mobile VoIP? According to a recent prediction, 50 percent of it will be mobile VoIP by 2019. Read about it on VoIP & Gadgets Blog.
Fring has secured an additional $10 million in funding. Read about it on VoIP News of the UK.
SIP to Skype calling? VoIP Watch says it’s just a matter of when.
Source: www.voip-news.com
The first-ever TSG-6 SIP-compliant phone has been released by Tone Commander and it’s already approved by the National Telecommunications Security Working Group (NTSWG) for government and military use. The US-made Tone Commander 7810 TSG-6 IP phone can be integrated wherever IP networks are available.
“We are proud to offer this dynamic, cutting-edge new phone,” said Steve Hill, president of Tone Commander. “This new device provides our customers with the best of both worlds – best-in-class quality and top-of-the-line functionality with excellent flexibility. And with NTSWG certification, the phone can now be deployed for sophisticated government and military applications.”
So what are the perks of this phone?
Among its key features and benefits, Tone Commander’s 7810 TSG-6 IP phone offers:
– RTP voice encryption (AES-128) and TLS signaling encryption included
in FIPS 140-2 approved algorithms.
– Unique DoD features specified in DISN RTS GSR (Defense Information
Systems Network, Real Time Services, Generic Systems Requirements).
– Multi-level precedence and preemption, providing barge-in capabilities
for high-priority calls.
Source: www.voip-news.com
snom has released a SIP conference phone with four-way calling, conference recording functionality and Konftel OmniSound full duplex sound technology. The snom MeetingPoint VoIP phone is compatibile with IP PBX, hosted VoIP and UC solutions.
“snom MeetingPoint marks the next step in the evolution of snom phones as we continue to broaden our product offerings to meet the needs of today’s global business market,” said Michael Knieling, snom’s Executive Vice President of Marketing. “It leverages Konftel’s patented OmniSound® 2.0 broadband technology enabling us to deliver full duplex audio that is almost concert quality without any distracting background noise.“
According to snom:
snom technology AG, developer and manufacturer of Voice-over-IP (VoIP) telephones, today introduced the snom MeetingPoint a SIP-based conference phone for the North American enterprise and small and medium-sized business (SMB) 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.
Source: www.voip-news.com
With the release of the beta version of Skype for SIP this week, there is considerable interest — and there should be. The new money-saving service could make VoIP communications an even more desirable avenue for businesses to pursue.
But what does the release mean for the SMB market?
VoIP-News tackles that question in this feature. Here’s a snippet:
Though it sounds like a good deal, the new service is hardly a breakthrough for SMBs. SIP-trunking providers already offer many of the same benefits. For example, they provide cheap inbound and outbound calling and virtual inbound numbers (like SkypeIn numbers) in different cities and even countries. They also offer capabilities Skype doesn’t, such as emergency E911 calling.
On the other hand, SIP-trunking operators don’t provide free inbound calls from Skype users. Businesses that find that feature important will surely want the new service. Similarly, some companies may have been considering buying a Skype gateway, one of the boxes or software applications that connect an IP PBX to Skype’s service. The new service would make such a purchase unnecessary.
Source: voip-tech.blogspot.com
Skype™ launch its beta version of SIP (Session Initiation Protocol), to allow companies and offices that own a PBX, a telephone exchange like Asterisk, to use the cheap rates of Skype™ to call telephones all over the world, obviously the Skype-To-Skype calls, by the SIP protocol, are free.
In a near future it will be possible to knock down almost completely the costs of normal phone calls, when most companies, offices and business activity will be equipped with easy and practical systems of voice communication by Internet, without the need to use a phone provider, if not for the broadband connection.
A defect is that the dispatch of normal fax is not still supported by the SIP protocol and it still need the use of a normale telephone line, even if soon, supposedly, the fax will be substituted by the Certified E-mail.
Source: www.voip-news.com
SmartSIP has launched, VoIP and Gadget Blog reports.
Jaxtr and TynTec are in an SMS deal, reports the other VoIP News.
eComm went off swimmingly, reports VoIP Watch.
Source: www.voip-news.com
Sonus Networks has received SIPconnect 1.0 complaint status from the SIP Forum for its Sonus Network Border Switch. The carrier class border control solution allows for centralized routing, media transcoding and enhanced security.
“The dramatic growth in communication choices provides enterprises with new options to enhance their workforce’s productivity,” said Marc Robins, managing director of the SIP Forum. “Interoperability is the single biggest hurdle to widespread adoption of new IP communications products and services. The SIP Forum established the SIPconnect framework to make it easier for enterprises to bridge IP from the service providers’ core to the end users desktop, enhancing quality and enabling new service creation.”
According to Sonus:
With Sonus IP Business Trunks, service providers can create a single broadband connection for voice and data, providing a more cost effective and efficient connection. This connection can help to drive new revenue through innovative services. Because the Sonus IP Business Trunk is a direct IP-to-IP pipeline between the core network and enterprise-based IP-PBX, service providers can quickly deliver hosted, secure services to enterprise customers without the need for additional Class 5 infrastructure.
The SIPconnect Technical Recommendation, an initiative of the SIP Forum, is a standards-based approach that provides detailed guidelines for direct IP peering and interoperability between IP PBXs and VoIP service provider networks. The associated SIPconnect Compliant Certification Program, through which eligible companies can gain SIPconnect validation and the right to license the use of the SIP Forum’s ‘SIPconnect Compliant’ certification mark, helps to validate interoperability among equipment manufacturers, software providers and service providers, as well as to promote further adoption of SIP as the standard for IP communications.
“The Sonus Network Border Switch has already proved its value in the market, supporting thousands of IP business trunks,” commented Shailin Sehgal, vice president marketing and product management at Sonus Networks. “The recognition of compliance with SIPconnect further demonstrates that Sonus is delivering an enhanced, standards-based solution for access border security. Sonus can offer service providers a path to transform their networks to all IP, from the core to the edge.”
