Fonality Becomes A Gold Sponsor Of ITEXPO WEST 2011

ITEXPO WEST 2011 http://snapvoip.blogspot.com/
TMC announced today that Fonality has become the latest Gold sponsoer of ITEXPO WEST 2011 to be held in Austin, TX at the Austin Convention Center, Septermber 13 to 15th. ITEXPO is the conference and trade show is the best attended Communications and VoIP IP Telephony trade show, be it either west or east.

NORWALK, Conn.--(BUSINESS WIRE)--TMC today announced that Fonality, North America’s fastest growing business communications company, has signed on to become a platinum sponsor of their ITEXPO conference and trade show, September 13-15, at the Austin Convention Center in Austin, Texas.
ITEXPO is the world’s largest and best-attended communications and technology trade show. ITEXPO West is currently ranked as the No. 3 fastest-growing event on Trade Show Executive Magazine’s Fastest 50 List. TMC expects to accelerate ITEXPO West’s growth by bringing the show to Austin, one of the fast-growing technology hubs in the U.S.

“ITEXPO is the industry’s leading venue for cloud-based and open source telephony solutions for enterprises and SMBs,” said Fonality CTO Rick Bushell. “We’re looking forward to interacting with the thousands of attendees who come to ITEXPO to show how Fonality’s solutions can help small- to mid-size businesses communicate smarter, faster and more affordably.”

Designed for small and mid-size companies, Fonality offers VoIP, Unified Communications (UC) and contact center solutions that are easy to use, simple to manage, and affordable to deploy. Fonality’s cloud-based model provides advanced features and services without the costly hardware, infrastructure or lengthy implementation cycles offered by legacy providers.

Total cost of ownership is dramatically reduced by 50 percent, or more, while users enjoy access to powerful communications capabilities typically associated with Fortune 500 firms. Fonality’s Head’s Up Display (HUD) UC dashboard offers presence based capabilities to seamlessly help manage voice, e-mail and chat dialogue and ensure users can reach the right person, with the right information, every time.

“Fonality serves a large and rapidly-growing community of users, who come to ITEXPO to learn more about the latest advancements in the industry,” said Rich Tehrani, CEO and conference chairman for TMC. “Fonality has made significant contributions to ITEXPO over the years, and we’re looking forward to working together once again at ITEXPO in Austin.”
Registration for ITEXPO is now open.

Sponsorship and advertising opportunities are also available, contact TMC’s Joe Fabiano (203-852-6800 x132) for more information. Members of the media can reserve press credentials by contacting Todd Keefe at For Immediate Release PR.

sipgate CEO Explains Why They lected Yate Open-Source VoIP Telephony Engine At VoIP User Conference.

http://snapvoip.blogspot.com/
sipgate CEO Thilo Salmon returns to Voip User Conference (Live, weekly on every Friday 12:00 Noon Eastern Time) this week with Diana Cionoiu, to speak about sipgates use of Yate, an open source VoIP IP Telephony engine from Null Team, Romania.

Following is the press release by SIPGate regarding the YATE.

05/04/2011 - sipgate has chosen Yate, the open source project of NullTeam, to power the core of it's soft-switch infrastructure.

The open-source telephony engine Yate is making inroads into what used to be the domain of proprietary equipment provided by large telecom suppliers. Following the certification of Yate SS7 stack by Deutsche Telekom's test lab, sipgate has started to deploy Yate at the core of it's softswitch. The decision came following extensive tests to determine stability, scalability as well as resilience.

"The open source telephony platform was easily integrated in the infrastructure of our softswitch", said Thilo Salmon, CEO of sipgate. "We are mostly impressed by the quality and scalability of Yate technology."

"We are pleased that sipgate has chosen Yate for implementing their telephony softswitch", said Diana Cionoiu, managing director of NullTeam. "This is a solid recognition of our 7-years development of Yate platform"

About sipgate:
With offices in Austin, Texas and Dusseldorf, Germany, sipgate is a well-established and recognized international telecom company. sipgate has been successfully building its business since 2004. Today, it is arguably the most popular network-independent phone service in a number of European countries catering to a six figure user base.

About Null-Team:
Null Team, based in Romania, is a software company started in 2004, whose founders have years of experience in telephony and software industries. Yate, primary product of the company is a Telephony Engine, usable in a variaty of customized applications. Null Team also sell solutions for call centers, telephony carriers, for enterprise and soho market.

SIPGate, and  YATE via VUC

Low Bandwidth Skype For Mobile Markets With Low 3G Penetration.

http://snapvoip.blogspot.com/
This week, Skype announced a low-bandwidth version Skype, for mobile markets in parts of the world where 3G service is not available or sparse. Skype describes it's mobile solution as a simple carrier solution utilizing purpose built client-server solution. The optimization of software for low bandwidth is certainly a plus for VoIP IP Telephony provider.

"We see a growing demand for differentiated services from mobile operators across the world , Our mobile operator program will give operators in emerging markets the possibility to tap into the global Skype community while providing a great user experience on a variety of mobile handsets." said Russ Shaw, vice president and general manager of Mobile at Skype.

Even though we complain at slightest dip in our bandwidths, be it mobile or land line, there are many parts in the world where getting a service is the issues rather than the bandwidth. With wired lines being costly to deploy in far interior villages, like in certain parts of Asia and Africa, mobile networks are pushing forward in providing much needed communication services.

LUXEMBOURG, 14 February 2011 – Skype today announced a Mobile Partner Program for operators in markets with low 3G broadband penetration. The program aims to bring the Skype experience to a broad range of mobile users and will enable carriers to differentiate their offerings in a competitive market environment. This program follows the successful operator partnerships that Skype has already launched with Verizon, KDDI and Hutchison 3.
The Mobile Partner Program from Skype is a simplified operator program that utilises a purpose built client/server solution. The solution developed specifically for operators, enables use of Skype on a broad range of handsets including smartphones and feature phones. Optimised for efficient bandwidth usage, it keeps users connected with minimal impact on battery life and data usage.
Operators taking part in this program will be able to offer their customers a fast, easy Skype experience that runs on a range of OS platforms covering more than 100 different smartphone and feature phone handsets. Through Skype their customers will have the ability to make free calls and send instant messages to other Skype users. In addition, their customers can take advantage of Skype’s low rates to make affordable calls to mobile phones and landlines virtually anywhere in the world.

“We see a growing demand for differentiated services from mobile operators across the world,” said Russ Shaw, VP and GM of Mobile at Skype. “Our mobile operator program will give operators in emerging markets the possibility to tap into the global Skype community while providing a great user experience on a variety of mobile handsets.”
Operators interested in this program can contact Skype’s mobile business development team at: mpp@skype.net.

Skype

Chromium Project Drops H:264 From Chrome!

#Chromium Project Drops #H:264 From Chrome! http://snapvoip.blogspot.com/
Looks like Google dropped a bomb in the websphere, when it announced that it is dropping H:264 support from Chrome. There were shouts of being evil, and threats of dropping Chrome all over the place, there is even a h:264 / MPEG supporter slashdot talking about how bad Google is. Well I guess he forgot how many patents (about 1200)are in h:264 and How evil they were before WebM came by.! 
I usually do not much like Apple's tactics but loved it when it dropped Adobe Flash, the worst spyware, (I shunned Adobe reader since 2006 when Adobe Reader started snooping into users affairs.for me, on the web. For Adobe flash' bad behavior flash cookies sharing personal data,  which I have restricted, (using the method given in the link) as much as I could. I will not trust Adobe with a ten meter yard stick. Even though I bashed Steve Jobs for it I secretively applauded Apple when it stopped using Flash. Did not matter how many sites ware using flash that chokes even fast computers, at times. But I also frowned when Google incorporated Flash in to Chrome. I am yet to find about Flash Supercookies on Chrome and how to block them as well. (I need to create a link to that MyADLETS article on VoIP IP Telephony)

So how evil Google may be underneath, I applaud it's decision to drop h:264 support. Because it really does not matter, as someone will surely develop a H:264 plugin and we get to know about some codecs that are following open standards. Perhaps h:264 itself will become more open, like it did when WebM emerged! So we welcome Google's Announcement; for providing us more options, in the video arena.

To that end, we are changing Chrome’s HTML5 support to make it consistent with the codecs already supported by the open Chromium project. Specifically, we are supporting the WebM (VP8) and Theora video codecs, and will consider adding support for other high-quality open codecs in the future. Though H.264 plays an important role in video, as our goal is to enable open innovation, support for the codec will be removed and our resources directed towards completely open codec technologies.

I have to thank Pin drop soup for the info but I will say "Google Gets it!"

snom technology AG Moves US Headquarters To Woburn, Mass.

snom technology AG Moves To Woburn, Mass http://snapvoip.blogspot.com/
snom technology AG, has moved its North American headquarters to Woburn, Massachusetts to accommodate the company's growth in USA and worldwide. The new facility will become home to 40 percent of snom's worldwide employees, including sales, support, marketing and product development functions in North and South America.

snom technology AG  is known as a maker of some of the best IP desktop phones, and a vast collection of VoIP IP telephony products. snom's capability to integrate with leading unified communications  platforms and all SIP based VoIP services like IPPBXs to cloud and hosted VoIP environments, has made the company more than a IP phone manufacturer, but a developer of highly-functional enterprise endpoints for IP communications. Such growth require better and bigger facilities!

"With the popularity of our products growing in North and South America, we needed more space to support our sales and operations in the region, but we wanted to stay in the same area as our previous office, this move is symbolic of our recent successes and marks a significant moment in the growth of snom, not just in the Americas, but as an international player in the IP telephony market." said Dr. Christian Stredicke, founder and CEO of snom technology. "

Flash-to-SIP From Zingaya Offers Versatile Click-To-Call Service

Flash-to-SIP From Zingaya http://snapvoip.blogspot.com/
Zingaya,a Russian VoIP IP Telephony company, is entering the US market,introducing Click to call for web sites via Flash based technology, with a introduction at DEMO conference.
Zingaya enables telephone conversations through any computer via an widget installed on a webpage. Users do not have to download any client software to their computers. They just click on Click-to-call button and the call could be forwarded to landlines, mobile phones, Skype, PBX, iPBX or even a another computer with VoIP end point.
This is all possible due to Zingaya's Flash-To-SIP technology, front end and the backend servers..

"Through the growth of Skype and now Google voice, consumers are becoming increasingly accustomed to making phone calls from their computer, We have created the first enterprise-grade, highly scalable, secure, cross platform click-to-talk solution that offers the best sound quality available today." said Alexey Aylarov, project manager and co-founder at Zingaya.

Next to their pricing plan, following seem to be the keypoints offered by Zingaya;

• No download required: End users don't download software. They just click the "Call" button and are automatically making a phone call through their computer using Flash.
• Best-in-class sound quality: Sound quality is paramount with VoIP services, and Zingaya's software supports all new and proven technologies. This includes TCP and UDP protocols, Acoustic Echo Cancellation, Adaptive Jitter Buffer, Noise Reduction, Automatic Gain Control, and more.
• Scalable: Zingaya has developed state-of-the-art software that provides optimized performance for the service's main tasks, such as handling signalization protocols and transcoding audio streams.
• Cross platform: Widgets can be used on Windows, Mac and Linux systems.
• Extensible: Zingaya will soon give web developers API access for sites that want to make the click-to-talk technology part of their service.

Zingaya 

Voxbone iNum Numbers Are Turned Into Single-Click Local Calls With Rebtel iPhone App

Rebtel iPhone App http://snapvoip.blogspot.com/
Rebtel iPhone App is bringing one-click dialing to Voxbone iNum numbers by allowing users to generate a local phone number that will map to an iNum for for all contacts in the address book on their iPhones.  This feature allows users to call iNums without having to enter access number or PIN to remember. Just tap on the contact icon to reach users from over 50 countries for the cost of a local call. Voxbone have been active in the VoIP IP Telephony scene for a while now and it is good to see that Mobile VoIP is no exception.


Press release

BRUSSELS, May 25 /PRNewswire/ -- Voxbone, a leading provider of telephone numbers to communications service providers and businesses worldwide, today announced that Rebtel's new iPhone application brings one-click dialing to its iNum® numbers by allowing users to generate a local phone number that will map to an iNum for any contact in their address book directly from their handset. As a result, people will be able to call iNums – with no access number or PIN to remember – by clicking on a contact icon from over 50 countries for the cost of a local call.

Through its partnership with Voxbone, Rebtel facilitates low-cost international calling by providing users with a local phone number in the country where they live which they can use to connect them with friends, family or work colleagues who live abroad. Rebtel delivers the more expensive international leg of the call over the internet using VoIP, at significantly lower rates than those typically charged by operators. Rebtel's newly launched iPhone application offers a simpler alternative to Rebtel's traditional setup mechanism, in which customers generate local number for their contacts via the Web and manually load them into their phones.
iNum is a different type of global phone number that is not tied to a particular location. Taking advantage of the +883 country code allocated by the ITU, iNums are allocated for free to Internet telephony providers that enable IP communications services to be accessed via traditional fixed or mobile networks. Many VoIP services still use private alphanumeric identities like those for IM or e-mail. iNum provides an ITU-approved e.164 number that is expressly intended for global use with VoIP applications, and enables users to make and receive free or low-cost mobile VoIP calls within participating networks, such as those of Rebtel, Truphone and Gizmo5.

"Here at Rebtel, we immediately bought into Voxbone's concept of an 'area code for Earth,' as it clearly matches our own vision of making all phone calls local, turning the world into a friendlier place and bringing people closer together," commented Andreas Bernstrom, CEO of Rebtel. "We're proud to be an active member of the iNum community and to be providing a hassle-free way of dialing an iNum for the cost of a local call."

Rod Ullens, CEO and co-founder of Voxbone, added:

"Voxbone's goal is to further extend and grow the community of providers who deliver calls to iNums for free or for the cost of a local call. We're very pleased to see Rebtel making iNums reachable from over 50 countries and are delighted to have them on board."

To see a video tutorial of how the Rebtel service works, go to http://www.rebtel.com/en/Learn/

Skype Manager, A Multi-functional Business Tool For Skype Business Users.

Skype Manager http://snapvoip.blogspot.com/

Skype just announced a new tool, Skype Manager. Skype Manager is a multi-functional business tool that lets you centrally manage Skype across your company. You can add members, allocate Skype Credit, assign features, and monitor usage and allocations. The new tool replaces Skype Business Control Panel (BCP). Among other things, it has a new Dashboard feature.
If you are using Skype for SIP (Which is how we connect our VoIP IP Telephony network to Skype Network)that too could be managed via the new Skype Manager. Using the product is pretty simple that the Skype team was able to present you a quick start instructions with graphics in a three page PDF.
You can find more information about Skype Manager Here

“Beyond SIP Trunking – Unify the Enterprise.” Webinar By XO Communications, TMCNet and BroadSoft

Beyond SIP Trunking – Unify the Enterprise http://snapvoip.blogspot.com/
A dynamic trio in the VoIP IP Telephony and collaboration fields have come together to provide one of a kind webinar to everyone who is involved in communications space. One will be able to learn about latest industry insights and current trends in SIP Trunking and how to utilize these knowledge in their managed voice endevours. Even those looking for SIP accounts or SIP solutions will benefit by attending the Webinar.Beyond SIP Trunking – Unify the Enterprise Webinar will be held on May 12th at 2:00 p.m. EST
Mike Toplisek, chief marketing officer for XO Business Services at XO Communications has this to say;

“With today’s economic climate, it’s more important than ever before that enterprises evaluate their managed voice options to ensure they’re allocating their IT budget in an efficient and strategic manner for their business, By bringing together industry-leading experts on SIP trunking services, we hope to empower enterprises with the tools and information necessary to transform their distributed voice network architecture into a more centralized and cost-effective VoIP solution.” 

Beyond SIP Trunking – Unify the Enterprise Webinar Details:
When: May 12th at 2:00 p.m. EST
Who Should Attend: Both technical and non-technical audiences from U.S.-based businesses
Registrations: Complimentary 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 (WAN) 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.

Click here to register for the webinar.
For more information about XO Enterprise SIP, visit www.xo.com/esip

Press Release;
HERNDON, VA – May 5, 2010 – XO Communications (OTCBB: XOHO), 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.
“With today’s economic climate, it’s more important than ever before that enterprises evaluate their managed voice options to ensure they’re allocating their IT budget in an efficient and strategic manner for their business,” said Mike Toplisek, chief marketing officer for XO Business Services at XO Communications. “By bringing together industry-leading experts on SIP trunking services, we hope to empower enterprises with the tools and information necessary to transform their distributed voice network architecture into a more centralized and cost-effective VoIP solution.”
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.

Verizon Global Wholesale VoIP IP Telephony Offers Promotions, Functionality And Security Boost.

Verizon SIP Gateway Service, Advanced Toll Free IP Termination and Carrier IP Termination Transport http://snapvoip.blogspot.com/

While touting to be the only Major Carrier to Offer IP Security Tunnel to Maintain End-to-End Confidentiality and Integrity of Data, Verizon has taken some major steps to make it's wholesale VoIP services very attractive. First and foremost is the security and integrity of customer data offered in variety of packages. You can read more in the press release after the jump.

Currently Verizon also offers some promotions for the VoIP services;

NEC IP PBX - SIP Gateway Bundle Promotion:
• 48 or 60 month 0% Fair Market Value Purchase Option: lease plan for NEC IP PBXs
• Offers a 40% saving on monthly recurring charge for Two-way Local and Two-way complete Telephone
numbers
• Applicable to New and Existing SIP Gateway Customers
• Promotion Period end June 30, 2010 

SIP Gateway Install Waiver:
• Waives any installation charge associated with new TN’s being ported over to the Verizon network under the
SIP Gateway product
• This promo waives the install charges for both Ported and Native TNs
• Promo in effect for 6 months from the signing of the promotional addendum
• Customers must meet the following requirements to be eligible for this promotion:
o Signed SIP Gateway Attachment to their TSA or WTSA
o Provided a forecast of anticipated number of TNs
o Completed interoperability testing
• Subscription period ends June 30, 2010
SIP Gateway Telephone Number MRC Promotion:
• Offers a reduced monthly recurring charge for Two-way Local and Two-way complete Telephone numbers of
$1.99
• This rate applies to Two-way Local with metered local or Two-way Complete with metered local telephone
numbers
• Applicable to New Customers only
• Subscription period ends June 30, 2010

BASKING RIDGE, N.J., May 6 /PRNewswire/ -- Verizon Global Wholesale is now offering additional functionality for three of its product platforms, providing voice-over-Internet-protocol customers with a secure connection for transmitting information over the Internet and other public networks.
This new functionality enables the product platforms -- SIP Gateway Service, Advanced Toll Free IP Termination and Carrier IP Termination Transport -- to share a single, secure virtual private network connection called an IP Security (or IPSec) tunnel. Currently, Verizon is the only major provider offering this connection, which provides additional security during call setup.  Because IPSec is an industry standard and supported by most major equipment vendors, there is no need to create vendor-specific security protocols.
"We bypass other security protocols in favor of IPSec tunnel because it offers the greatest level of flexibility to meet our customers' security needs for VoIP traffic," said Mike Yancey, director of wholesale voice services for Verizon Business.  "Verizon wholesale customers can rest assured we go the extra mile to protect their business traffic on our world-class global network."
An IPSec tunnel maintains integrity and confidentiality because all data is encrypted prior to transmission, then authenticated upon receipt to verify the data has not been altered. The source of the data is also authenticated to prevent the possibility of such security breaches as "spoofing" an IP address – a fraudulent network technique in which a program or person masquerades as another.
Verizon's wholesale VoIP products continue to grow as more customers migrate from traditional voice platforms to IP platforms while taking advantage of the quality and extensive reach of the Verizon network.
Verizon's SIP Gateway Service, which carries billions of minutes of use each quarter, comes in four configurations: Inbound Complete, Two-Way Local, Two-Way Complete and Outbound.
SIP Gateway Services are complemented by Carrier IP Termination Service. This offering allows customers to take advantage of the cost savings of originating long-distance traffic and terminating that traffic via SIP protocol over a global network.
Verizon's Advanced Toll Free Service, which has been growing since its introduction in 2008, allows U.S.-based customers to support toll-free calling solutions on a global basis. Verizon's International Toll Free Service supports in-country toll-free dialing outside the U.S. across more than 80 countries and Universal International Freefone Numbers across more than 40 countries, giving customers a superior offering that combines expansive coverage with flexible configurations and robust features.

Verizon's SIP Gateway Service
               Carrier IP Termination Service.

               Advanced Toll Free Service

Sangoma Releases D100 Voice Transcoding Card And Supports Open Source Telephony Applications

Sangoma D100 http://snapvoip.blogspot.com/
Sangoma has released the D100, a PCI or PCI Express voice transcoding card (converts simultaneous channels of transcoding from one type of codec (e.g. G.711) to another (e.g. G.729)) which is designed for optimum voice quality for low to high density system! Voice transcoding is an essential faction of a VoIP IP Telephony System and the Sangome D100 supports transcoding sessions to up to 30, 60, 120, 240 or 480 channels of any-to-any voice codec conversion per card.
Among numerous capabilities of the card system, the outstanding features include;

    * 30 to 480 transcoding sessions
    * Integrates in Asterisk® and FreeSWITCH™
    * Codecs supported: G.711, G.722, G.722.1, G.726, G.729AB, GSM-FR, GSM-EFR, AMR, AMR-WB (G.722.2), iLBC, L8 (Linear 8K), L16 (Linear 16K)
    * Supports Linux (all versions, releases and distributions from 1.0 up)
    * Compact 2U form factor
    * PCI and PCIe

 Here is the press releasel
TORONTO, ONTARIO--(Marketwire - May 4, 2010) - Sangoma^® Technologies Corporation (TSX VENTURE:STC), a leading supplier of hardware and software enabling server-based voice and data communication applications, today announced the launch of a new series of PCI and PCI Express cards enabling up to 480 channels of voice transcoding per card.
"Voice transcoding is an essential, but costly, component of most VoIP applications and networks," said Frederic Dickey, director of product management at Sangoma. "While software solutions can reasonably handle a handful of channels on a server, our card-based solution allows for hundreds of simultaneous voice transcoding channels in a single server, without compromising on audio quality."
Most IP telephony applications require the use of multiple types of voice codecs, which are used to digitally compress the voice signals at any given deployment. While voice signals from the Public Switched Telephony Network (PSTN) always come in the form of the G.711 codec, the VoIP terminal equipment and networks support a variety of voice codecs including G.729, G.726, AMR, etc. VoIP infrastructure most often needs to include the capability to mediate between endpoints supporting different codecs, but this 'transcoding' digital signal processing task is often costly, resource intensive and can affect the quality of the voice signals if it introduces too much latency or delay.
The D100 card, available in PCI and PCI Express form factors, allows to convert numerous simultaneous channels of voice communication from one type of codec (e.g. G.711) to another (e.g. G.729), without affecting latency or using up precious host CPU resources. The card allows to run 30, 90, 120, 240 or 480 channels of any-to-any voice codec conversion, with unmatched quality.
The D100 software drivers provide "plug-and-play" capabilities for both Asterisk^® and FreeSWITCH™, two leading open source telephony projects. With the compatible drivers, the open source telephony platforms can use the D100 cards as seamless voice transcoding resources. This means that existing Asterisk and FreeSWITCH applications can readily start leveraging the D100 capabilities. Further, the open source telephony software can be used as a gateway or session border controller to provide network-based transcoding services.
In open source telephony software, if a hosted PBX vendor wants to reduce the cost of bandwidth to reach customers' SIP phones via a data connection, the cost, scalability and latency issues of software-based transcoding solutions would be prohibitive. However, the D100 card provides a reliable and cost-effective alternative.
In a contact center operation, the cost of voice transport over a data link to connect remote agents can rapidly become a sizeable part of the operational budget. Instead, the D100 card allows the contact center to use the low bit-rate codecs for connecting remote agents to centralized infrastructure and local telephony lines, thereby reducing the cost of connections to remote agents.
"The D100 card allows us to deliver, as an integral part of our hosted contact center infrastructure, the low bit-rate codecs required by our customers, using a flexible, centralized and affordable open source solution," said Matt Florell, president of the Vicidial Group, an enterprise-class, open source call center suite. "It delivers great audio quality in a small form factor and a very attractive cost per port."
Starting at a list price of US$ 750, the D100 card is currently in customer trials and is expected to be generally available for purchase in the current quarter.

“Truth in Caller ID Act of 2010” Passed In Congress, Makes Caller ID Spoofing Illegal.

Voice 2.0 IMS 2.0 has published about Congress passing the “Truth in Caller ID Act of 2010” (PDF) act
and making caller ID Spoofing an illegal act under certain circumstances, such as with the cases intent to defraud or deceive people. The law covers Legacy Telephony, VoIP IP Telephony, basically all telephony.
Caller ID Spoofing is, changing what someone will see on their caller ID display when they receive a phone call from a spoofed call. You can make a call to someone and make it appear to come from someone they know through caller id spoof.

Some of the legal caller id spoofing that are allowed are;

• Commercial answering-service bureaus which forward calls back out to a subscriber's cell phone, when both parties would prefer the CNID to display the original caller's information.
• Most calling-card companies display the Caller ID of the calling-card user to the called party.
• Business owners have been known to use Caller ID spoofing to display their business number on the Caller ID display when calling from outside the office (for example, on a mobile phone).

You can read more on the suject by by following the article on Voice 2.0 IMS 2.0

Spokn, Web Based Telephony Service Launched!

Spokn snapvoip.blogspot.com

Andy Abramson led me to this new web based VoIP IP Telephony service Spokn out of India. It seems like They have managed to get together Google Voice, SIP trunking with features likes of from Truphone together.
The company and the product is offering a global service at a cost effective rates. Spokn is available on Windows Mobile, Nokia, blackberry, PCs and any standard SIP device like SIP Phones. Check it out! links after the jump.
Here is their press release from last year when they launched the product;

Mumbai, India, Nov 10, 2009: Geodesic, an innovator in software products focused on Information, Communication and Entertainment, enters the voice communication space with  Spokn, the definitive internet telephony service.
Internet voice communication so far has always had the feeling of compromise – With Spokn that’s finally no longer the case.
Spokn is the most comprehensive service internet telephony service to date with a host of breakthrough features that will change the way voice communications is used forever.
A global service from day one with a worldwide launch, Spokn is a new generation service that is a true bridge between traditional Public Switched Telephone Network (PSTN) and Voice over Internet Protocol (VoIP) services.
Traditional internet telephony has required one be connected to the internet. Spokn frees internet telephony from the tyranny of the computer giving users the benefit of inexpensive internet calling rates on a wide variety of devices – mobile phones, smart phones, regular land lines in addition to computers. Incoming calls can ring any or all of these devices keeping Spokn users in touch whether they are online or not.
Spokn’s Free Incoming Service is a world’s first that allows Spokn subscribers to be reached at the cost of a local call from over 40 countries worldwide for free. And at a small cost they can even choose to receive these incoming calls on their mobile phones or land lines.
Voice Messaging Service (VMS) is another Spokn breakthrough. With VMS one records and sends a 20 sec message to any telephone or email account worldwide. VMS to emails are free and VMS to phones have a low and fixed cost that includes a free 20 sec reply.
Spokn also offers a traditional voice mail that is delivered to subscribers email address at no cost – when users are not connected to the Internet or are offline
Spokn also brings subscribers a free Voice Mail facility. Callers are directed to record a voice mail that is delivered to the subscribers email address at no cost when one is not connected or offline.
Spokn subscribers have the unique option to acquire their own local phone number in any of ten countries at just $1 a month. This way they can be reached at a cheaper-to-call country. Businesses with a client base in these countries can be accessible to their clients at a local call as well.
Underlining all these innovations is the fact that Spokn brings to market perhaps the lowest prices long distance calling has seen. In addition Spokn prices are a refreshing change from the complications and hidden charges of many internet telephony operators. Spokn has the transparent and clear cost structures with no hidden charges, connection fees and complicated caveats. Calls are charged in blocks of a minute and credits never expire.
These are just some of what will make Spokn the reference point for all voice communications providers.
Mr. Anil Raj heads Spokn. Anil is a seasoned telecom professional who has spent over a decade with the Ericsson group in Asia and Europe and in addition to launching the Hutch brand in India has been Chairman of Symbian Plc as well as on the board of numerous companies.
Mr. Anil Raj says, “Spokn is a true revolution in the world of telecom. Not just a new technological offering but also a new attitude. Innovative but also honest, radical in approach but simple and straightforward to use. Finally I think callers have a product that truly brings together the promise of Internet telecommunications with the convenience of traditional telephony.”
Mr. Kiran Kulkarni, Co-founder/Managing Director of Geodesic Limited adds, “Spokn is an important offering that has the potential to redefine the way voice communications and the Internet synergize. There is a huge and growing scope here and our stakeholders can look forward to many exciting developments from Spokn that changes the way people communicate.
In addition, Spokn completes Geodesic’s suite of communication services adding Voice to our current portfolio of email, Internet Messaging(IM) and SMS offerings keeping Geodesic in the forefront of communications providers.”

Andy Abramson 

Spokn can be downloaded from http://www.spokn.com

VoIP IP Telephony, Leading Tech Companies And Google

VoIP IP Telephony And Google snapvoip.blogspot.com
As usual Andy Abramson (VoIP Watch) made me read a whole post in a gulp and come away educated. Andy responds to some points raised by John Dvorac of PC Magazine regarding Google Voice and the lack of VoIP IP Telephony services by other tech companies. He lays out how others had the right ideas of which some failed, some stayed lukewarm and some actually lead to the formation of Google Voice.
Very nice refresher
Where Have The Big Internet Players Gone in VoIP and Why

Talk Perl To Google Voice

Send System Alert SMS via Google Voice
I have quite a bunch of servers under my command ranging from VoIP IP Telephony servers, to Oracle database servers. The permanently alive ones has monitoring services connected and informs us via email, SMS and simple phone call.
But half of the above severs are experimental at any given time and assigning a monitoring service, either paid or free, could wake up your colleagues in the middle of the night, if you are playing around with the servers. Even yourself might be falsly alarmed if you phone goes beep in a quite night, when you stop some srvice on a remote server.
I found a perl script that can send SMS messages via Google Voice. So now I have this script on my experimental severs (and on a central watchdog server for permanent servers) that alerts me via SMS through Google Voice.
The script was written by a Trixter, (Bret McDanel trixter AT 0xdecafbad.com) and some contributors whom he credits. The script is written in perl and named googlevoice.pl.
If any of my vital services go down, the server itself or the watchdog server can send an SMS message to 123-456-7890 via the script

./googlevoice.pl sms 1234567890 Alert Server XYZ lost a vital service

 In addition to sending SMS, the script can also place and cancel calls;

./googlevoice.pl call 1234567890
./googlevoice.pl call 1234567890 9012345678

or 
         ./googlevoice.pl cancel

You will find the script and Trixter at 0xdecafbad.com or download it directly. But visiting the site will give more infromation and you can join the conversation too!

Continuing To Be Among Top VoIP Blogs!

Continuing On Top VoIP Blogs!Just as we were settling down from the excitement of being named "7 VoIP Blogs 2 Watch", We Have been listed again!
This time by Compare Business Products. The list selects 33 VoIP bloggers and we are one of them. 
 "
VoIP Telephony is another covers-everything VoIP blog that has been runnning since 2005.

http://snapvoip.blogspot.com/
"

I guess it pays to work hard :) Now we will even work harder!
Thank you Compare Business Products

Making Calls With Google Voice

Google Voice Howto.
Once you sign up and activate, Google Voice, unlike other VoIP IP Telephony services, it does not replace your existing phone number. In fact you receive a phone number with capabilities to direct all calls you receive at this number to any of your existing phones.
But sometimes you may want to make a call from your new Google Voice phone number. Say for instance you want to display Google Voice number in Caller ID, Yes it is possible with Click2call feature from Google Voice.
You will need access to the internet and from any device or any browser, surf to Google Voice site
Click the Call button at the top left of the site, in the resulting dialog, enter a contact’s name or phone number if it is in the contact database. Otherwise just enter your destination number. Then in the second box "Phone to Ring" drop down and select the phone you want to use for the conversation. But say if your phone is not working or if you are at another place, enter the number where you are at and click the Connect button. Google Voice will call you at your number and when you pickup you will hear a dialing tone, that is Google Voice calling the destination. If the person at the other end picks up phone the call will be completed! It is that easy.

Tags: Google Voice phone number, Google Voice, Caller ID, VoIP IP Telephony, Google Voice Howto