Source: snapvoip.blogspot.com

FCC, Federal Communications Commission, states that "Voice over Internet Protocol (VoIP), is a technology that allows you to make voice calls using a broadband Internet connection instead of a regular (or analog) phone line. Some VoIP services may only allow you to call other people using the same service, but others may allow you to call anyone who has a telephone number - including local, long distance, mobile, and international numbers. Also, while some VoIP services only work over your computer or a special VoIP phone, other services allow you to use a traditional phone connected to a VoIP adapter."
Let’s see how others describe it.
Howstuffworks tries to give you much information as possible with the article "How VoIP Works". I have gone through the complete article and found it to be ok. My writing ability is not as good as Robert Valdesz (It is getting there), otherwise I would try to write a simpler version as well.
He starts with how VoIP Works, How current PSTN, Public Switched Telephone Network, works which continues to the bits and bytes involved in a regular Telephone call, Advantages of VoIP, Disadvantages of VoIP and Codecs, softswitches and protocols.

All in all you get an idea of How VoIP Works but I wonder if the article ib being kept updated, for instance about E911, A lot has changed since it was written. So that is "How VoIP Works" until my writing improves. If I find any other VoIP introductions, I will Post them here.

Source: snapvoip.blogspot.com

After acquiring SwitchVox, Digium has released a free version of the SwitchVox. This was announced at the Fall VON, but the press release was buried under all those notes I took at Fall VON. Better later than never and here is the Press Release.
Also you can download the release directly from SwitchVox site. It is great way to test or have a trial run of the software which comes in a easily installable package. But for production, it is suggested that the the commercial version, SOHO or SMB, is used. The package includes, Fedora Core 6 Linux, the open source Asterisk PBX software, plus all the software necessary to make Switchvox such a revolutionary communications solution.

BOSTON (Fall VON and Digium|Asterisk World) — October 31, 2007 — One month after its acquisition of Switchvox, a leading provider of IP PBX phone systems for small- and medium-sized businesses (SMBs), Digium®, Inc., the Asterisk® Company, announced the immediate availability of Switchvox Free Edition. The new product includes the telephony features of the widely used Switchvox SOHO edition and can be downloaded and installed in minutes on existing hardware. The release underscores Digium’s commitment to delivering full-featured telephony systems at low price points to allow organizations around the world to benefit from the Asterisk open source movement and flexibility of voice over IP (VoIP).

Switchvox products are based on Asterisk®, the open source telephony software created and owned by Digium. Switchvox Free Edition includes tools such as an interactive voice response (IVR) editor that allows system administrators to create auto attendants and menus. It also includes features such as voicemail to email, find me/follow me, unlimited calling queues (automatic call distribution, or ACD) and advanced reports on system use.

Switchvox Free Edition supports Digium’s line of analog cards. Users can install Switchvox Free Edition on existing hardware or use Switchvox-certified hardware. For details on hardware support, see www.switchvox.com/certified_hardware.

“Asterisk is the most widely deployed open source telephony platform, and by introducing Switchvox Free Edition, Digium has the opportunity to introduce the technology to a broader and more mainstream customer base,” said Tristan Degenhardt, Switchvox product line director at Digium. “We anticipate thousands of downloads in the coming weeks by companies interested in a better way to communicate with their customers. The smallest businesses will find Switchvox Free Edition to be an excellent way to get started with a full-featured phone system and be positioned to easily transition to more advanced Switchvox systems as their needs change and grow.”

Switchvox Free Edition is on display this week at Fall VON 2007 and Digium|Asterisk World at the Boston Convention and Exhibition Center. The product may be downloaded from www.switchvox.com

Source: snapvoip.blogspot.com

Press Release:
Redwood City, Calif., – October 22, 2007 — Nominum, the leading provider of network naming and addressing solutions, announced today the release of Navitas 3.0, an IP-application Routing Directory (IPRD) that uniquely accelerates the deployment of IP telephony. Navitas 3.0 provides a platform that integrates legacy telephony numbering plans and IP routing identifiers, giving carriers an evolutionary path toward profitable business models using Voice over IP (VoIP). Additionally, the directory enables the consolidation and simplification of routing plan management processes, improves telecommunication margins, and optimizes usage of profitable and reliable routes.

Many carriers have made the decision to migrate to IP telephony. They are challenged by declining voice revenues and face increasing competition and industry pressures to adapt their existing business model to remain profitable. At the same time, they need to invest and execute very carefully as they move to IP. Cost, control and complexity are significant barriers. Navitas 3.0 enables carriers to easily transition to an IP telephony business model and extend telephony services to support a voice enabled Web, without having to duplicate network functionality between legacy signaling systems - such as Signaling System 7 (SS7) - and IP.

“As the convergence juggernaut continues, carriers will increasingly struggle to sustain their business model and profit margins,” said Brian Partridge, program manager in Yankee Group’s Enabling Technologies Service Provider group. “Rather than equating VoIP and VoIP peering for settlement free services, carriers are increasingly seeking networking functions such as IP-application Routing Directories to manage and bill telephony services on IP as they see fit.”

Navitas Answers Carriers Needs Today, Prepares Them for the Future
Navitas 3.0 serves as the foundation for next-generation networks like IP Multimedia Subsystem (IMS) by allowing carriers to reuse existing routing plans, and modernize the integration into next-generation networks through automation. Through its enhanced management and reporting functionality, Navitas 3.0 provides a higher Quality of Service (QoS) over IP by ensuring reliable call deliveries across interconnects. The solution also helps carriers avoid increased spending on Time Division Multiplexing (TDM) and SS7 technology, and facilitates the development of new IP-based applications.

"VoIP interconnects are growing around the globe and service providers require scalable address resolution as well as security and control at the interconnect borders," commented Seamus Hourihan, Vice President Marketing and Product Management, Acme Packet. "Both Acme Packet and Nominum are attentive to our customers’ requirements for VoIP peering and we continue to innovate and deliver features designed to build reliable, scalable and trusted VoIP interconnects."

Navitas Delivers Advanced Features
Navitas 3.0 dynamically accesses, stores and manages telephone routes and number mapping, while serving both legacy and converged IP switching infrastructure with route resolution for call termination. The solution features functionality for advanced service routing capabilities such as voice, multi-media sessions and IP applications.

New to Navitas 3.0 are the Application Routing Module and Element Management System. The Application Routing Module (ARM) enables Navitas 3.0 to become a switch-agnostic signaling data repository that refines telephony and multimedia routes based on session parameters like cost, quality, capacity and content (end user context like persona, presence, location, etc.) The core technologies and features include:
• Cascading query engine federates multiple sources of routing data
• Support for parameter-based routing including Least Cost Routing applications improves telecom margins through profitable and reliable routes
• Framework for Service Oriented Routing (SOR) for resolution of multiple routing decisions

“As the industry transitions to an all-IP world, carriers require a robust IPRD that is switch and data source agnostic and can serve as a control point for advanced services,” said Tom Tovar, chief executive officer, Nominum. “Navitas 3.0 provides the pivotal first step toward a cost-effective migration from SS7 to IP. This product addresses the immediate needs of the VoIP and VoIP peering markets as well as future network requirements, without requiring carriers to wait for all-IP or IMS architectures.”

Navitas Element Management System (EMS) is a turn-key solution that gives carriers a single point for administration across a network of Navitas IPRD and ARM servers. It provides enhanced data collection and reporting and allows for accurate Quality of Service (QoS) measurements.

A podcast of this release is available at: http://www.nominum.com/content/spotlight/chunks/Navitas 3.0 launch.mp3 For more information, please visit www.nominum.com.

Source: snapvoip.blogspot.com

Convicted hacker Robert Moore, who is set to go to federal prison this week, says breaking into 15 telecommunications companies and hundreds of businesses worldwide was incredibly easy because simple IT mistakes left gaping technical holes. Moore, 23, of Spokane, Wash., pleaded guilty to conspiracy to commit computer fraud and is slated to begin his two-year sentence on Thursday for his part in a scheme to steal voice over IP services and sell them through a separate company. While prosecutors call co-conspirator Edwin Pena the mastermind of the operation, Moore acted as the hacker, admittedly scanning and breaking into telecom companies and other corporations around the world.

According to DSL Reports, Moore wasn’t all that bright — considering he made only $20,000 out of the more than one million dollars in illegal profit, the majority of which went to 23-year-old mastermind Edwin Pena. Pena, a Miami "consultant," secretly routed some 500,000 calls through compromised Net2Phone systems

Information week

Source: snapvoip.blogspot.com

SCOTTSDALE, Ariz.–(BUSINESS WIRE)–The market for consumer Voice over IP (VoIP) services boomed in 2006, with Europe leading the way, reports In-Stat (http://www.in-stat.com). Total Voice over IP (VoIP) subscribers worldwide increased by 34 million subscribers in 2006, the high-tech market research firm says.

“Europe showed the largest gain in consumer VoIP subscribers,” says Keith Nissen, In-Stat analyst. “The European consumer VoIP market increased by over 14 million subscribers last year. The European VoIP market is being aided by local loop unbundling (LLU), the introduction of cable telephony and triple-play service bundles, as well as operator consolidation. In contrast, US wireline operators added only 4 million VoIP subscribers in 2006. No one seems interested in selling anything other than plain-old-telephone-service”.

Recent research by In-Stat found the following:

• The countries with the largest VoIP subscriber gains in 2006 include France, Germany, and the Netherlands.
• In 2011, the US will represent only 18% of the global consumer VoIP market.
• By 2011, In-Stat predicts the consumer VoIP market will total nearly $44 billion worldwide.

The research,“Europe Leads the Booming Consumer VoIP Market”(#IN0603603WWI),covers the worldwide market for Voice over Internet Protocol. It provides forecasts for global VoIP subscribers, segmented by region, through 2011. It also includes analysis of major VoIP markets around the world. Analysis of market drivers and challenges is provided.

For more information on this research or to purchase it online, please visit: http://www.instat.com/catalog/mmcatalogue.asp?id=272

Source: snapvoip.blogspot.com

According to a news release by Pika technologies, Intuitive Voice, a customer of PIKA, has introduced a low-cost, high-quality PBX phone system designed for small businesses. The Evolution PBX 2.3 combines the cost benefits of open-source PBX technology with the quality of PIKA voice processing, along with an easy-to-use interface designed especially for small businesses with no technical staff.

Chris Jones, President of Intuitive Voice, said, “With Evolution PBX, there are no barriers to prevent small businesses from realizing the benefits of big-business PBX features. At less than $1,500US for a four-line PBX, Evolution PBX is truly affordable for a small business, without sacrificing voice quality or usability.”

For a fraction of the price of traditional PBXs, Evolution PBX provides small businesses with features such as an automated attendant; unique extensions and voicemail boxes for each employee; automatic call routing; integration with Microsoft® Outlook™ and voicemail-to-email; on-hold music; support for telecommuters, voice-over-IP, and much more.

Ease-of-use is assured with the Evolution PBX turnkey software interface, which automates setup and provisioning, and allows non-technical staff to easily maintain the system.

Adds Jones, “These breakthroughs were made possible by PIKA’s high-quality voice processing for Asterisk. PIKA’s low density analog board is unique because it provides advanced functionality such as DSP-based echo cancellation, fax, and native switching, at a very competitive price point.”

Terry Atwood, Vice President of Sales, Marketing and Customer Care at PIKA Technologies, added, “We are very pleased that Intuitive Voice has leveraged our landmark PIKA Connect for Asterisk software to fill a major gap in the PBX market. While the open-source Asterisk solution makes PBX technology affordable, only a company like Intuitive Voice can develop it into a fully-featured and easy to use product that makes sense for small business end users.”

Links;
Pika technologies
EVO PBX by Intuitive Voice

Source: snapvoip.blogspot.com

AS you all know, TrixBox is a telephone system based on the popular open source Asterisk PBX (Private Branch eXchange) Software also know as Asterisk IPPBX. TrixBox was formally known as Asterisk@home for those are slow to catch up! TrixBox allows an individual or organization to setup a telephone system with traditional telephone networks as well as Internet based telephony or VoIP (Voice over Internet Protocol). SugarCRM, a Open Source Customer Realtionship Management, can be integrated with Asterisk, and is bundled with Trixbox.

Now there is a book to help you. Trixbox made easy by Kerry Garrison, something to add to your VOIP IP Telephony toolbox.
The book introduces telephony concepts before Jumping in to creating a PBX, detailing how to plan a telephone system and moving on to the installation, configuration, and management of TrixBox, a feature packed PBX.

This book contains practical examples and tools. It provides examples of well laid out telephone systems with accompanying spreadsheets to aid the reader in building stable telephony infrastructure.
Even if you are a TrixBox, Asterisk Guru, I think it is a handy tool to have near your telephony gear!

Here are the Chapter description;

Chapter 1 introduces the essential telephony and IP telephony concepts to give the reader the necessary background.

Chapter 2 gives an overview of Asterisk the PBX software at the core of TrixBox and gives the reader a feel for the features of a powerful VoIP telephone system.

Chapter 3 explains the relationship between Asterisk and TrixBox and introduces the enhancements and power the combination of these tools provides.

Chapter 4 walks the user through planning a telephone system with accompanying spreadsheets to fill in, in order to properly plan for the installation and configuration of the system.

Chapter 5 gives the reader details on how to install TrixBox and how the basic administration components are used.

Chapter 6 applies the previous planning to the configuration of TrixBox to provide the features the reader requires from their telephone system.

Chapter 7 covers the telephone system from the point of view of the telephone handset and how the user of the telephone system can interact with it.

Chapter 8 looks at more advanced configuration options and differing types of telephone line that can be managed with TrixBox.

Chapter 9 briefly introduces the SugarCRM customer relationship management tool, integrated with TrixBox.

Chapter 10 shows the reader how to secure and backup TrixBox to ensure reliability of their system.

Appendix A has some acronyms and terms used throughout this book, which are also common terms in Telephony. This can be used as a quick reference to the terms when reading the book or configuring the TrixBox system

Links;
TrixBox Made Easy

Source: snapvoip.blogspot.com

Solarion at Geemodo reports about Qnext desktop messaging applicatio, The application said to be a category-breaking application that uses a single Java-based client to securely deliver P2P services such as Universal Instant Messaging (Qnext, MSN, AIM, Yahoo, ICQ), Video Conferencing, Voice over IP, File Transfer, File Sharing, Group Text Chat, Online Games, Photo Sharing and Remote PC Access.
Geemodo: SUN shines on Qnext Desktop Messaging

Source: andyabramson.blogs.com

After looking at the blogs out there I decided that it’s time to start one on Voice Over IP and share my views on the a subject that will be hotly discussed in 2004 and beyond.

When I’m not writing and posting to this BLOG, I’m also covering Voice Over IP at KenRadio on the daily “World Technology Roundup” or on our weekly show, “Speculations.”

So, if Voice Over IP has your interest, this is the place to be.

Cheers,

Andy Abramson