OpenVBX is an open source Web-based business phone system that’s powered by Twilio. It’s just like a lightweight PBX with typical features such as menus, voicemail, messaging, etc. except there’s one catch: it requires Twilio as the cloud telephony provider.

Not cool, right?

That requirement translates to OpenVBX having some limitations such as no international numbers or international SMS, along with some, shall we say, “speech impediments.” The good folks at Disruptive Technologies kindly released OpenVBX 2.0 that incorporates bug fixes, Tropo compatibility and features, and VoiceVault biometrics API.

Or as Chris Matthieu of Voxeo (backers of Tropo) puts it, OpenVBX has been “jailbroken“:

The coders at Disruptive Technologies added full support for the the Tropo API and Phono SIP-based VoIP web phone to the communications layer of the OpenVBX project. Of course, when selecting the Tropo API, users will now get access to all of the more advanced features of the Tropo network: speech recognition and text-to-speech in 24 languages, phone numbers in over 40 countries, international SMS, in/outbound SIP VoIP support, inbound Skype support, multiple phone numbers per callflow script, improved conferencing.

Disruptive Technologies also extended OpenVBX with the VoiceVault API to support Voice Biometrics in password resets. After adding VoiceVault credentials on the API Accounts Tab, the password reset dialog will provide an option to request a phone call to reset your OpenVBX account password.

Giving users more choices is always awesome especially in the exciting world of Web telephony. It’s also interesting to note that Tropo is Twilio’s biggest competitor and each command a loyal following of developers.

Now I cannot wait to see the next iteration of OpenVBX…

Some folks have half-jokingly said that Skype is a good fit for Microsoft because both companies like proprietary technologies. There’s certainly some truth to that. Even though both of these companies enjoy great numbers of users, they are far from being proponents of open standards and technologies.

That’s why it’s important to spread the gospel of an open source project such as GNU Telephony:

GNU Telephony is a project to enable anyone to use free as in freedom software for telephony, and with the freedom to do so on any platform they choose to use. We also wish to make it easy to use the Internet for real-time voice and video communication, and in fact for all forms of real-time collaboration. Finally we wish to make it possible to communicate securely and in complete privacy by applying distributed cryptographic solutions. Our goal is to enable secure and private real-time communication worldwide over the Internet that is free as in freedom, and is also free as in no cost too!

Over the weekend the project released GNU SIP Witch 1.0, the first stable release of GNU SIP protocol provisioning and peer-to-peer call server. It’ll also be used in the “anti-Skype” open source project, GNU Free Call (covered here previously).

The acquisition of Skype by Microsoft has made some people nervous about the future of the popular VOIP service. This is exactly why open source projects are important and deserve our support.

The official announcement:

May 14, 2011 (Bayonne, NJ). We are distributing today a 1.0 release of the GNU SIP protocol provisioning and peer-to-peer call server, GNU SIP Witch. GNU SIP Witch is developed within GNU Telephony and has been selected for use in the GNU Free Call project. This will provide a stable release that we will support for existing applications while we actively develop GNU Free Call services.

GNU SIP Witch is available as part of the GNU project. Stable releases will also power a web site later this summer to provide initial worldwide secure calling services for free directly to the general public for use in conjunction with any ZRTP enabled standards compliant softphone applications and SIP devices. GNU SIP Witch can be used to deploy private secure calling networks, whether stand-alone or in conjunction with existing VoIP infrastructure, for private institutions and national governments.

GNU SIP Witch is distributed as free software, that is, it is licensed using the GNU General Public License (GPL) version 3 (or later), explicitly to provide others the freedom to use, modify, learn from, redistribute, and participate in it’s continued development, and can be obtained in source directly fromhttp://ftp.gnu.org/gnu/sipwitch. A number of GNU/Linux distributions already distribute GNU SIP Witch in binary form for easy installation. GNU SIP Witch is cross-platform and can also be built on Apple OS/X, BSD systems, and for Microsoft Windows. Future releases will also support Android devices for use in GNU Free Call. Our services and applications are intended to offer the benefits of software freedom on all common computing platforms.

GNU SIP Witch is a free software project and is being developed by volunteers from around the world. The Free Software Foundation and the GNU project provides technical, infrastructure, and organizational support for GNU SIP Witch development. Future work will focus on delivering GNU Free Call services such as self-organizing peer-to-peer calling networks directly to the desktop and mobile devices of users worldwide.

In conjunction with this release, the GNU Free Call project is distributing an initial release of our technological assistance package for common computing platforms by providing our switchview desktop client for use with GNU SIP Witch on your local machine. In the future TAP will enable multi-platform personal encryption, include further support for desktop and mobile secure calling, and provide other basic and common computing services missing on some platforms.

About the Free Software Foundation:
The Free Software Foundation, founded in 1985, is dedicated to promoting computer users’ right to use, study, copy, modify, and redistribute computer programs. The FSF promotes the development and use of free (as in freedom) software—particularly the GNU operating system (used widely today in its GNU/Linux variant)— and free documentation. The FSF also helps to spread awareness of the ethical and political issues of freedom in the use of software. Their web site, located at http://www.gnu.org/, is an important source of information about GNU/Linux.

About GNU Free Call:
GNU Free Call is a project to develop and deploy secure self-organized communication services worldwide for private use and for public administration. We use the open standard SIP protocol and GNU SIP Witch to create secured peer-to-peer mesh calling networks, and we welcome all participation in our effort.

Contact Information:
Haakon Eriksen – Project Coordinator – haakon.eriksen at far.no
David Sugar – Project Architect – dyfet at gnu.org
Mailing List – Participation – sipwitch-devel at gnu.org

 

I love Skype. It’s no secret that I’m a fan of this communications tool. The software evolved from desktop-only to an app that also resides on your smartphone. Recent feature enhancements include desktop sharing and group video. The company has grown tremendously to a point that it hired a Cisco exec as CEO and plans to IPO this year.

But I also realize that it uses proprietary technology. It is voice-over-IP but it is not an open standard. Now a new open source project dubbed GNU Free Call will attempt to offer everything Skype has and then some: SIP adoption.

That’s a big “and then some”… An excerpt of its goal:

What
Our goal is to make GNU Free Call ubiquitous in a manner and level of usability similar to Skype, that is, usable on all platforms, and directly by the general public for all manner of secure communication between known and anonymous parties, but without requiring a central service provider to register with, without using insecure source secret binary protocols that may have back-doors, and without having network control points of any kind that can be exploited or abused by external parties. By doing so as a self organizing meshed calling network, we further eliminate potential service control points such as through explicit routing peers even if networks are isolated in civil emergencies.

That is quite ambitious, but I have faith in the open source community and will be tracking the project’s progress regularly.

Digium and snom are two companies that have been in the VOIP market for a long time and both are fierce proponents of open source and open standards. This blog has covered them here and here. I spent a little time with both companies at the show to find out how things are going in lately…

Digium

Open source FTW

Angie Reed, Product Manager, and Mark Amick, Director of Product Management, were very excited to share with me some latest developments in Asterisk, the open source telephony project primarily known for the IP PBX. Here were the highlights:

• While most companies are now starting to wake up to the world of unified communications, let’s not forget that Asterisk should be considered a UC pioneer, capable of common UC features even 4-5 years ago
• Lots of growth in Europe where many governments and municipalities have declared “open source only” in its IT systems
• Often used in disaster or crisis situations (e.g. Haiti earthquake)
• Developer community continues to flourish: 8,000 code commits in 2010
• Asterisk Exchange, an online market place for Asterisk products and services, continues to grow
• Version 1.10 by AstriCon 2011
• TCO study Switchvox vs. Other IP-PBX finds over 60% savings on initial investment and 80% on total cost
• Asterisk SCF (Scalable Communications Framework) will give Asterisk even greater capabilities and capacities; beta by September 2011; version 1.0 by end of 2011

snom

snom knows VOIP

COO Mike Storella graciously took some time to demo snom IP phones (there were a lot!) at the booth. I also found out that the company makes a conference speaker phone to rival Polycom (HD voice and all). If you’re a fan of a multi-platform (Linux, Windows, Mac) IP-PBX then you owe it to yourself to stop by the booth for a snom ONE Free CD. Now due to customer demand, snom makes an actual server — snom ONE plus — to house its software. All snom products are sold through distributors and VARs, however. The company has grown and continues to grow in markets worldwide. (I actually saw quite a few snom IP phones being used at other booths on the show floor.)

John Stepp, president of Free Tech Consultants, continues the coverage of ITEXPO held in Miami Beach earlier this month.

While covering StartupCamp 3 at ITEXPO in Miami, I also had an opportunity to interview some established companies.  I sat down with snom COO Michael Storella to learn more about the snom phones and the snomONE (announcement covered here previously), an SMB IP phone system that was just released in October 2010.

Once we discussed their upcoming M9 DECT phone with color display and the snom 300 being the only standards based phone qualified by Microsoft on Lync, we started discussing the snomONE.  The snomONE consists of three different offerings:

• snomONE Free for up to 10 extensions downloadable from the snom website for free
• snomONE Yellow for up to 20 extensions at $895
• snomONE Blue for up to 150 extensions at $1495

Since the smallest offering is basically freeware, it is very similar to open source.  The only caveat is that you have to use snom devices on it.  The phones are not the flashiest out there, but given the low cost and high reliability of their devices, the snomONE is a really good deal.  While you are forced into the proprietary desktop devices, you also get a secure software stream that is supported by snom, so you do not have to worry about the topsy-turvy world of open source software updates.  If you choose to change systems in the future, you can reuse the phones since they are standards based SIP compatible.  So, to me, it really does occupy a middle ground between open source and a fully proprietary phone system.

Storella stressed the reliability and security of the system and the fact that it could operate on Microsoft, Linux, or Mac OS X. The ability to deploy on Macs makes it a real fit for small Mac shops.  But the thing that struck me most was the feature set.  Snom is changing the dynamic by providing a host of PBX features such as unified messaging, cell phone integration, IVR, call center and call recording in the snomONE product line. Given the complexity of the feature set, I would use a VAR to help with initial setup of the system.  Not that the system is not intuitive, it’s plug and play.  The system can do so much that you will want to take advantage of all that it has to offer and learn the settings changes that you would want to utilize on an ongoing basis.  It is best to get front end training from a VAR no matter which system you want to use.  I highly recommend checking out snomONE if you are a small business that requires secure complex call handling features and have experienced sticker shock with other systems.

Just a few days ago Digium proudly released Asterisk 1.8 — over 200 enhancements, security updates, and new features. This version will even work with Google Voice and calendar integration with Outlook, CalDAV, and iCalendar. And that’s in addition to the fancy PBX-y stuff it already does like caller ID, call waiting, hold, transfer, etc.

More reason for any small/medium business to evaluate Asterisk’s potential usefulness in team communications and collaboration.

Also, AstriCon starts today in the nation’s capitol. Too bad I cannot be there to bask for three days in open source nirvana, but if you’re in attendance and feeling generous, why not give us a glimpse by leaving a comment below?

This conference is somewhat refreshing in the telecom industry because you won’t see the usual suspects there — no Cisco, no Avaya, no Siemens, no Microsoft, no Aspect, no Alcatel-Lucent, … There, I just gave you a few good reasons to attend AstriCon next year!

The world can never get enough of open source goodies, especially in the realm of benefiting the communications community. Most of us enjoy reasonably good to really exceptional audio over IP with common programs like Skype and Google Voice, not to mention high quality (and expensive) gears from the likes of Avaya and Cisco. Obviously, the higher the voice quality, the more bandwidth it requires. That’s why there are companies whose bread and butter are compression algorithms and patented codecs.

Well, good for them. But a lot of the world still operate in low bandwidth, dial-up speeds. Fancy HD audio codecs wouldn’t stand a chance in many third world or developing economies. Cost effective communications technology is something that builds communities, spurs innovation, and stimulates economies. There is definitely a need for royalty free, open source, implementation friendly audio codecs.

Enter Codec2, first concocted by David Rowe, an engineer with a Ph.D. from south Australia.

Rowe is an embodiment of the open source movement. In his own words from his blog:

I’m David Rowe, an electronic engineer living in Adelaide, South Australia. My mission is to improve the world – just a little bit. I do this through releasing Free (as in speech) hardware designs and writing open source software for telephony. My passion is low cost telephony for people in the developing world. A telephone call should be a human right, not a privilege.

In January 2006 I quit corporate life as an Engineering Manager to become an open source developer. I now develop open telephony hardware and software full time. I like to build advanced telephony technology – then give it away.

Amazing. The world needs more generous developers like him!

More importantly, his work has gotten the attention of other developers and Ham radio enthusiasts. For example, popular open source soft switch FreeSWTICH recently included Codec2 support.

Currently in V0.1 alpha release, Codec2 is a fully functional 2550 bit/s codec. Check out the audio samples on the project page — not shabby at all for an alpha.

The project is looking for testers in real world implementations as well as developers and sponsors. Please consider helping its efforts.

While the industry is abuzz about UC products from Microsoft, Cisco, Avaya, Siemens, and more, nobody has given much thought on open source UC solutions. Why not? We owe much thanks to open source projects like Linux and Asterisk to create the competition and innovation seen in modern software and telephony applications. Can anyone imagine a Web without Firefox and Chrome? Or office productivity without OpenOffice? Or customer relationship management without SugarCRM? Or database systems without MySQL? Or more recently, mobile communications without Android?

So as the likes of Microsoft, Cisco, and Avaya pump out press releases and tweets like there is no tomorrow in order to gain an edge in the market buzz, I’ve wondered why there hasn’t been much news about open source UC. Obviously the free market has created the demand for UC solutions from Big Telecom, and whenever there is a demand there’s usually open source competition, especially in an enterprise application like UC.

I managed to find a couple: Druid (by Voiceroute) and Elastix (by PaloSanto Solutions).

Unfortunately, Druid’s site hasn’t seen updates since late 2009, even though the forum still has some recent activities.

Elastix is developed by an Ecuadorian company which got its start in California. The site seemed to have been last updated in July 2010.

I do hope these and more open source UC solutions survive and flourish. If the company you work for is an open source advocate then consider contributing to projects such as Druid and Elastix. Having more options to choose from on the UC menu can only be a good thing to customers.

Remember the good ol’ days when you could build (or buy) a blue box to make free long-distance phone calls? The blue box generates the 2600 Hz tone which instructs the old phone switches to allow a LD call. It’s funny, be it Karma or whatever you call it, now the phone companies get to stick it to us with convoluted billing statements, overpriced SMS plans, and ridiculous tethering allowances.

But I digress. The 2600hz Project and what it aims to do:

2600hz is home to a collection of open-source telephony software that enables the use of the FreeSWITCH, Asterisk and YATE switching libraries. Initially built around the blue.box project, we aim to provide a collection of software to power your GUI, your cloud-based telephony switch and/or your monitoring and maintenance tool set.

The project was started by some core developers behind FreePBX. According to Om, the venture received $250,000 in funding already.

It’s always good to know that open source telephony continues to receive the attention it deserves.

On Monday at VoiceCon I attended an afternoon session about the adoption of open source PBX in the enterprise. The session was presented and led by John Malone, CEO of the Eastern Management Group. Steve Sokol, Director of Product Marketing at Digium, and Jeronimo Romero, Managing Partner at EUS Networks, were panelists.

Malone’s team did a study on open source PBX market share and found that it has an impressive foothold:

Open Source PBXs accounted for 18 percent of the 15.88 million line market for business telephone systems in North America in 2008, based on a new Eastern Management Group study “2009 Open Source PBX Market Forecast”. This represents a 40 percent growth of Open Source PBXs in the last year. Open Source now accounts for a larger slice of the market than any single manufacturer of PBXs, and eight percent more than Nortel, the largest traditional vendor.

Additionally, open source PBX usage is found in all sectors and sizes of businesses, debunking the myth that it’s often the high-tech firms that embrace open source PBX. And the primary driving force? Value, of course. No licensing costs, no hidden fees, no upgrade purchases. Anyone can go download a full-featured PBX like Asterisk today and have it up and running in no time, albeit with very basic setup.

If you’re an open source fan like I am, then you’re probably very excited about the seemingly bright future ahead for open source PBXs. However, Malone predicts open source telephony gaining no more than 33% of the market based on his research into Linux’s history. We’re all aware of Linux’s catapult into stardom and receiving attention from the likes of IBM and Microsoft. Yet today Linux’s market share appears to have plateaued. According to Malone, there are just some companies that won’t be receptive to open source software and there’s nothing to be done.

Still, it’s worth noting that Digium, a company with less than 200 employees, has created a telephony project in just a few years to capture nearly 20% of the market and turning up the heat on Big Telephony. However, one major advantage for Big Telephony is its well-established VARs and integrators — these are the guys who are driving the sales of Big Telephony PBXs.

The road ahead for open source PBX will be filled with potholes and bumps. Steve Sokol’s job to promote Asterisk and other open source PBXs remains challenging. But if he can convince some folks from the flourishing Asterisk community to take a good look at the commercial opportunities in becoming a VAR or systems integrator (like Jeronimo Romero), then there may be a chance of smashing Malone’s prediction of 33%.