O’Reilly author and ETel regular Brian McConnell has just published a thought provoking essay, The End Of The Language Barrier. Brian is the the leader of the Worldwide Lexicon project, which we’ve highlighted here before. He predicts that the language barrier, at least for published content, will be history in less than three years. In his essay, he argues that the real breakthrough is social, not technological, and that people will organize themselves to create and share translations for interesting web content, much as they already participate in collaborative systems like Wikipedia. If his essay is any sign of things to come (it has already been translated into 5 languages), this is a trend worth watching.

Matthew Chmiel describes his Language Dialer service, which uses Asterisk to record practice conversations for language students in any language in our latest ETel article. Matthew traces the history of the project, born to fill a personal need while he was attending NYU’s Interactive Telecommunications Program, through where it stands today as a fully functioning example of an emerging telephony application that utilizes voice in a web-based service. For an inspiring use of telecommunications services, check out ETel and Your Second Language.

Adhearsion is an open-source framework written in Ruby that was developed to improve complex Asterisk development. Jay Phillips is the creator of Adhearsion and he has written a section about the open source Ruby framework for the upcoming second edition of Asterisk: The Future of Telephony. We’re proud to present this expanded excerpt of Jay’s material here on ETel. Our latest ETel article explains how to get up and running with Adhearsion, provides working examples, and a detailed discussion of Adhearsion’s database integration. For the all of the latest Asterisk info, the newly updated version of O’Reilly’s best-selling Asterisk book is available now as a Rough Cut.

Gershon Goren describes the pitfalls and eventual payoffs that WebDialogs has experienced by partnering with Skype and becoming an active member of Skype’s Developer Program, in our latest ETel article: Skype Developer Program: A Tale of Pioneering and Persevering. Gershon has sound advice for developers considering these kind of partnerships.

I’ll be speaking at SpeechTek in the session “The Future of Speech.” I’ll present a few of my ideas on the future of speech interfaces.

But I need some help, folks. I would like to present a list of the current non-speech interfaces to cell phones, and the list seems rather short. Cell phones have keys, keypads, buttons, and joysticks. Some have Bluetooth-based virtual keypads, although I can’t say I’ve seen one in use; and Apple’s iPhone will have a touch screen. (Some cell phones have speech recognition interfaces, but I’ve rarely seen one in use.)

I’d like to include other up-and-coming interfaces for the cell phone, either commercially available or close to deployment. If you’ve a suggestion about a technology to mention, please let me know by leaving a comment here.

Johannes Ernst explains why we need a user-centric models for identification, and how this change will impact the communications industry in our latest ETel article, Communications in the User-Centric Economy.

Time Magazine famously told us last December that we, the users, control the Information Age. It’s a matter of some dispute whether that is indeed true already or only will be, but there is no question that user-centric technologies enabled by the Web—such as blogging, tagging, video sharing, podcasting, syndication, and others—have been causing a dramatic and accelerating shift of power and control from vendors to users.

What does this mean for communications? Nothing short of turning this multi-trillion-dollar industry inside out. Let me try to explain.

Click on through to Johannes’ article for the goods. And if you’re interested in these issues, don’t miss the upcoming Internet Identity Workshop, May 14-16 in Mountain View, CA, at the Computer History Museum.

There’s been quite a buzz around Sonopia recently, including some interesting observations from ETel conference chair Surj Patel and ETel blogger Imran Ali. Sonopia’s attempting to fundamentally shift the value proposition in the mobile phone market from the carriers to an affinity model where all parties involved share revenue, and have built a platform that allows for user generated content and community building amongst these affinity groups. Aaron Huslage recently spoke to Sonopia founder and CEO Juha Christensen to find out more about this interesting new service. Check out Aaron’s latest ETel article, Sonopia Reinvents the MVNO for an introduction to Sonopia, and stay tuned to ETel for an upcoming in-depth review of the service.

I’ve just gotten five phone calls, one after another, in Spanish. No, wait, it’s ringing again — phone call number six. They’re a few minutes apart.

They’re from AT&T, as far as I can tell, and I haven’t got the foggiest notion of what they’re about or how to make them stop.

Someone is going to pay for this…

We Make Money Not Art are running an article, Pay phone murder mystery, on playful and novel uses for now defunct payphones.

Sadly, they didn’t suggest ratifying the role of Britain’s iconic K6 as general purpose public toilets…despite their urine-soaked, puke-encrusted history ;)

We’ve just published Peter Drescher’s Singing With Your Thumbs: How To Make User Interfaces Musical over on our Digital Media site. This article is a lightly edited version of Peter’s presentation at the 2006 Austin Game Audio Conference, “Audio UI as Interactive Music.” Peter is the Sound Designer at Danger, Inc., makers of the Hiptop mobile internet device. Lots of good stuff here for those who are thinking about interface design and mobile devices. Don’t forget about the audio interface!

A handful of companies own virtual fiefdoms: by government fiat, they control the radio frequencies used in cellular telephone calls. In the US they play this advantage to the absolute maximum. They dictate what services appear on the networks, they provide deliberately-lobotomized cellular phones, they charge outrageous prices that their oligarchy easily sustains. We’re all cellular serfs, dancing to the tune of the spectral aristocracy.

But a revolt is brewing. Skype filed a petition with the US FCC to require that cellular companies disaggregate the sale of airtime from the sale of handsets, and allow the bandwidth to be used in any manner — even to carry Skype VoIP calls.

A few pressure groups (to use the British term) have gotten behind this effort. Zack Exley wants to start OpenPhoneProject.org (which is not yet operational) to collect 100,000 requests for service via an “open” phone. If all carriers refuse to provide this service, he could present that refusal to the FCC as evidence of un-met consumer demand and complain that the existing carriers are not serving the public interest.

I admit that I’m skeptical. Without a firm commitment to purchase, this is a chimera; and 100,000 customers dispersed across the US is hardly a worthwhile market for a nationwide carrier. Perhaps 100,000 in a single city would catch the eye of a cellular provider, but I doubt it.

But the most basic flaw in the plan is that it ignores the economics of of the cellular phone market. The cellular companies sell you a handset; they charge you the cost of service and the handset; the handset is paid off after two years; they continue to charge you the full price, and that’s all found money to them. Additionally, the cost of the handset hangs over your head during the duration of the contract, which binds you to them as a customer and prevents your defection to a company with lower prices or better service.

So will this effort succeed? I think not. Is it worthwhile? Well, perhaps; it’s certainly a low-cost effort — nothing ventured, nothing gained. But the only true solution is wireless VoIP, and that solution depends on reclaiming the airwaves from our current lords and masters.

American Heritage is carrying a very interesting article on the inception of the mobile phone from Motorola’s perspective…

Motorola told its engineers: “We need a hand-held mobile telephone in three months.” Nobody had ever made one before.

Read more here and here…

I am giving a talk titled “Building A Telecom Business Without Selling Your Soul” Tuesday morning at ETel. The talk is about how to bootstrap a business without taking on millions of dollars of other people’s money (I funded all three of my businesses with less than $250,000 in investment). A key theme in my talk is about how telecom is full of niche opportunities which are themselves decent sized businesses. Corporate IVR, which we are all victimized by, is one example of how there are still plenty of opportunities in telecom…

In our latest ETel article, Doesn’t the Social Web Realize That People Talk?, Trevor Baca examines the sad state of voice integration in the social web, looking at the factors that have kept voice from being a common component of many popular web sites and services. Baca will present Voice and the Web: The New Terrain at next week’s Emerging Telephony conference in San Francisco. There’s still time to register for this ground-breaking conference, I hope to see you there!

Power Line Communications (PLC) is the use of existing electrical cables to transport data, and it has been around for a very long time. In our latest ETel article, An Introduction to Power Line Communications, Xavier Carcelle takes a look at the current state of PLC.

Brian McConnell introduces the latest development project in his ongoing attempts to make websites accessible in many languages. The Worldwide Lexicon is an open source RSS and wiki translation service built using Ruby on Rails. Learn all about this exciting new project in Brian’s new ETel article, The Worldwide Lexicon Reloaded.

Brian McConnell wrote a highly-persuasive article about the merits of Nokia’s N80i cell phone. It’s created a genuine quandry for me: should I purchase the N80i (which has WiFi) or the FIC Neo1973? The Nokia has WiFI, which makes internet telephony possible on the phone itself; but the Neo1973 runs Linux…

Now the decision is slightly harder. Tommi, a developer over at Nokia, posted an note on his blog about a software update that transforms an ordinary N80 into an N80i. Notes in his blog indicate that not all updates are successful quite yet, but it also means that if you want an N80i you might be able to take an N80, readily available everywhere, and transform it into an N80i.

In our latest ETel article, The IMS Debate, Lee Dryburgh explores some of the concerns surrounding the IP Multimedia Subsystem (IMS) which will be used in next-generation operator networks and which lies at the core of their future strategies, and sets the stage for a live debate on the topic he will lead at next month’s Emerging Telephony conference. Whether you’re a believer or a skeptic about IMS, you should read this article. Lee has been a consulting engineer for a number of telcos and equipment vendors and is the author of the best-selling book on SS7, so while his opinions on the topic may be controversial, they are also well-informed. Lets get the conversation started now in the comments section of The IMS Debate. (Note: I’ve turned off comments on this posting to encourage readers to leave their comments after the actual article instead.)

You may have heard the persistent rumors that the old AT&T network operated a series of highly-secured, highly-reliable telephone switches. That these switches were buried ten stories underground at strategic locations; that the knowledge about where these switches were located was closely guarded; that the switches were hardened to withstand a near-hit from a nuclear devices.

These rumors are true. Even the wall clocks in these underground locations are shock-mounted on springs so a near-miss won’t shake them off the wall.

I thought of these switches as I read about yesterday’s earthquake and how Internet services were disrupted. The earthquake occurred along a narrow corridor that houses all the undersea cables that connect Taiwan with Hong Kong; and the connections to Taiwan ultimately lead to the US. These cuts and others disrupted both Internet and telephone services across Asia and between Asia and the US.

I will be interested to see what the consequences are for Internet telephony. As companies scramble to restore service, what priority will Internet telephony services be given? Ordinary Internet data can travel by satellite; voice calls do best on wires; a logical choice would be to reserve bandwidth on the remaining cables for voice calls and move data with either lower priority or via satellite. This argues that we’ll see long-term degradation of Internet telephony quality, until service is back up and running; and that could mean movement away from Internet telephony, which will be tagged as less reliable than classical telephony.

The other possibility: since Internet telephony can scavenge packets from any Internet connection, Internet telephony will gain in reputation as something that keeps working even when ordinary connections are unavailable.

Let’s see what happens. It ought to be interesting.

When considering social software, we tend to conceive of software that is a filtered aggregation of individuated and personalised experiences.

Nokia’s Jan Chipchase and Indri Tulusan reframe this perception by asking what happens when people share an object that is inherently designed for personal use?

There’s a lotta talk about the ‘next billion’ mobile customers, largely from the developing world, but very little real empirical study of what those users might need. Contrast Doom-playing OLPCs with the work of The Fonly Institute…

Chipchase and Tulusan’s field study of Ugandan mobile, this past July, documents some very revealing observations…

• Phone borrowing is is driven by cost and price sensitivity.
• Phone lending is driven by hospitality, personal relationships and community well-being.
• The notion of ’sente’, using prepay airtime as a form of cheap, secure and convenient banking.
• Employing missed calls - ‘beeping and flashing’ - as a form of free messaging.
• Phones as community ATMs.
• Pooling prepay credit between customers when sufficiently small prepay denominations are available.
• Mediated Calls - where literacy becomes a barrier to participation.
• Community address books to encourage repeat business and conveniently recall commonly dialled numbers.
• Step messaging - physically carrying a phone containing a message to its recipient…

Chipchase and Tulusan conclude that sharing is driven by cost, but that low costs lowers the propensity to share; with initial experiences governed by sharing, they also conclude that this may shape future usage. It’ll be interesting to see how individual ownership might affect social cohesion and mobile usage in the very same communities.

What’s striking about the research is that all the observed innovations in shared usage are a result of user inventiveness, rather than handset design or network services; a case of user-generated services that really serve the needs of the consumer…if the mobile industry paid closer attention to such innovation, it might provide that ‘next billion’ users with the tools they actually need.

–

BTW, during a vacation in Pakistan this year, I noticed that a lot of people carried 2-3 handsets and SIMs as tools to mediate their friendships, family and professional availabilty…

(I started writing this post at my personal blog, then realised it would actually make for an interesting ETel article, so apologies for those who have picked it up twice!)

So the iPhone really is here. (Actually, it’s been here since October 2005 at least, but more on that later). I was recently given an evaluation unit of the new Linksys CIT300 Dual-Mode Internet Telephony kit, a member of the Linksys iPhone line. The kit includes a combination landline / Skype cordless phone and a base unit that connects to your landline and a PC running Skype. I was excited to put this new gadget through some paces. I’m not a huge Skype user, but if this device truly integrates Skype with my existing home phone service, that has the potential to change my calling habits.

I’ll say right up front that I had an overall very positive experience with the CIT300. The phone is well designed, easy to use, and could go a long way towards bringing Skype into even more households. It does an admirable job of including key Skype functionality on a small handset, and handles its dual life as a landline cordless phone gracefully. But for several factors, the biggest being the inconsistent quality I experience with SkypeOut and SkypeIn calls, I’m not going to be switching to a device like this in my home anytime soon.

John Littler has written an excellent series of articles on the Linux-based Nokia 770 Internet tablet computer over on our ONLamp LinuxDevCenter, but his latest installment will be of particular interest to readers of this site as the 770 now supports SIP calling. VOIP on the Nokia 770 Internet Tablet covers how to upgrade the 770, make SIP calls, and even how to put Asterisk on the tablet computer!

Nokia’s 770 Internet Tablet is more than a Linux-based device; a recent software update made it a handy VOIP device. John Littler examples how the upgrade works and walks you through setting up VOIP and Asterisk.

Don’t miss this article if you’re interested in the Nokia 770 or interesting Asterisk implementations.

The mobile phone market doesn’t need yet another innovative device design. We’re well-served by RAZRs and Treos and Dash’s (oh my!). But it desperately needs innovation in the smartphone OS area.

Cross-listed from the OnJava blog, go to the original post.

Yesterday evening I heard a talk from Amazon about their new web services — well, new to me, at least, since I hadn’t been paying attention. I think “web services” is a minomer; what they’re actually providing is a group of web components that you can use to create your own web services.

Amazon provides a set of crucial services that let an aspiring designer create services with no upfront hardware cost. Need storage space online? $0.10/GB per month, $0.20/GB to transfer. Need a stack of CPUs? You can get a virtual Linux box from Amazon for $0.10/hour.

I’ve already dreamed up with a couple of telephony-based applications based on Amazon’s services. When storage is innfinitely deep and fast it becomes far easier to create recordings of telephone conferences. When I can create viritual servers on demand, I can have a hundred Asterisk servers running for an hour — total cost $10 — to blast out a few hundred thousand calls.

I have to give Amazon a lot of credit. They’ve disaggregated their business, separating out their knowledge of server and Web design from the rest of their corporate expertise, to create an entirely new business. And they’ve created disaggregated components that have already spawned dozens of new businesses.

In Subspace Networks: Hiding Circuit Switched Networks in a Packet Switched Network, Brian McConnell offers up another one of his thought-provoking ETel articles. Brian describes a trick that he calls “subspace networking”, which can be used to embed circuit switched networking within packet switched LAN hardware, while applications are unaware that this is going on behind the scenes.

I hate unsolicited pre-recorded telephone calls. I’ve filed complaints with the FCC against callers, I’ve started an open-source project called Stop Rude Calls to screen out automated calls, and I’ve even been known to threaten callers with a few thousand return calls from my own equipment if they didn’t stop bugging me.

On the other hand, I also make pre-recorded outbound calls — to opt-in subscribers who thank me for the service. Every year, for fifty days after the beginning of Passover, subscribers around the world receive calls from me every evening (it’s a Jewish thing, ok?). This year I hope to charge subscribers a nominal fee as part of a charity campaign drive, or perhaps provide the service for free but tack on a solicitation for charity at the end of the call.

But a new US Federal Trade Commission rule may put the kibosh on my service and on many other related services. The FTC just proposed a new rule that severly restricts pre-recorded calls. Even if there’s a pre-existing business relationship, the FTC rule virtually eliminates pre-recorded calls that include a soliciation.

Well, my first reaction was “Bravo!” I hate calls, pre-recorded or live, from businesses with whom I have a “pre-existing relationship” that consists of filling out a registration card or purchasing a hard drive; it’s gotten so bad that I’ve started the Disposable Phone Number project so I can give businesses a number but then toss the numbers away. But after my friends and business colleagues at Voxeo chimed in I had second thoughts. Voxeo sounded the alarm with this analysis, and after a conversation with their in-house lawyer (a very sharp cookie) and a personal letter from the company CEO addressing some of my questions, I’m convinced they have some very legitimate concerns: The proposed FTC rule could kill many promising Phone 2.0 services, including my very own outbound calling service.

Follow link below to rest of article…

I recently uncovered evidence that Investor’s Business Daily, the second-largest print and online newspaper devoted to financial information, had at least part of its subscriber list stolen.

Now I’ve become curious. I’ve heard about selective attacks against specific targets to harvest email addresses. I can’t recall ever hearing about a Trojan or virus that harvested phone numbers.

Certainly that day can’t be far off. Phone calls sometimes attract the attention of police and regulators; therefore it makes sense from a criminal’s point of view to make as few phone calls as possible and to direct the calls at lucrative targets. As a result criminals have even more incentive to gather lists of “high quality” phone numbers than they do to gather lists of email addresses.

But I don’t recall ever hearing of a Trojan or virus that targets phone numbers or snail-mail address; the most leading-edge thefts I’m aware of are directed at IM addresses. Now I’m wondering if I missed the memo, if I’m forecasting the furture, or if there are stealthy attacks in progress…

In the wake of the proposed Truth in Caller ID Act, a lengthy discussion of the issues around reliable Caller ID data recently took place on the excellent Voice Over IP Security Alliance VOIPSEC mailing list. While many points were debated, two things are clear: VoIP-based systems connecting to the PSTN have increased the unreliability of Caller ID data, and law enforcement agencies have legitimate needs to be able to get access to call record information that is becoming increasingly difficult to obtain.

Solving the Caller ID Problem is John Todd’s proposal for a vendor-neutral industry consortium as an alternative to potentially stifling legislation to address these concerns, and was adapted from a post that John made to the VOIPSEC list.

John points out that it is not always possible or obvious how to treat Caller ID data when moving to or from VoIP and the PSTN networks, and there are even business models which are predicated on the ability of caller ID to be transmitted to the PSTN with a value that is not “expected.”

Advanced combinations of transmitting Caller ID are part of the natural progression of next-generation services, and customers are demanding a better set of methods to control the way their company and personal calls are identified. It’s going to get much more complex and customized from the customer and service provider’s perspective, and it’s up to us as an industry to figure out how to provide accountability for our customers and ourselves to law enforcement agencies.

John has given this problem a lot of thought and has come up with a plan that I think is better than other proposed solutions and well worth serious consideration. If you have an interest in the future of Voice 2.0 companies and services, I think you’ll find John’s proposal well worth a read.

I may have re-invented the wheel: Tossable Digits uses the same extension-based scheme I proposed. They’ve got a different business model than I had in mind, they have a different security model than I had in mind, and they don’t seem to have some of the other usage-tracking features I’ve been thinking about — but on the other hand, they have a commercial service you can sign up for today, and all I have are doodles and a demo.

I wish them success, and if you want to use Disposable Phone Numbers immediately, they’re the place to go. If you end up using their service, feel free to comment here and let us all know how well it works.

As promised, the Disposable Phone Numbers™ project is now available as open source via Sourceforge.

To answer some other questions that have been asked: The equipment you need to run DPN is actually quite minimal. The demo box you reach when you call the demo telephone number (+1 312 967 6568) is a nice Wiindows XP machine, but nothing truly extraordinary. The Voxeo “Prophecy” server requires Windows XP (for now; Linux is coming “soon”) and if you use pre-recorded announcements instead of text-to-speech, you don’t really need much in the way of horsepower.

The demo box itself isn’t connected to the telephone network directly. The telephone number is supplied by Voxeo (again, as a freebie to developers) and the demo box uses SIP to accept the incoming call. I don’t bother routing the call to an actual telephone handset, so this demo box doesn’t even have an ATA.

In other words: if you’ve got a desktop machine with some spare disk space, you can set it up as a speech recognition/text-to-speech/web server/PHP/servlet/SIP server and run the DPN demo without any additional equipment.

Voxeo, BeVocal, and some other companies also have free developer accounts; you can run your application on their hosts, and connect to them via Free World Dialup or the regular phone network.

A friend recently asked me how many email addresses I had; her family thought she was a geek because she had six. I guess I’m a little more geeky: I have over one hundred eighty email addresses. Every time a company asks me for an email address, I generate a new one just for that company and arrange for incoming mail to show up in my normal inbox (I’ve automated this to a single command line in Linux). If the company abuses my email address by sending me spam, I disable the email address. Disposable email addresses guard my privacy.

I’d certainly like the same system for telephone numbers, because sometimes a company will get hold of my phone number and never let go. (Horror stories deleted; fill in your own.) Internet Telephony provides a solution. One reason Internet Telephony is a revolutionary innovation is that IT takes the ability to establish telephone numbers away from a central authority and gives it into the hands of anyone who wants to terminate calls. Telephone numbers become an unlimited commodity instead of a precious — and lucrative — resource.

Until IT becomes ubiquitous and all telephones can dial using the “sip:foo123@example.com” syntax, I’ve devised an alternative scheme to achieve disposable telephone numbers: use a single telephone line, but assign each company its own extension.

I’ve written a demo Disposable Phone Numbers application in CCXML, VoiceXML, and Java, running on Voxeo’s free Prophecy server. As each call comes in, the caller is asked for an extension number, which the caller provides using spoken words or touch tones. The extension goes to local Java servlet, which does a check of the extension number (it should be a database lookup, but my prototype just checks against a short internal list). If the extension is valid, the call goes through; if the extension is invalid, the application drops the call. If anyone is interested in hearing the pre-alpha — and the user interface is just barely good enough at the moment — they can try it by calling the demo number at +1 312 957 6568, which terminates at a computer on my desktop. At the moment, the only two valid extensions are 3350 and 8932, and all calls are dropped. If there’s enough interest I’ll release the application as open source.

In his latest ETel article, Matthew Gast explains how he set up his home Asterisk system to handle his out-of-office call processing needs. This detailed article includes the code Matthew used to allow and validate user input of the time and date for when out-of-office processing should be used.

One of my primary motivations for setting up my own Asterisk system was my travel schedule. I had an impossibly complex series of rules that I wanted people calling me to follow about when to use my home phone, when to use my cell phone, and how to avoid disturbing me when I was far from home. With Asterisk, I could set those rules down in code and allow callers to use one number to reach me. My initial project was to enable remote SIP extensions to keep track of the local time at my remote site and avoid bothering me at inappropriate times.

It used to be that building advanced telecom applications required a minimum investment of several million dollars, a barrier which kept small entrepreneurs from experimenting with telecom services in the way that web services developers have for years. With the development of open standards telephony, VoIP, and hosted on-demand computing services, it is finally possible to use a low-cost development track to create next generation telecom services. Brian McConnell examines this new landscape for telecom development in his latest ETel article Building Advanced Telecom Apps on a Shoestring.

Matthew Hamrick is the co-founder of the Silicon Valley Homebrew Mobile Phone Club, and while he appreciates the strides we’re seeing in openness in telecom with devices like Trolltech’s QTopia Greenphone, he makes a plea for a truly open mobile phone platform in this ETel op ed piece.

Let me pitch you on the idea I’m calling “the complete open phone.” The goal of the complete open phone is to provide developers a complete platform for innovation. Starting with the mobile phone hardware and ending with the wireless network services. I think we all know that a nice open source operating system is a key component. How ’bout Linux? Add to that the GNU tools and we’re well on our way. Just add a good idea and some talented software engineers and you’re half-way to changing the world, one handset at a time.

I recently saw a demo of a home-made Linux-based cell phone that was put together with inexpensive off-the-shelf parts at an O’Reilly planning session for our upcoming Emerging Telephony Conference, and I’d like to think this future that Matthew is hoping for is just about upon us. One thing that I know for sure is that the people who are pushing this envelope the hardest will be at next February’s ETel conference, and I’m already looking forward to it.

When it comes to open source telecom projects, Asterisk is by far the dominant player, but it isn’t the only kid on the team and FreeSWITCH is considered by many to be one of the up and comers. Read my recent interview with Anthony Minessale–the developer behind the FreeSWITCH open source telephony project–about the status of FreeSWITCH, how it compares to Asterisk, and Minessale’s future plans for FreeSWITCH.

In our latest Emerging Telephony article, Maciek Kaminski presents the Yet Another Telephony Engine (YATE) project. YATE is an ideal environment for rapidly prototyping telephony applications since it allows developers to write scripts in higher level languages while taking advantage of the performance of native libraries, without sacrificing too much efficiency. Maciek demonstrates the usefulness of YATE with several practical examples.

Dev2Dev has just published an excellent article by David Burke and Darragh O’Flanagan that describes an IMS application based on SIP servlets and VoiceXML. The article provides nice introductions to IMS and VoiceXML and then goes on to demonstrate how to build a full-featured Personal Assistant application that is consistent with the IMS architecture. Chock full of flow-charts, sample code, and an extensive resource list, this article looks like a great starting point for developers looking to learn about IMS, SIP Servlets, and VoiceXML.

Asterisk’s rich set of features and extensibility made it easy to create a module that incorporates the functionality of the iotum relevance engine. Much like the recent iotum/PhoneGnome deal, this new Asterisk module will help get iotum’s exciting technology into the hands of people who want to use it now. iotum developer Todd Jefferson dives under the hood and explains the inner workings of the iotum Asterisk module. If you’re interested in using or deploying iotum with an Asterisk-based system or creating other extension modules for Asterisk, check out Todd’s article Building the iotum Asterisk Module.

Televolution’s PhoneGnome is both an innovative product and platform, and they have recently partnered with iotum to offer the iotum Relevance Engine service to their users. Televolution understands the difficulties that exist in getting new products and services launched in the current telco-dominated telecom industry and is a strong advocate for making it easier to bring innovations to market. We’re proud to publish this article by Mark Petrovic that presents a case-study showing how iotum took advantage of the PhoneGnome LiteSig API that uses XML-RPC to easily integrate their product into the PhoneGnome platform.

I’m eager to point people to three new articles we’re publishing on ETel today by Brian McConnell on using information technology to help prepare for a possible pandemic. Besides plenty of good general advice for businesses, schools, and organizations on how to think about and prepare for such a crisis, one of the articles provides some very specific and actionable instructions on how an organization can cheaply set up a large-scale teleconferencing system using open source tools (asterisk and gizmo). The ability for companies do affordably do this on their own is relatively new, and I haven’t seen any other articles detailing it the way Brian does.

1. For SARS Press 1, for Bird Flu Press 2…
2. How to Implement Telecommuting in a Hurry
3. Building Your Own Teleconference System with Asterisk and Gizmo

Brian has been thinking about these issues for a long time, and it shows. The information he provides here could be critically important if the widely-feared avian flu pandemic ever materializes, but is also very practical for companies wanting to set up these kind of systems regardless of the motivation. We’ve tried hard not to be too alarmist about the topic, but since all the experts agree it’s not “if” but “when” we’ll see another flu pandemic, Brian points out its a good idea to do some thinking and planning about this possible scenario now.

The Washington Post published an article just today, Business Plan for a Pandemic?, noting that most firms haven’t prepared for the possibility of a global outbreak. Hopefully Brian’s articles will help in this regard. I think this is very important information that should get circulated as widely as possible. If you agree please help spread the word about these articles.

Last summer, I wrote that reliability should not always be a goal in network design, especially with applications like VoIP. “Better late than never” is true for a stream of bits you need to reproduce perfectly, but it’s not the case for a stream of bits you’re trying to deliver in less than 200 milliseconds. If something is late, it’s better to forget it and move on.

Well, as the saying goes, the difference between theory and practice is greater in practice than in theory. The always-interesting Joel Snyder recently tested VoIP call quality through a variety of VPNs for Network World. From the pure theoretical perspective, going through a VPN technology that ensures in-order delivery should offer no help to call quality, and would be expected to cause some problems, especially on low-reliability networks that have frequent loss and retransmission.

That’s not what he found. After finding that call quality improved when using SSL, he used a network analyzer to find out what was going on:

In every case, adding an SSL VPN to a VoIP call over a good broadband network improved call quality. So in effect, wrapping a VoIP call in SSL gives it more structure, kind of like the rind of good Brie. What we had not counted on was the huge difference between what VoIP requires (64Kbps) and a typical broadband connection of 500Kbps or more. Because the broadband connection was so fast, TCP was able to repair the impairments without reducing voice quality.

(I’ll disagree with Joel on the cheese analogy, because I hate Brie. Rind or not, I find it disgusting.)

To see the improvements, check out the pop-up graph about half-way through the article. It’s a graph that compares the mean opinion score (MOS), a measure of call quality, to an unencrypted reference for four test scenarios. Data points farther to the right are better.

The biggest surprise came from the test on the “bad” network (the orange data points in the picture). SSL VPNs delivered a signifcant performance improvement. On the graph, you can see that the orange data points are far to the right of the unencrypted reference line, and offer the widest gap between the reference line and the measured SSL-protected quality.

The best news of all our testing came when we set up the bad network, representing the lower end of quality of the broadband services. In this test, TCP and a high-speed network again came to the rescue. All but three of our SSL VPN vendors also improved the unacceptable call but took call quality up enough for the call to be considered acceptable. In these tests, we saw as much as a 45% to 50% improvement in call quality.

Interestingly, the call improvement is specific to SSL. Juniper’s product offers two operational modes, SSL (which is based on TCP, and therefore does re-order packets) and IPsec ESP (which is an IP datagram service, and does not). The SSL mode resulted in significantly better call quality as long as the network was providing decent service. In the last test, with a very poor quality network, the datagram services provided the best call quality, but it was still so bad as to be unusable.

I only have one minor technical nit to pick with the article. It states that a phone call requires 64 kbps. It requires 64 kbps of data payload throughput, but adding on the encapsulation headers for UDP and RTP pushes the required throughput to 80 kbps. (See Table 1 at the bottom of the first page in my VoIP capacity analysis for more codec details.) As the test shows, 100 kbps is not enough to sustain an 80 kbps stream with any kind of quality because it has no headroom.

The moral of the story is that quality of service through overprovisioning is not such a bad idea after all. The key appears to be having enough burst headroom to recover from the impairment and fill in the gap. When an error occurs, you need to be able to detect it and retransmit enough data quickly to fill in the gap and move on. At six times the required data rate (500 kbps), that’s easy. At 25% higher than the required data rate (100 kbps), it just isn’t possible.

To confirm that the headroom capacity is what matters, a great follow-up test would replace the 80 kbps G.711 codec with, say G.729, which only requires 24 kbps. At that point, the network would have four times the necessary capacity to transmit a voice call, so there is significant excess capacity that can be used to retransmit error recovery packets.

While the rest of the technology industry has marched along, fixed line and mobile operators have not updated their voice mail systems in well over a decade. These systems force you to call into an IVR system to listen to your messages in sequential order. A better way to do this is to push the voice message onto the destination device so the recipient does not need to phone in to fetch it.

A little known way to do this is MMS (multimedia messaging service), now widely supported on most carrier networks, especially GSM networks such as T-Mobile. If your device supports MMS, you can go to the messaging menu (sometimes the same menu as the SMS/text messages menu), create a SMS, and attach an audio clip. This is usually easier to do on smartphones such as Palm and Symbian devices.

When you send the message, the recipient’s phone will typically receive the message in its entirely and cache it for offline playback. This is a handy feature when you need to send a voice message to someone who you know uses text messaging a lot, or to someone who wanders in and out of coverage range. If you have a decent phone, it’s easy to use, and easier to deal with than typing. The phone manufacturers could do a little more work to make this a one-button operation, but this is already an improvement.

NOTE TO MOBILE OPERATORS: your IVR voice mail systems are very, very tired. You need to update these to push voice messages onto subscribers’ phones using MMS or email if they want this option. I know you want to capture another billable minute every time someone dials into the IVR, but it is a bad user experience, and besides you charge for MMS on a per message basis, so who cares what method the subscriber uses to fetch his or her voice messages.

I spend a _lot_ of time on the phone, usually several thousand minutes per month, although this is spread across several different devices and accounts. I have a mobile line, a POTS line, VoIP service from Broadvoice, and an 800 number.

Recently my travel patterns changed, and as a result, I found that I was spending a lot more time on my cellular phone than I was aware of. A rude surprise in the form of a $500 phone bill awoke me to this fact. I normally don’t even bother to read my phone bills if they are more or less in line with what I expected to pay.

I am generally not a miser with phone service. Time is money, and if I am not being gouged, I don’t have time to call “customer service” over a few dollars. However, I don’t like getting clocked with “overage” charges from my mobile provider. I am currently with T-Mobile, and am quite happy with them overall. They provide a good mix of services at reasonable prices, provided you don’t go over your monthly allotment (if you do, additional minutes cost $0.40/minute, which is pretty steep, although this is the norm in the US for cellular airtime).

Much of the time I spend on my mobile phone is spent at other clients’ offices, coffee shops and the like. I am usually stationary at these locations, and can use wi-fi combined with my Broadvoice VoIP account and XTEN VoIP client to make phone calls. Since I spend a lot of time at airports and other locations with paid wifi hotspots, I signed up for T-Mobile’s $19.95/month hotspot plan.

At first $20/month for yet another internet access plan seemed steep, but then I realized that if I moved a fraction of my cell calls to wifi, it would pay for itself with only 50 minutes of usage per month (50 minutes times 40 cents/min). I figured, conservatively, I spend several hours per month in airport lounges, otherwise dead time that is perfect for returning phone calls. Within a month, my cellular use dropped back below my 1400 minute cap on daytime usage, saving me at least $200 per month relative to recent bills, and making my $20 month spend on T-Mobile wi-fi look like a bargain.

Voice over Wi-Fi is not a replacement for cellular, but if you are a heavy phone user and frequently use the phone in stationary settings like airports, this is a viable strategy for keeping your phone bills under control. It isn’t free, but if you pair wifi with a flat-rate VoIP plan such as Broadvoice’s Unlimited World plan, you can call pretty much anywhere in the developed world for a capped monthly rate.

This is why I think cellular providers will be smart to embrace dual mode cellular+WiFi devices. The ideal package will be something like this. $50 per month gets you unlimited domestic and international calling to select countries via SIP, combined with a mobile phone that can latch onto WiFi networks. Whoa? $50 per month sounds steep. Yes, but not really because that bundle would include: flat-rate VoIP service that would normally cost $20-30 per month separately, license for a SIP client on the mobile phone, and perhaps access to the mobile operators own network of WiFi hotspots. Bonus points if this also included the option to use PC based softphones with the same SIP login credentials.

This is a smart strategy for two reasons. First, it will enable the telco to offer power users an attractive product where they would otherwise be forced to look at other options. Second, it is the nail in the coffin for POTS lines. If my T-Mobile phone would work with my home Wi-Fi service, especially if I could have a wired SIP phone linked to the same account, I could finally get rid of my increasingly less useful POTS service. I got rid of SBC DSL several years ago (thank God), and now would like to give my POTS service the heave-ho.

Fixed-mobile convergence was a big theme at this year’s GSM expo, so the major carriers are clearly thinking about this. What remains to be seen is if any of them will launch a product that is truly innovative with aggressive pricing, or if they will botch this by coming out with an overly complicated, overpriced and not very attractive product (as they’ve done with streaming video).

Metrix Pebble is a variant of the popular Pebble Linux distribution that adds a web-based configuration GUI, support for OLSRD and EVDO, automatic generation of network flows, wireless users, and radio link status graphs using rrdtool, and all fits in just under 64MB of flash memory. In this new article, Rob Flickenger describes how to use Metrix Pebble to set up a wireless mesh network.

I had the pleasure of working with Rob at O’Reilly a couple of years ago, and he really knows his stuff. He blew just about everyone away with his Pringles can WiFi antenna (his weblog entry about that project is the most most popular O’Reilly Network weblog post ever with nearly 2 million page views!). Rob is the author of the highly-acclaimed Building Wireless Community Networks and co-author of the second edition of Wireless Hacks. If you’re interested in wireless mesh networking, check out what Rob is up to. You’ll be glad you did.

The traditional PSTN is a complex creature, but it must be understood in order to successfully integrate with VoIP. Sean Walberg walks us through the world of traditional telephony, identifying the key terms, concepts, and acronyms along the way. You may have noticed that the field of telecommunication is really big on acronyms, and if you’ve ever wondered what a DS0, an FXS, or a DID is, Sean’s Telecom Terms and Concepts article is for you. If you believe the cliche that you have to know where you’ve been to know where you’re going, you’ll want to bookmark this clear, concise reference to basic telephony.

Voipster released their OpenZoep client-side VoIP engine as an open souce project licensed under the GPL at ETel, and they also released an OpenZoep Firefox extension. Read all about the OpenZoep project and the technology behind it in this new article by Erik van Eykelen. Erik details the APIs and provides resources for those who want to work with OpenZoep. (And if like me you’d been wondering about the pronunciation, Erik helps us out with that too — it’s like “open soup”).

Andrew Sheppard opines that the battle between VoIP and PSTN/POTS is over, and the battle for mobile telephony is about to begin. Andrew explains why he thinks the telcos are poised to fail and that mobility is the killer app for VoIP in The Next Battleground for VoIP. Andrew is the author of O’Reilly’s recently-released Skype Hacks.

Based on his work developing a stable wireless mesh platform that allows true peer-to-peer multi-hop network connectivity, Chris Ngan discusses some proof-of-concept applications that demonstrate the power of this network infrastructure and the ease with which text/chat, voice, and video applications can be made location-aware. Chris will discussing these concepts in more detail at the upcoming O’Reilly Emerging Telephony conference.

Andy Oram examines the new concept of a “webcaster’s right” that major Web portals are trying to introduce through a World Intellectual Property Organization treaty. The treaty would allow Web sites to control the dissemination of content they put up. Using the failed database protection laws as an example, and in the context of the carrier’s desire to create a tiered Internet, Andy analyzes this new threat to the public domain.

What would a webcasters right mean? It would mean you couldn’t retransmit content put up by someone else on the web without permission. The proposal tries to indicate that the restriction covers only images and sound, but it’s not clear that a line can be drawn between such content and other things, including text. At any rate, the idea of extending the broadcasters right to the Web is bizarre and fundamentally out of sync with how the Web works. The whole basis of the Web is making links; people don’t normally copy and retransmit material.

As usual, Andy’s perspective is insightful and inspiring. He’s included the full text of a letter he wrote to the U.S. delegates to WIPO on the webcasters right. Make sure and educate yourself on this important and little-known issue. Andy’s article is a great place to start.

Ed Stephenson recently spoke to ETel co-chair Surj Patel about why we think Internet Telehphony is important and what to expect at the upcoming O’Reilly Emerging Telephony conference, this January 24-26 in San Francisco. Developers now have the right tools and the right motivation to build a wide range of new desktop applications, telephone services, and corporate phone systems that integrate voice with the Web, IM, WiFi, and more.

It’s now as easy to create a voice application as it is to create a web application, and many of the same technologies are used in both areas. This creates tremendous opportunities for developers, enterprise users, and even the telcos — telephone companies — though most of them are loathe to believe it at the moment.

ETel will both showcase and examine the spirit of innovation behind VoIP communications. If you haven’t signed up yet, you just have until Monday 1/9 to take advantage of the Early Registration discount. And don’t forget, readers of this site can qualify for an additional 40% off that price by using the promo code etel06v40!

What do you get when you mix Asterisk with Ruby on Rails? How about a powerful and open platform for creating the next generation of voice applications. Joe Heitzeberg has developed the Ruby Asterisk Gateway Interface (RAGI), an open-source framework for bridging the Ruby on Rails web application server environment and Asterisk, the open-source PBX. According to Joe, the two fit together like chocolate and peanut butter.

Using RAGI, you can now easily and quickly develop voice applications that run with the power of Asterisk underneatht the hood. Check out Joe’s article for a taste of this Voice 2.0 peanut butter cup, and if it leaves you craving more be sure to attend his Ruby on Rails with Asterisk workshop at our upcoming Emerging Telephony Conference.

I posted a teaser of this on Wednesday, but here’s the full text of my interview with Norman Lewis, the director of reseach for France Telecom’s Home office and a keynote speaker at our upcoming Emerging Telephony conference. Norman has a lot to say about the future of the VoIP industry, the role that telcos will play, and he clearly understands that paradigms are rapidly changing in the world of communications. He’s also been actively involved in working on issues related to the Digital Divide. I’m really looking forward to his ETel keynote, Telco Is Dead–Long Live the Communications Company.

Remember, readers of this site can register for the Emerging Telephony conference in San Francisco, January 24-26, and get a 40% discount! Just use the discount code etel06v40 when you sign up.

I always enjoy O’Reilly editor Andy Oram’s take on things, he’s not afraid to express an opinion or delve into the social, cultural, and political aspects of the technologies he’s watching. His latest article is no exception and his wide-ranging report from this year’s VON conference covers a lot of ground as he analyzes the latest trends and issues in the VoIP industry. From new products to security issues to what the politicos are (and aren’t) saying about the future of VoIP, there’s something for everyone. Andy notices that the enterprise is still where the action is in North America in VoIP is All Business at VON.

Matthew Gast delves into the mathematics of capacity calculations in his latest O’Reilly article How Many Voice Callers Fit on the Head of an Access Point? If you’re engineering a VoIP network you should take a look at Matthew’s analysis of the theoretical maximum capacity of access points to carry VoIP traffic over 802.11a, b, and g. Matthew is the author of 802.11 Wireless Networks: The Definitive Guide.

Over the summer, I came across a capacity calculation in the manual for a Cisco voice over IP phone detailing the number of simultaneous calls that can be supported on an access point. Intrigued, I extended the analysis. Voice and data on wireless LANs require opposing preconditions for good performance. High-quality voice requires that frames containing voice data can be transmitted very quickly after arrival, and they need to be transmitted on a very regular schedule with tight timing requirements. Good data throughput comes from stuffing the transmission queue as full as possible. Individual frames might suffer long delays, but the overall capacity is high. Voice quality is often very sensitive to network load.

Say you want to route interoffice calls over your company’s wide area network instead of via conventional telephone lines or point-to-point leased lines. Your telephone system is an outdated heap of junk that was designed circa 1970. How do you trick it into routing interoffice calls via VoIP? Brian McConnell shows you exactly how to achieve this bit of VoIP wonder in his recent O’Reilly Network weblog Telecom Recipes - Upgrading Old Telephone Systems To Use VoIP For Inter-Office Calls.

Probably old hat for readers of this site, but if you’re looking for something to point your boss at to explain all this VoIP stuff, Ted Wallingford has written an excellent summary in his O’Reilly Network article, What is VoIP?

Ted covers the different technologies that make up VoIP, as well as looks at some of the reasons for it’s growing popularity and where the technoloy is headed. Ted knows of what he speaks, he’s the author of Switching to VoIP, which has been getting rave reviews as a hands-on practical guide for telecom professionals making the switch.

Voice over Internet Protocol (VoIP) is the family of technologies that allow IP networks to be used for voice applications, such as telephony, voice instant messaging, and teleconferencing. VoIP entails solutions at almost every layer of an IP network–from specialized voice applications (like Skype) all the way down to low-level quality measures that keep those applications running smoothly.