First they ignore you, then they laugh at you, then they fight you, then you win.
We have arrived at the final chapter of this book. We’ve covered a lot, but we hope that you now realize that we have barely begun to scratch the surface of this phenomenon called Asterisk. To wrap things up, we want to spend some time exploring what we might see from Asterisk and open source telephony in the near future.
While prognostication is always a thankless task, we are confident in asserting that open source communications engines such as Asterisk herald a shift in thinking that will transform the telecommunications industry. In this chapter, we will discuss some of our reasons for this belief.
Although Alexander Graham Bell is most famously remembered as the father of the telephone, the reality is that during the latter half of the 1800s, dozens of minds were working toward the goal of carrying voice over telegraph lines. These people were mostly business-minded folks, looking to create a product through which they might make their fortunes.
We have come to think of traditional telephone companies as monopolies, but this was not true in their early days. The early history of telephone service took place in a very competitive environment, with new companies springing up all over the world, often with little or no respect for the patents they might be violating. Many famous monopolies got their start through the waging (and winning) of patent wars.
It’s interesting to contrast the history of the telephone with the history of Linux and the Internet. While the telephone was created as a commercial exercise, and the telecom industry was forged through lawsuits and corporate takeovers, Linux and the Internet arose out of the academic community, which has always valued the sharing of knowledge over profit.
The cultural differences are obvious. Telecommunications technologies tend to be closed, confusing, and expensive, while networking technologies are generally open, well-documented, and competitive.
If one compares the culture of the telecommunications industry to that of the Internet, it is sometimes difficult to believe the two are related. The Internet was designed by enthusiasts, whereas contributing to the development of the PSTN is impossible for any individual to contemplate. This is an exclusive club; membership is not open to just anyone.
The International Telecommunication Union (ITU) clearly exhibits this type of closed thinking. If you want access to its knowledge, you have to be prepared to pay for it. Membership requires proof of your qualifications, and you will be expected to pay thousands of dollars to gain access to its library of publications.
Although the ITU is the United Nations’s sanctioned body responsible for international telecommunications, many of the VoIP protocols (SIP, MGCP, RTP, STUN) come not from the hallowed halls of the ITU, but rather from the IETF (which publishes all of its standards free to all, and allows anyone to submit an Internet Draft for consideration).
Open protocols such as SIP may have a tactical advantage over ITU protocols, such as H.323, due to the ease with which one can obtain them. Although H.323 is widely deployed by carriers as a VoIP protocol in the backbone, it is much more difficult to find H.323-based endpoints; newer products are far more likely to support SIP.
The success of the IETF’s open approach has not gone unnoticed by the mighty ITU. It has recently become possible to download up to three documents free of charge from the ITU web site.Openness is clearly on its minds. Recent statements by the ITU suggest that there is a desire to achieve “Greater participation in ITU by civil society and the academic world.” Mr. Houlin Zhao, the ITU’s Director of the Telecommunication Standardization Bureau (TSB), believes that “ITU should take some steps to encourage this.”
The roadmap to achieving this openness is unclear, but the ITU is beginning to realize the inevitable.
As for Asterisk, it embraces both the past and the future: H.323 support is available, although the community has for the most part shunned H.323 in favor of the IETF protocol SIP and the darling of the Asterisk community, IAX.
One of the oddest things about all of the standards that exist in the world of legacy telecommunications is the various manufacturers’ seeming inability to implement them consistently. Each manufacturer desires a total monopoly, so the concept of interoperability tends to take a back seat to being first to market with a creative new idea.
The ISDN protocols are a classic example of this. Deployment of ISDN was (and in many ways still is) a painful and expensive proposition, as each manufacturer decided to implement it in a slightly different way. ISDN could very well have helped to usher in a massive public data network, 10 years before the Internet. Unfortunately, due to its cost, complexity, and compatibility issues, ISDN never delivered much more than voice, with the occasional video or data connection for those willing to pay. ISDN is quite common (especially in Europe, and in North America in larger PBX implementations), but it is not delivering anywhere near the capabilities that were envisioned for it.
As VoIP becomes more and more ubiquitous, the need for ISDN will disappear.
It can take months, or sometimes years, for the big guys to admit to a trend, let alone release a product that is compatible with it. It seems that before a new technology can be embraced, it must be analyzed to death, and then it must pass successfully through various layers of bureaucracy before it is even scheduled into the development cycle. Months or even years must pass before any useful product can be expected. When those products are finally released, they are often based on hardware that is obsolete; they also tend to be expensive and to offer no more than a minimal feature set.
These slow release cycles simply don’t work in today’s world of business communications. On the Internet, new ideas can take root in a matter of weeks and become viable in extremely short periods of time. Since every other technology must adapt to these changes, so too must telecommunications.
Open source development is inherently better able to adapt to rapid technological change, which gives it an enormous competitive advantage.
The spectacular crash of the telecom industry may have been caused in large part by an inability to change. Perhaps that continued inability is why recovery has been so slow. Now, there is no choice: change, or cease to be. Community-driven technologies such as Asterisk will see to that.
Traditional telecommunications companies have lost touch with their customers. While the concept of adding functionality beyond the basic telephone is well understood, the idea that the user should be the one defining this functionality is not.
Nowadays, people have nearly limitless flexibility in every other form of communication. They simply cannot understand why telecommunications cannot be delivered as flexibly as the industry has been promising for so many years. The concept of flexibility is not familiar to the telecom industry, and very well might not be until open source products such as Asterisk begin to transform the fundamental nature of the industry. This is a revolution similar to the one Linux and the Internet willingly started more than 10 years ago (and IBM unwittingly started with the PC, 15 years before that). What is this revolution? The commoditization of telephony hardware and software, enabling a proliferation of tailor-made telecommunications systems.
In “Paradigm Shift” (http://tim.oreilly.com/articles/paradigmshift_0504.html), Tim O’Reilly talks about a paradigm shift that is occurring in the way technology (both hardware and software) is delivered. O’Reilly identifies three trends: the commoditization of software, network-enabled collaboration, and software customizability (software as a service). These three concepts provide evidence to suggest that open source telephony is an idea whose time has come.
Every good work of software starts by scratching a developer’s personal itch.
In his book The Cathedral and the Bazaar (O’Reilly), Eric S. Raymond explains that “Given enough eyeballs, all bugs are shallow.” The reason open source software development produces such consistent quality is simple: crap can’t hide.
In this era of custom database and web site development, people are not only tired of hearing that their telephone system “can’t do that,” they quite frankly just don’t believe it. The creative needs of the customers, coupled with the limitations of the technology, have spawned a type of creativity born of necessity: telecom engineers are like contestants in an episode of Junkyard Wars, trying to create functional devices out of a pile of mismatched components.
The development methodology of a proprietary telephone system dictates that it will have a huge number of features, and that the number of features will in large part determine the price. Manufacturers will tell you that their products give you hundreds of features, but if you only need five of them, who cares? Worse, if there’s one missing feature you really can’t do without, the value of that system will be diluted by the fact that it can’t completely address your needs.
The fact that a customer might only need 5 out of 500 features is ignored, and that customer’s desire to have 5 unavailable features that address the needs of his business is dismissed as unreasonable. Until flexibility becomes standard, telecom will remain stuck in the last century—all the VoIP in the world notwithstanding.
Asterisk addresses that problem directly and solves it in a way that few other telecom systems can. This is extremely disruptive technology, in large part because it is based on concepts that have been proven time and time again: “the closed-source world cannot win an evolutionary arms race with open-source communities that can put orders of magnitude more skilled time into a problem.”
One of the stumbling blocks of the traditional telecommunications industry has been its apparent refusal to cooperate with itself. The big telecommunications giants have all been around for over a hundred years. The concept of closed, proprietary systems is so ingrained in their culture that even their attempts at standards compliancy are tainted by their desire to get the jump on the competition, by adding that one feature that no one else supports. For an example of this thinking, one simply has to look at the VoIP products being offered by the telecom industry today. While they claim standards compliance, the thought that you would actually expect to be able to connect a Cisco phone to a Nortel switch, or that an Avaya voicemail system could be integrated via IP to a Siemens PBX, is not one that bears discussing.
In the computer industry, things are different. Twenty years ago, if you bought an IBM server, you needed an IBM network and IBM terminals to talk to it. Now, that IBM server is likely to interconnect to Dell terminals though a Cisco network (and run Linux, of all things). Anyone can easily think of thousands of variations on this theme. If any one of these companies were to suggest that we could only use their products with whatever they told us, they would be laughed out of business.
The telecommunications industry is facing the same changes, but it’s in no hurry to accept them. Asterisk, on the other hand, is in a big hurry to not only accept change, but embrace it.
Cisco, Nortel, Avaya, and Polycom IP phones (to name just a few) have all been successfully connected to Asterisk systems. There is no other PBX in the world today that can make this claim. None.
Openness is the power of Asterisk.
In the past few years, it has become clear that standards evolve at such a rapid pace that to keep up with them requires an ability to quickly respond to emerging technology trends. Asterisk, by virtue of being an open source, community-driven development effort, is uniquely suited to the kind of rapid development that standards compliance demands.
Asterisk does not focus on cost-benefit analysis or market research. It evolves in response to whatever the community finds exciting—or necessary.
After Mark Spencer attended his first SIP Interoperability Test (SIPIT) event, he had a rudimentary but working SIP stack for Asterisk coded within a few days. This was before SIP had emerged as the protocol of choice in the VoIP world, but he saw its value and momentum and ensured that Asterisk would be ready.
This kind of foresight and flexibility is typical in an open source development community (and very unusual in a large corporation).
The Asterisk-users list receives many email messages per day. More than 10,000 people are subscribed to it. This kind of community support is unheard of in the world of proprietary telecommunications, while in the open source world it is commonplace.
The very first AstriCon event was expected to attract 100 participants. Nearly 500 showed up (far more wanted to but couldn’t attend). This kind of community support virtually guarantees the success of an open source effort.
So what sorts of things can be built using Asterisk? Let’s look at some of the things we’ve come up with.
Asterisk can be used as a fantastic bridge between an old PBX and the future. You can place it in front of the PBX as a gateway (and migrate users off the PBX as needs dictate), or you can put it behind the PBX as a peripheral application server. You can even do both at the same time, as shown in Figure 15-1.
Here are some of the options you can implement:
Companies that have spent vast sums of money in the past few years buying proprietary PBX equipment want a way out of proprietary jail, but they can’t stomach the thought of throwing away all of their otherwise functioning equipment. No problem—Asterisk can solve all kinds of dilemmas, from replacing a voicemail system to providing a way to add IP-based users beyond the nominal capacity of the system.
Provide the PBX a list of numbers where you can be reached, and it will ring them all whenever a call to your DID (Direct Inward Dialing, a.k.a. phone number) arrives. Figure 15-2 illustrates this technology.
If a legacy telephony connection from an Asterisk PBX to an old PBX can be established, Asterisk can provide access to VoIP services, while the old PBX continues to connect to the outside world as it always has. As a gateway, Asterisk simply needs to emulate the functions of the PSTN, and the old PBX won’t know that anything has changed. Figure 15-3 shows how you can use Asterisk to VoIP-enable a legacy PBX.
Many people confuse the term “Interactive Voice Response,” or IVR, with the Automated Attendant (AA). Since the Automated Attendant was the very first thing IVR was used for, this is understandable. Nevertheless, to the telecom industry, the term IVR represents far more than an AA. An AA generally does little more than present a way for callers to be transferred to extensions, and it is built into most proprietary voicemail systems—but IVR can be so much more.
IVR systems are generally very expensive not only to purchase, but also to configure. A custom IVR system will usually require connectivity to an external database or application. Asterisk is arguably the perfect IVR, as it embraces the concepts of connectivity to databases and applications at its deepest level.
Here are a few examples of relatively simple IVRs that an Asterisk system could be used to create:
Using the Internet, you can obtain text-based weather reports from around the world in a myriad of ways. Capturing these reports and running them through a purpose-built parser (Perl would probably eat this up) would allow the information to be available to the dialplan. Asterisk’s sound library already contains all of the required prompts, so it would not be an onerous task to produce an interactive menu to play current forecasts for anywhere in the world.
Ed Guy (the architect of Pulver’s FWD network) did a presentation at AstriCon 2004 in which he talked about a little math program he’d cooked up for his daughter to use. The program took him no more than an hour to write. What it did was present her with a number of math questions, the answers to which she keyed into the telephone. When all the questions were tabulated, the system presented her with her score. This extremely simple Asterisk application would cost tens of thousands of dollars to implement on any closed PBX platform, assuming it could be done at all. See Chapter 9 for further details. As is so often the case, things that are simple for Asterisk would be either impossible or massively expensive with any other IVR system.
The cost of a proprietary IVR system is such that when a company with many small retail locations wants to provide IVR, it is forced to transfer callers to a central server to process the transactions. With Asterisk, it becomes possible to distribute the application to each node and, thus, handle the requests locally. Literally thousands of little Asterisk systems deployed at retail locations across the world could serve up IVR functionality in a way that would be impossible to achieve with any other system. No more long-distance transfers to a central IVR server, no more huge trunking facility dedicated to the task—more power with less expense.
These are three rather simple examples of the potential of Asterisk.
This little gem is going to end up being one of the killer functions of Asterisk. In the Asterisk community, everyone finds themselves using conference rooms more and more, for purposes such as these:
Small companies need an easy way for business partners to get together for a chat.
Sales teams have a meeting once per week where everyone can dial in from wherever they are.
Development teams designate a common place and time to update one another on progress.
Asterisk is still too much of an über-geek’s tool to be able to serve in the average home, but with no more than average Linux and Asterisk skills, the following things become plausible:
Parents who want to check up on the babysitter (or the kids home alone) could dial an extension context protected by a password. Once authenticated, a two-way audio connection would be created to all the IP phones in the house, allowing Mom and Dad to listen for trouble. Creepy? Yes. But an interesting concept nonetheless.
Going out for the night? Don’t want the babysitter tying up the phone? No problem! A simple tweak to the dialplan, and the only calls that can be made are to 911, your cell phone, and the pizza parlor. Any other call attempt will get the recording “We are paying you to babysit our kids, not make personal calls.”
Pretty evil, huh?
You get a call while on vacation that your Mom wants to borrow some cooking utensils. She forgot her key and is standing in front of the house shivering. Piece of cake; a call to your Asterisk system, a quick digit string into the context you created for the purpose, and your alarm system is instructed to disable the alarm for 15 minutes. Mom better get her stuff and get out quick, though, or the cops’ll be showing up!
How about allocating a specific phone-time limit to your teenagers? To use the phone, they have to enter their access codes. They can earn extra minutes by doing chores, scoring all As, dumping that annoying bum with the bad haircut—you get the idea. Once they’ve used up their minutes... click... you get your phone back.
Incoming calls can be managed as well, via Caller ID. “Donny, this is Suzy’s father. She is no longer interested in seeing you, as she has decided to raise her standards a bit. Also, you should consider getting a haircut.”
We’ve come to love the Internet, both because it is so rich in content and inexpensive, and, perhaps more importantly, because it allows us to define how we communicate. As its ability to carry richer forms of media advances, we’ll find ourselves using it more and more. Once Internet voice delivers quality that rivals (or betters) the capabilities of the PSTN, the phone company had better look for another line of business. The PSTN will cease to exist and become little more than one more communications protocol that the Internet happily carries for us. As with most of the rest of the Internet, open source technologies will lead this transformation.
The dream of having our technical inventions talk to us is older than the telephone itself. Each new advance in technology spurs a new wave of eager experimentation. Generally, results never quite meet expectations, possibly because as soon as a machine says something that sounds intelligent, most people assume that it is intelligent.
People who program and maintain computers realize their limitations and, thus, tend to allow for their weaknesses. Everybody else just expects their computers and software to work. The amount of thinking a user must do to interact with a computer is often inversely proportional to the amount of thinking the design team did. Simple interfaces belie complex design decisions.
The challenge, therefore, is to design a system that has anticipated the most common desires of its users, and that can also adroitly handle unexpected challenges.
For Asterisk, an obvious value of text-to-speech might be the ability to have your telephone system read your emails back to you. Of course, if you’ve noticed the somewhat poor grammar, punctuation, and spelling typically found in email messages these days, you can perhaps appreciate the challenges this poses.
One cannot help but wonder if the emergence of text-to-speech will inspire a new generation of people dedicated to proper writing. Seeing spelling and punctuation errors on the screen is frustrating enough; having to hear a computer speak such things will require a level of Zazen that few possess.
If text-to-speech is rocket science, speech recognition is science fiction.
Speech recognition can actually work very well, but unfortunately this is generally true only if you provide it with the right conditions—and the right conditions are not those found on a telephone network. Even a perfect PSTN connection is considered to be at the lowest acceptable limit for accurate speech recognition. Add in compressed and lossy VoIP connections, or a cell phone, and you will discover far more limitations than uses.
Asterisk now has an entire speech API so that outside companies (or even open source projects) can tie their speech recognition engines into Asterisk. One company that has done this is LumenVox. By using its speech recognition engine along with Asterisk, you can make voice-driven menus and IVR systems in record time! For more information, see http://www.lumenvox.com.
As we gain access to more and more bandwidth, it becomes less and less easy to understand why we still use low-fidelity codecs. Many people do not realize that Skype uses a higher fidelity than a telephone; it’s a large part of the reason why Skype has a reputation for sounding so good.
If you were ever to phone CNN, wouldn’t you love to hear James Earl Jones’s mellifluous voice saying “This is CNN,” instead of some tinny electronic recording? And if you think Allison Smith sounds good through the phone, you should hear her in person!
In the future, we will expect, and get, high-fidelity voice though our communications equipment.
Beginning in Asterisk 1.4, there is limited support for the G.722 codec. As more and more hardware vendors start building support for high-fidelity voice into their VoIP hardware, you’ll see more support in Asterisk for making better-than PSTN quality calls.
While most of this book focuses on audio, video is also supported in many ways within Asterisk. Video support is not complete, however. The problem is not so much one of functionality as it is one of bandwidth and processing power. More significantly, it is not yet important enough to the community to merit the attention it needs.
The concept of video-conferencing has been around since the invention of the cathode ray tube. The telecom industry has been promising a video-conferencing device in every home for decades.
As with so many other communications technologies, if you have video-conferencing in your house, you are probably running it over the Internet, with a simple, inexpensive webcam. Still, it seems that people see video-conferencing as a bit gimmicky. Yes, you can see the person you’re talking to, but there’s something missing.
Video-conferencing promises a richer communications experience than the telephone. Rather than hearing a disembodied voice, the nuances of speech that come from eye-to-eye communication are possible.
There are some challenges to overcome, though, and not all of them are technical.
Consider this: using a plain telephone, people working from their home offices can have business conversations, unshowered, in their underwear, feet on the desk, coffee in hand. A similar video conversation would require half an hour of grooming to prepare for, and couldn’t happen in the kitchen, on the patio, or... well, you get the idea.
Also, the promise of eye-to-eye communication over video will never happen as long as the focal points of the participants are not in line with the cameras. If you look at the camera, your audience will see you looking at them, but you won’t see them. If you look at your screen to see whom you are talking to, the camera will show you looking down at something—not at your audience. That looks impersonal. Perhaps if a videophone could be designed like a TelePromptR, where the camera was behind the screen, it wouldn’t feel so unnatural. As it stands, there’s something psychological that’s missing. Video ends up being a gimmick.
Wi-Fi is going to be the office mobility solution for VoIP phones. This technology is already quite mature. The biggest hurdle is the cost of handsets, which can be expected to improve as competitive pressure from around the world drives down prices.
This is a term that has been hyped by the telecom industry for years, but adoption has been far slower than predicted.
Unified Messaging is the concept of tying voice and text-messaging systems into one. With Asterisk, the two don’t need to be artificially combined, as Asterisk already treats them the same way.
Just by examining the terms, unified and messaging, we can see that the integration of email and voicemail must be merely the beginning; Unified Messaging needs to do a lot more than just that if it is to deserve its name.
Perhaps we need to define “messaging” as communication that does not occur in real time. In other words, when you send a message, you expect that the reply may take moments, minutes, hours, or even days to arrive. You compose what you wish to say, and your audience is expected to compose a reply.
Contrast this with conversing, which happens in real time. When you talk to someone on a telephone connection, you expect no more than a few seconds’ delay before the response arrives.
In 2002, Tim O’Reilly delivered a speech titled “Watching the Alpha Geeks: OS X and the Next Big Thing” (http://www.macdevcenter.com/pub/a/mac/2002/05/14/oreilly_wwdc_keynote.html), in which he talked about someone piping IRC through a text-to-speech engine. One could imagine doing the reverse as well, allowing us to join an IRC or instant messaging chat over our Wi-Fi phone, our Asterisk PBX providing the speech-to-text-to-speech translations.
As monopoly networks such as the PSTN give way to community-based networks like the Internet, there will be a period of time where it is necessary to interconnect the two. While the traditional providers would prefer that the existing model be carried into the new paradigm, it is increasingly likely that telephone calls will become little more than another application the Internet happily carries.
But a challenge remains: how to manage the telephone numbering plan with which we are all familiar and comfortable?
The ITU defined a numbering plan in its E.164 specification. If you’ve used a telephone to make a call across the PSTN, you can confidently state that you are familiar with the concept of E.164 numbering. Prior to the advent of publicly available VoIP, nobody cared about E.164 except the telephone companies—nobody needed to.
Now that calls are hopping from PSTN to Internet to who-knows-what, some consideration must be given to E.164.
While the concept of ENUM is sound, it requires cooperation from the telecom industry to achieve success. However, cooperation is not what the telecom industry is famous for and, thus, far ENUM has foundered.
The folks at e164.org are trying to contribute to the success of ENUM. You can log on to this site, register your phone number, and inform the system of alternative methods of communicating with you. This means that someone who knows your phone number can connect a VoIP call to you, as the e164.org DNS zone will provide the IP addressing and protocol information needed to connect to your location.
As more and more people publish VoIP connectivity information, fewer and fewer calls will be connected through the PSTN.
Distributed Universal Number Discovery (DUNDi) is an open routing protocol designed to maintain dynamic telecom routing tables between compatible systems (see Chapter 14 for more information). While Asterisk is currently the only PBX to support DUNDi, the openness of the standard ensures that anyone can implement it.
DUNDi has huge potential, but it is very much in its infancy. This is the one to watch.
As is true with any worthwhile thing, Asterisk will face challenges. Let’s take a glance at what some of them may be.
These days, the Internet is changing so fast, and offers so much diverse content, that it is impossible for even the most attentive geek to keep on top of it all. While this is as it should be, it also means that an enormous amount of technology churn is an inevitable part of keeping any communications system current.
Yes, it’s coming. There will always be people who believe they have the right to inconvenience and harass others in their pursuit of money. Efforts are under way to try and address this, but only time will tell how efficacious they will be.
The industry is making the transition from ignorance to laughter. If Gandhi is correct, we can expect the fight to begin soon.
As their revenue streams become increasingly threatened by open source telephony, the traditional industry players are certain to mount a fear campaign, in hopes of undermining the revolution.
There is a rumor making the rounds that the major network providers will begin to artificially cripple VoIP traffic by tagging and prioritizing the traffic of their premium VoIP services and, worse, detecting and bumping any VoIP traffic generated by services not approved by them.
Some of this is already taking place, with service providers blocking traffic of certain types through their networks, ostensibly due to some public service being rendered (such as blocking popular file-sharing services to protect us from piracy). In the United States, the FCC has taken a clear stand on the matter and fined companies that engage in such practices. In the rest of the world, regulatory bodies are not always as accepting of VoIP.
What seems clear is that the community and the network will find ways around blockages, just as they always have.
The recently departed Chairman of the United States Federal Communications Commission, Michael Powell, delivered a gift that may well have altered the path of the VoIP revolution. Rather than attempting to regulate VoIP as a telecom service, he has championed the concept that VoIP represents an entirely new way of communicating and requires its own regulatory space in which to evolve.
VoIP will become regulated, but not everywhere as a telephony service. Some of the regulations that may be created include:
One of the characteristics of a traditional PSTN circuit is that it is always in the same location. This is very helpful to emergency services, as they can pinpoint the location of a caller by identifying the address of the circuit from which the call was placed. The proliferation of cell phones has made this much more difficult to achieve, since a cell phone does not have a known address. A cell phone can be plugged into any network and can register to any server. If the phone does not identify its physical location, an emergency call from it will provide no clue as to the where the caller is. VoIP creates similar challenges.
Law enforcement agencies have always been able to obtain wiretaps on traditional circuit-switched telephone lines. While regulations are being enacted that are designed to achieve the same end on the network, the technical challenge of delivering this functionality will probably never be completely solved. People value their privacy, and the more governments want to stifle it, the more effort will be put toward maintaining it.
These practices are already being seen in the U.S., with fines being levied against network providers who attempt to filter traffic based on content.
When it comes to regulation, Asterisk is both a saint and a devil: a saint because it feeds the poor, and a devil because it empowers the phrackers and spammers like nothing ever has. The regulation of open source telephony may in part be determined by how well the community regulates itself. Concepts such as DUNDi, which incorporate anti-spam processes, are an excellent start. On the other hand, concepts such as Caller ID-spoofing are ripe with opportunities for abuse.
Due to the best-effort reality of the TCP/IP-based Internet, it is not yet known how well increasing realtime VoIP traffic will affect overall network performance. Currently, there is so much excess bandwidth in the backbone that best-effort delivery is generally quite good indeed. Still, it has been proven time and time again that whenever we are provided with more bandwidth, we figure out a way to use it up. The 1 MB DSL connection undreamt of five years ago is now barely adequate.
Perhaps a corollary of Moore’s Law will apply to network bandwidth. QoS may become moot, due to the network’s ability to deliver adequate performance without any special processing. Organizations that require higher levels of reliability may elect to pay a premium for a higher grade of service. Perhaps the era of paying by the minute for long-distance connections will give way to paying by the millisecond for guaranteed low latency, or by the percentage point for reduced packet loss. Premium services will offer the five-nines reliability the traditional telecom companies have always touted as their advantage over VoIP.
Open systems require new approaches toward solution design. Just because the hardware and software are cheap doesn’t mean the solution will be. Asterisk does not come out of the box ready to run; it has to be designed and built, and then maintained. While the base software is free, and the hardware costs will be based on commodity pricing, it is fair to say that the configuration costs for a highly customized system will be a sizeable part of the solution costs—in many cases, because of its high degree of complexity and configurability, more than would be expected with a traditional PBX.
The rule of thumb is generally considered to be something like this: if it can be done in the dialplan, the system design will be roughly the same as for any similarly featured traditional PBX. Beyond that, only experience will allow one to accurately estimate the time required to build a system.
There is much to learn.
Open source telephony creates limitless opportunities. Here are some of the more compelling ones.
Some people would tell you that price is the key, but we believe that the real reason Asterisk will succeed is because it is now possible to build a telephone system as one would a web site: with complete, total customization of each and every facet of the system. Customers have wanted this for years. Only Asterisk can deliver.
Anyone can contribute to the future of communicating. It is now possible for someone with an old $200 PC to develop a communications system that has intelligence to rival the most expensive proprietary systems. Granted, the hardware would not be production-ready, but there is no reason the software couldn’t be. This is one of the reasons why closed systems will have a hard time competing. The sheer number of people who have access to the required equipment is impossible to equal in a closed shop.
The design of a PBX was always a kind of art form, but before Asterisk, the art lay in finding creative ways to overcome the limitations of the technology. With limitless technology, those same creative skills can now be properly applied to the task of completely answering the needs of the customer. Open source telephony engines such as Asterisk will enable this. Telecom designers will dance for joy, as their considerable creative skills will now actually serve the needs of their customers, rather than be focused on managing kludge.
Ultimately, the promise of open source comes to nothing if it cannot fulfill the need people have to solve problems. The closed industries lost sight of the customer and tried to fit the customer to the product.
Open source telephony brings voice communications in line with other information technologies. It is finally possible to properly begin the task of integrating email, voice, video, and anything else we might conceive of over flexible transport networks (whether wired or wireless), in response to the needs of the user, not the whims of monopolies.
Welcome to the future of telecom!
 Contrast this with the IETF’s membership page, which states: “The IETF is not a membership organization (no cards, no dues, no secret handshakes :-)... It is open to any interested individual... Welcome to the IETF.” Talk about community!
 Considering the thousands of documents available, and the fact that each document generally contains references to dozens more, the value of this free information is difficult to judge.
 Much of the following section is merely our interpretation of O’Reilly’s article. To get the full gist of these ideas, the full read is highly recommended.
 From the perspective of the closed-source industry, this attitude is understandable. In his book The Mythical Man-Month: Essays on Software Engineering (Addison-Wesley), Fred Brooks opined that “the complexity and communication costs of a project rise with the square of the number of developers, while work done only rises linearly.” Without a community-based development methodology, it is very difficult to deliver products that at best are little more than incremental improvements over their predecessors, and at worst are merely collections of patches.
 Eric S. Raymond, The Cathedral and the Bazaar.
 Gordon Moore wrote a paper in 1965 that predicted the doubling of transistors on a processor every few years.
 This term refers to 99.999%, which is touted as the reliability of traditional telecom networks. Achieving five nines requires that service interruptions for an entire year total no more than 5 minutes and 15 seconds. Many people believe that VoIP will need to achieve this level of reliability before it can be expected to fully replace the PSTN. Many other people believe that the PSTN doesn’t even come close to five-nines reliability. We believe that this could have been an excellent term to describe high reliability, but marketing departments abuse it far too frequently.