Well carriers ARE in fact one of the largest users and most ideal candidates for Open Source VoIP because they’re often the ones with the most to gain from the benefits of this technology- carriers spend squillions of $$$ on telecom infrastructures and thus they have the most to profit by simply replacing existing (and costly) proprietary hardware with Open Source software and COTS hardware. Large chunks of a telecom infrastructure can be replaced by various elements of Open Source Software and since telecom infrastructures are so expensive, theses savings can be astounding. Carriers also tend to have the in-house technical chops needed to work with Open Source software which is a skill few mid-sized businesses have. In fact I’ve found that increasingly carriers are requiring their engineers to be trained and well versed in software just like Asterisk and OpenSER.

One such carrier who not only uses Open Source VoIP everywhere, but embraces and openly acknowledges their use of Open Source is Bandwidth.com. Recently a registered CLEC in all 50 states, Bandwidth.com is growing Flash Gordon style. They’ve managed to top Inc. Magazines fastest growing tech companies 3 years and counting, all while using Open Source software to profitably grow their network and infrastructure at a pace and scale that has reliably kept up with their growing demand.

Now I don’t want to turn this article into an advertisement for Bandwith.com, because that’s not the goal here. The goal is to show that even a large successful carrier with thousands of business customers, with over a dozen telecom products, and a rising star in the telecom world, relies heavily on Open Source VoIP for a good chunk of their network infrastructure. Here at OS-VoIP we’re dedicated to proving OS VoIP’s ability to satisfy the needs of even the most demanding large enterprise… so from where I look at things, I really don’t see that big of a difference between the way in which a carrier network would be engineered and the way in which a large Fortune 1000’s VoIP network is built, in fact I would say that a carrier requires higher levels of redundancy (downtime means lost customers) and a far greater level of flexibility since carrier products and services must shift with market demand with speed and efficacy. So Mr./Mrs. CIO, take note because if Open Source VoIP is suitable for Bandwidth.com and many carriers alike, why not see what it could do for your organization?

I had the pleasure of speaking with Anders Brownworth, head of research and development at Bandwidth.com, and as a long time employee since 2002 (back when there were only 14 people; now there’s 175), I get the impression that Anders has been largely influential in the extent to which Bandwidth.com has adopted Open Source VoIP software. Anders is also a fellow writer at his self titled blog anders.com where you’ll regularly see posts about what he’s up to over at Bandwidth.com.

Bandwidth.com is a next generation telecom company where TDM switching is predominantly a thing of the past; replacing these old TDM infrastructures (typically the backbone of most RBOC’s) are IP networks which is the case for most young carriers building out a new infrastructure. Unless you’re a telco with existing investments in a legacy network, it makes about as much sense as a toothless carnivore to not build your network foundation on IP. Now the folks over at Bandwidth.com could have very easily built their IP network using a myriad of proprietary hardware (which they use in some places) but instead, like most startups do, they went the route of a more financially feasible and flexible option and that ended up being Open Source software. But alas, even while I’m writing this Bandwidth.com has solidified a greater partnership with Sonus Networks to build their Next Generation Network (NGN); a move spurred by their recent CLEC status. All of Bandwith.com’s gateway’s to the PSTN have always been Sonus, like most carriers, but their Sonus network is obviously going to grow even larger which will help them open up shop in more US markets to provide direct “last mile” access to their network…..but we’re talking about Open Source VoIP and that means we’ll talk about Bandwidth.com’s IP network.

Anders tells me that from day one Bandwidth.com has been a heavy user of OSS including Linux, Apache, and MySQL, but most importantly for us over at OS-VoIP is their use of Open Source VoIP software like OpenSIPS (formerly OpenSER) which has fixed Bandwidth.com’s core IP infrastructure on Open Source software from the very beginning… and it’s role is paramount. OpenSIPS is a SIP proxy/router software which Bandwith.com uses to route ALL of their SIP traffic; accounting for the majority of their VoIP calls and the billions of minutes each year that run through Bandwidth’s IP network. With SIP becoming a predominant standard in telephony, OpenSIPS has the potential to completely crush the proprietary IP routing and SBC market with its ability to support extremely large traffic loads while scaling in ways far more cost efficient than anything you’ll find in the proprietary market… all on COTS hardware!

But what do we all know about Open Source?… it’s that Open Source software is not always easy to work with. There’s no question the functionality is there, but I’ll admit that if you haven’t worked with something like OpenSIPS before, you should probably get your hands dirty (very dirty) before deploying something so mission critical as a SIP proxy for a carrier. The other option is hire a firm that knows what they’re doing. I’ve said it many times over, and I’ll say it again, the successful deployment of OS VoIP software for businesses or carriers is as much reliant on the engineer or firm who implements it as it does the software; make the right choices and you’ll reap endless benefits.

When it comes to delivering reliable VoIP services to customers over the Internet, the cruelest VoIP monster is packet loss- which causes latency- which in-turn causes jitter and dropped calls…not an ideal situation for a company trying to portray a professional image. The internet is not designed for the transmission of real time applications which has been the route of countless criticisms about the quality of VoIP. The farther your phone is located from the hardware terminating that call into the PSTN, the longer your latency and the greater your chances are for packet loss and thus poor call quality. There are dozens of VoIP providers today who are small businesses with “who-knows-what” running on the back-end and an infrastructure sitting in a single geographic location… these are the companies who usually give internet based VoIP a bad name. For example, if you’re a hosted VoIP customer in NYC and your hosted VoIP provider’s network is located at a data center in LA, there’s a good probability that call quality could be an issue since you’re talking about running packets coast to coast over the internet which as I said was never designed for real time transmission of data. What you want to do is use a hosted VoIP provider with multiple PoP’s (point of presence) throughout the country so that the distance your call has to travel over the internet is reduced dramatically. Sorry for ragging on you small VoIP providers but it’s just a simple fact… small VoIP providers with a network in one spot are best to serve customers who are geographically close to the network hub… but then this issue of latency and packet loss is a crap shoot, sometimes it happens, sometimes it doesn’t. Ok, latency and hosted VoIP provider pros and cons can be left for another article, another day. So where’s this going?….

Bandwidth.com on the other hand operates ~9 server farms and have POP’s on the east coast, west coast, and some in between. This dramatically reduces the hop your call has to make in order to get into Bandwidth.com’s network…. which in turn reduced latency and increases the quality and reliability of your call. The key is to get that VoIP call out of the Internet and into the carriers IP backbone as quickly as possible. I wanted to briefly touch on their network architecture just to explain some of the benefits of a distributed network which is what I think really separates the boys from the men in this hosted VoIP industry.

So which other piece of Open Source software is running behind the scenes at Bandwidth.com? The next is a new yet increasingly popular piece of software called FreeSWITCH. FreeSWITCH is somewhat of a competitor to Asterisk and while many will argue that one of the biggest advantages to FreeSWITCH is its ability to support up to 4 times the call volume of Asterisk, FreeSWITCH doesn’t have nearly the same breadth of capabilities and support found in Asterisk. Take a gander at a comparison I wrote about the two. FreeSWITCH is what sits behind Bandwidth.com’s new PhoneBooth product, a hosted VoIP solution, which was released over a month ago on Sept. 15th. PhoneBooth is a web based user interface to Bandwith.com’s hosted VoIP solution, providing their customers easy access to features and an admin portal that lets them manage their services. Developing an easy to use admin/user interface that integrates with the likes of Asterisk or FreeSWITCH has always been the golden egg of any company who ventured into developing their own interface of this type. Developing UI’s for Open Source software is always a time consuming process which is why the companies who spend the most amount of time and in-turn develop the most reliable interface will typically close up the code and license their newly developed interface.

Just to go off on a little tangent, Anders and I were discussing our frustration with Open Source developers who unfortunately give little or no consideration to how their product would look and work from a user interfaces perspective. Often OS software is written in the command line by hardcore programmers and by not including a UI, it unfortunately gives some OS software an elitist status because few people know how to work with it. Anders made a great comment which was that he’d “love to see some strong projects in the open source world that approach things from the designers perspective, allowing the designer to say “this is what should happen” rather than the user/admin interface being an after thought. I don’t know why more developers don’t do this because a sexy UI is perhaps the single most important thing general consumers look for…. and I digress…

PhoneBooth is the first robust interface I’ve heard of that was designed to work with FreeSWITCH (although Anders tells me that PhoneBooth WAS originally designed with Asterisk but later re-engineered for FreeSWITCH). Other examples of GUI’s designed to work with Open Source software like Asterisk include Switchvox, Trixbox, FreePBX, PBXtra, PBX in a flash, Intuitive Voice, and many many more. Each of the mentioned UI’s were engineered with varying degrees of success where the free GUI’s are typically less stable than the likes of Switchvox or Trixbox which are now licensed pieces of software; even though the foundation of these systems are built using Open Source Asterisk. Because Bandwith.com operates a tenant based environment, with thousands of customers, Anders and his team developed FreeSWITCH in parts, each part with a different responsibility and capacity to support larger loads. This is one distinction between Asterisk and FreeSWITCH which is FreeSWITCH’s ability to be easily broken up into pieces. Bandwidth.com developed separate conferencing, media servers, and databases from which PhoneBooth directly reads and writes.

I am told by Anders that Bandwidth.com just might open source their PhoneBooth project which would be absolutely fantastic for the general Open Source community! Some folks might even pee their pants. I do have my doubts that this will happen since PhoneBooth is already a valuable piece of Bandwidth.com’s business but if it is engineered as solid as I’d expect it to be, then PhoneBooth just might be the first robust GUI I know of for FreeSWITCH and perhaps it could be easily adapted back into working with Asterisk…. as I’m a little more of an Asterisk fan myself, this would be saweet.

And lastly no Open Source VoIP infrastructure would be complete without a dash of Asterisk here and there. When it comes to Bandwidth.com, Asterisk is primarily being used as a TDM to voip gateway which is just one functional characteristic to a piece of software that seems to know no boundaries in telephony functionality. Bandwidth.com has hundreds of these Asterisk boxes spread across the country many of which are used to trunk between Bandwidth.com’s IP network and legacy TDM phone systems or bridging the gap from a TDM carrier network to their IP backbone. It’s a simple role but Asterisk plays it very well.

If you made it to the end, and hopefully you did with a final sense of accomplishment, I want to thank Anders for taking the time and allowing OS-VoIP to dig into some great pieces of Open Source software running behind the scenes over at Bandwith.com. Open Source VoIP software, like those used by Bandwidth.com is being leveraged in places that most people wouldn’t even think of and in ways that are infinitely flexible. We currently live in a world where Open Source software (not all but some) has become so powerful, flexible, secure, reliable, and cost effective that ignorance is often the only argument left for not giving Open Source the brain space it deserves. I know I know.. not everyone shares the same passion for OSS and the first person to make it through this article who disagrees with me (hello you) will instantly, as if subconsciously wired into their brains, refer to support as the biggest issue facing Open Source…. and although I will agree that this is a problem for some Open Source projects, this argument is used WAY TOO MUCH as a generalization referring to all Open Source projects, because the support which exists for many OS projects can be remarkable.

Open Source VoIP software has progressed so much that knowledge of these systems has become a standard skill requirements amongst engineers working in this space. With hundreds of companies developing, implementing, and maintaining Asterisk (as an example), you’d have a hard time convincing me that Asterisk is lacking an appropriate support infrastructure. But, like all walks of life, there are firms who are better than others so if you’re looking to find a reliable Open Source VoIP engineering firm, with the ability to support your needs effectively, just make sure you evaluate your options thoroughly, and don’t always make your decisions based on price because if you do, you’ll usually get what you pay for. One thing many Open Source projects should take from the proprietary world is a more stringent and selective certification process. Having a particular certification to separate the boys from the men when it comes to Open Source engineering would make it much easier for firms to disseminate between a solid OS engineering firm and one which may be full of jokers.

If I’ve achieved anything by this article, look at the technologies Bandwidth.com uses and when you’re in the market for any enterprise grade telephony solution, I hope you’ll give OS VoIP technologies the attention they deserves.

I think an even better comparison between Asterisk and Freeswitch was written by Anders Brownworth which looks at the differences between the two from a slightly more technical overview. As head of R&D for Bandwidth.com, I’m glad to hear Anders is playing with Open Source software like Asterisk and FreeSWITCH. My last post about Junction Networks discussed the use of OS software in a carrier network, it would be good news for OS-VoIP to learn that a big player like Bandwidth.com also uses OS software somewhere within their infrastructure, and where (they probably do already but won’t admit it like most carriers). Perhaps Mr. Brownworth can shed some light on the topic for another OS-VoIP article???

Most people will agree that Asterisk in its current state has more feature capabilities than FreeSWITCH in its current state. What largely differentiates FreeSWITCH features and Asterisk features is how they operate as you begin to scale a system and the way in which those features and dial plans are managed.

I’m admittedly more biased towards Asterisk because it’s been around longer and well, because my company is a Digium partner, but I’m also not one to ignore new software even if it feels like sleeping on the other side of the bed. That’s the problem with these large stagnant corporate IT infrastructures, it’s that the people in charge of them have largely relied on their proprietary vendors for information about new technology, and have become too comfortable with relying on these folks for the right information. It takes a true IT leader to step out of their comfort zone and see whether there’s a better way of doing things, something other than that which has been spoon fed to them by vendors. A very simple way to prevent this type of comfortable stagnation is to simply read a few select magazines, and/or blogs on a regular basis; just to keep you up to speed with everything. Throw a wrench into the machine; rustle some vendor feathers; go ahead and see what’s new, source some technology solutions from competitors of existing vendors… there’s little to lose- either you find something better or your vendor freaks out enough to offer better pricing, it’s a win win!

Back to FreeSWITCH and Asterisk. So why would one consider using FreeSWITCH and why Asterisk? There’s no easy answer to this question because it truly depends on what you’re trying to do, and since both pieces of software offer near limitless possibilities, I’m left with only the time and patience to discuss just a few. Depending on what you’re trying to achieve, and what you need done, FreeSWITCH and Asterisk in my experience are typically used in a complementary fashion. Since FreeSWITCH was largely designed to satisfy the carrier space, perhaps its biggest advantage over Asterisk is in its distinctly different architecture. The general consensus amongst developers is that FreeSWITCH is capable of handling larger call loads on less hardware yet Asterisk has far more feature capabilities and is therefore perhaps the most suitable of the two in small to mid sized IP PBX deployments.

For anyone who has worked with Open Source VoIP software, they will know that in order to build the most stable VoIP system, you’ll probably end up using a collection of Open Source software (maybe even proprietary software as well). Our rule for production systems is only ever use the software that does the job the best, and if that means proprietary then so be it (most of the time there’s still a perfectly suitable OS alternative). This is the beauty of using software which is highly interoperable. One example in the design of an IP PBX would be using OpenSER for handling routing and load balancing, FreeSWITCH could be used as the IVR media gateway and conferencing, while Asterisk is left to handle the majority of the PBX features. Technically Asterisk could take care of all this but with a little more complexity, especially when handling thousands of simultaneous calls. I would argue that in the ~1000 extension space (still a fairly large system for Open Source VoIP standards) Asterisk may be all you need to build a complete IP PBX.

Asterisk has loads of features and although most work near flawlessly, there’s also a couple that don’t. One simple example is the call barge feature. I work with Polycom phones and wouldn’t have it any other way but the call barge feature for some reason or another does not work properly between Polycom and Asterisk. If anyone at Asterisk/Polycom is reading this, GIT-ER FIXED! So FreeSWITCH can actually be used in such an instance to provide this standard “key system” functionality.

For very specific applications like conferencing and media serving, FreeSWITCH is the clear winner.

As the above quote from Anders Brownsworth states, FreeSWITCH is also excellent for conferencing. In fact Junction Networks, see post, also uses FreeSWITCH for their conferencing service. One of the reasons why FreeSWITCH is so good at conferencing is that call conferencing is a very resource intensive activity. Each call added into a conference requires an exponential amount of computing resources. Although Asterisk handles conferencing quite well, FreeSWITCH can support more calls on less hardware.

I haven’t done much testing on the topic, but from what I hear people saying, typically a single Asterisk server has the capacity to handle ~250-300 simultaneous calls whereas FreeSWITCH users claim that with the same server ~1000 simultaneous calls can be handled. Remember that the purpose behind FreeSWITCH is…well… switching and call control, therefore most of the processes running FreeSWITCH aren’t all that resource intensive hence more calls/less hardware.

At the end of the day, it helps to know where you’re trying to go. If you plan to implement some element of Open Source telephony into your corporate communications infrastructure, you need to know exactly how the system must scale because scaling is one of the most important differentiators for which Open Source VoIP software to use and how you use it. If for example a large corporation decided to replace their entire Avaya infrastructure with Open Source VoIP software, the typical approach is start with a few small locations and eventually migrate everything into one large centrally managed yet geographically dispersed system. The difficult part about this approach is a single office might have 100 users, where Asterisk would be the software of choice, but a larger centrally managed system will likely be built for a much larger user population using a combination of the software I’ve already mentioned. You can’t expect that the lessons learned building a small Asterisk system will map well to a larger clustered system built with various OS VoIP software.

Highly skilled Open Source VoIP engineers are few and far between, my advice to anyone interested in OS VoIP is to either use a highly skilled OS engineering firm, or run your Linux engineers through weeks (if not months) of training. You might say, well what about a consultant? Consultants can be an excellent resource for projects like these, but my experience is that only 1 in 10 really know what they’re doing. There’s a lot of amateurs out there who might have plenty of Trixbox deployments under their belt but the second you say custom development or troubleshoot, they’re stuck with a finger up the bum, wide-eyed, and not a single clue as to what to do. I say all this because every week at SpecialAI some poor business gives me a call because their OS VoIP project got stuck where their consultants IQ ran out. Perhaps I’ll write an article about how to choose the right OS VoIP consultant/freelancer but the responsible move for most organizations with a stretched IT department is to spend the extra money and simply hire the right company with the right support infrastructure to build, deploy, and maintain mission critical OS VoIP systems.

Here are the two chapters extracted from the PackT AsteriskNOW book:

Chapter 5 – Tentacles of the PBX — The Calling Rules Tables
Chapter 7 – “For Annoyance, Press 1″ — Voice Menus and IVR

It seems that AsteriskNOW has largely been neglected by Digium ever since they purchased Switchvox. AsteriskNOW was Digium’s first attempt to an easy to use administrative interface to the Asterisk software but once Switchvox came into the picture it seems their engineering resources went away from the AsteriskNOW UI and into the Switchvox UI. This move obviously makes business sense since Switchvox is a real revenue generator for Digium yet I hope they soon re-direct some focus towards the eventual development of a comprehensive OS UI for Asterisk.

To date, there still does not exist (to my knowledge) a suitable Open Source Asterisk UI which I’d feel comfortable implementing in a large scale IP PBX. There are certainly plenty of Asterisk UI flavors but most have been created as a licensed product. In a perfect Wallgreens world, Digium or some other knight in shining armor will engineer an open source UI for administration and a UI for users; one which is reliable regardless of scale or clustering, and one which is Open Source. This would certainly result in the eventual nail in the coffin for many proprietary IP PBX systems. So far the closest I’ve found is Druid by Voiceroute which is still a work in progress.

Here at OS-VoIP, one of the things I’m trying to do is help individuals differentiate between Open Source VoIP apps that are ready for the enterprise and which ones are not. A lot of the misconceptions in Open Source VoIP stems from software which hasn’t been finely tuned enough to be enterprise ready or from implementors who just don’t know what the hell they’re doing.

Many of the OS Apps in this list of 74 are already widely deployed within the communication infrastructures of enterprises and carriers- like Asterisk and OpenSER. Others have a much lower adoption rate and still require a lot more development until they’re ready for enterprise adoption.

This isn’t a perfect world and hence some functionality required in a phone system is sometimes best left to proprietary software (for now). The good news is that the most important part of a communications system, the brains of a PBX, is perfectly satisfied by OS software like Asterisk and OpenSER. Proprietary software has it’s place in delivering added features and functionality to a system who’s core is built from Open Source software. Functionality like speech to text for example is (for now) best left to licensed software like LumenVox or using a proprietary contact center solution like Aspect on-top of Asterisk.

With the near limitless capabilities of software like Asterisk combined with an ever growing list of Open Source VoIP apps, the difficult part is to know which apps may compromise the stability of the system as a whole and which ones will best complement the functionality of your IP PBX. Either way there’s no question in my mind that the tools and applications already exist to turn Open Source VoIP into an IP communication system that rivals the likes of large proprietary systems by the likes of Avaya, Cisco, and Nortel.

For those of you in the US, have a great 4th of July weekend!

I really like the functionality of QC which includes web based and remote management of conferences, plus a standard calendar scheduling format for export into clients like Outlook.

“Quick Conference can be integrated with any PBX through SIP trunking or a SIP/TDM gateway and offers a range of security features to ensure the privacy and integrity of a conference call. QC can also be deployed as a managed hosted service, residing both inside and outside the enterprise network.” More info here.

The thing with QC, and for that matter all other feature based apps that work with Asterisk is that it’s just another bolt on to your PBX. It is easy to get carried away and start using multiple bolt ons with Asterisk but at some point there’s a limit. I believe it’s unreasonable to think that a normal user should be asked to use any more than 1-2 computer based telephony applications. The idea is to unify communications, not add to an already growing set of business applications.

Overall, QC is nice, sleek, and easy to use. For a larger businesses that does lots of conferencing, users would welcome an application like this regardless of it being something else to learn… because it gives them control otherwise reserved for more expensive conferencing systems from the proprietary guys.

Just today I stumbled on this article written by Flavio E. Goncalves which compares Asterisk and OpenSER. It’s a short and concise article worth sharing with you all. Here’s a few snippets.

“If you work with IP telephony, it’s quite possible that you have not heard about OpenSER, but certainly you must have heard about Asterisk. Well, I love a polemic headline and I have seen this question asked in the forums many times. So, I will dare to compare these two very popular softwares dedicated to the VoIP market. The idea here is not to show you which one is the best, but mainly how they are different from each other. Below is a comparison topic by topic.”

“Asterisk is a Back to Back User Agent (B2BUA), while OpenSER is a Session Initiation Protocol (SIP) Proxy. This makes all the difference between them. The SIP proxy architecture is faster than a B2BUA because it deals only with signaling. On the other hand, the B2BUA architecture, even being slower, handles the media and it is capable of several services not available in a SIP proxy such as Codec Translation (that is G729<->G.711), Protocol Translation (SIP<->H323), and services related to media such as IVR, Queuing, Text to Speech, and Voice Recognition.”

Read the rest of the story here

APSTEL’s Visual Dialplan is a GUI interface for configuring an Asterisk based PBX system. The big news is that Visual Dialplan is now available for Linux where it was previously only available for Windows. For anyone with an engineering background, this probably complicates things but for a small business owner it seems like this could be an easy way to setup a dialplan and actually understand how it all works conceptually.

I can’t praise or criticize APSTEL’s Visual Dialplan simply because I’ve never used it but would love to hear some feedback from the community. I’ll leave you all with some more screenshots.