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.

This acquisition marks a milestone for OS VoIP as a technology because it A) shows that Open Source VoIP is a viable business model and B) it reaffirms that Open Source VoIP is finally established enough, reliable enough, and mainstream enough to warrant acceptance by one of the largest proprietary communications manufacturers around.

This move by Nortel is ballsy for their industry but mostly because they’re just one of the first. It also plays well into their plan to become a software centric company. At OS VoIP we’ve always said that proprietary companies like Cisco, Nortel, and Avaya will need to adjust their business to stay competitive in a world of Open Source VoIP, and guess what, Nortel did. Since Nortel’s image has been a few PR campaigns behind Cisco and Avaya, a move like this is exactly what they need to re-vamp their image as a cutting edge communications company, plus it makes sense considering the “open” direction the entire technology industry is heading. From the quote below, you’ll see that Nortel has positioned more than one chess piece towards being “open”.

Over a year ago Nortel joined the open source community established by SIPfoundry** as an active contributor to the sipXecs open source project (led by Pingtel Corp), providing more than 300 new applications and features to date. The acquisition of Pingtel Corp by Nortel will further accelerate the development of a global open source ecosystem and reinforce Nortel’s direction and leadership in the development of interoperable and open unified communications solutions.

So on this day, August 13th 2008, mark Nortel down as the first large proprietary telephony company to take the leap into offering their own Open Source telephony solution. I don’t expect it to be very long for other proprietary businesses to follow Nortel’s lead but I don’t expect Cisco or Avaya to be scrambling for their cash in an effort to puchase an Open Source VoIP company. What I do suspect is that these proprietary companies will either begin to Open Source parts of their own software (doubtful), or partner with an Open Source VoIP company like Digium.

3Com partnered with Digium to re-sell their SMB solutions, Dell partnered with Fonality/Trixbox for their own small business solutions, I don’t think it’s unreasonable for more proprietary vendors to take the same approach.

Now don’t get me wrong, I think this Nortel/Pingtel acquisition is great for the overall evolution of Open Source VoIP and its acceptance in the market place, but it’s not like Nortel is all of a sudden going to be the next big Open Source VoIP company. There are too many established players like Digium, Switchvox, Fonality, and many more for Nortel’s newfound openess to eat away much of their business. If anyone should be threatened by this move it’s Microsoft. Microsoft’s Office Communications Server ‘07 is a software based unified communications solution which although sexy, costs a queens dowry and doesn’t play very well with others. With an Open Source UC solution offered by a billion dollar corporation, Nortel should be able to (with some development and good marketing) compete rather well against Microsoft’s OCS.

So if you take anything from this blog, I’m not saying you should go run off to a Nortel vendor for your next OS VoIP system, because you should run off to me :) but what you should do is realize there are plenty of reasons why an Open Source IP PBX or UC solution might suite your company just as well as any proprietary option. Nortel obviously thought Pingtel’s Open Source UC solution was good enough to buy the whole damn company, so why wouldn’t your organization at least look at Open Source VoIP as an option for your next IP PBX.

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.

When a company as big as Microsoft decides to form a partnership with a VoIP provider who’s network is openly, and almost entirely open source, it’s a big deal for people like us and another win for OS VoIP. I’m going to talk a little bit about this whole Junction Networks and Microsoft thing, then go off on a tangent…

Response Point and Junction Networks

Last Tuesday July 8th, it was announced that Microsoft had partnered with Junction Networks as a recommended service provider for their small business VoIP solution called Response Point which comes already pre-configured for a free trial with Junction Networks. Recent Microsoft news normally makes me throw up a little in my mouth but this got me thinking.

Response Point is truly a small business phone system, so much so you’ll be able to pick one up at your local Costco! My goal here is not to sell people on Response Point, but to “point” out that the approach Microsoft has taken with this IP PBX is not a whole lot different than how vendors piece together an Open Source IP PBX. OS software like Asterisk is typically installed on a combination of COTS (commercial off the shelf) hardware and similarly Response Point is nothing more than Microsoft software installed mostly using your own hardware. A business can utilize an existing Windows PC, you can piggyback off an existing LAN including your switches, and the SIP IP phones are from a collection of companies including Aastra (an Asterisk favorite), D-link, or Quanta Syspine (for operator functionality).

Because Response Point is particularly designed to use internet based VoIP service, there’s no need for telephony specific interface cards. So for internet based VoIP service, there are three companies whom Microsoft is sending their Response Point customers to- New Global Telecom, Cbeyond, and our friends Junction Networks. Out of these three, Junction Networks is the only company who automatically provisions new accounts online, so no dealing with over zealous sales reps, no waiting for proposals, and overall much less provisioning headaches.

Junction Networks is a leading internet based SIP/IAX trunking provider who additionally sell a hosted VoIP solution called OnSIP. The reason why I’m even writing about this on OS-VoIP is because Junction Networks’ entire infrastructure is almost completely built using Open Source software. I had a chance to speak with Junction Networks CTO John Riordan who was a good sport and gave me some insight into this Microsoft partnership and the “Open Source’ness” of Junction Networks’ infrastructure.

John tells me that the Junction Networks PSTN Gateway infrastructure is primarily built using OpenSER and FreeSwitch. They also use stripped down pieces of Asterisk in their OnSIP hosted VoIP service. I’ve been hearing more and more about FreeSwitch these days which is “reliable, stable, and efficient” says John. FreeSwitch is a fantastic piece of Open Source telephony software and one of the functions John says FreeSwitch is particularly good for is their conference bridging. Open Source telephony software is ideal for businesses and VoIP providers alike because not only do you have something which is highly customizable and malleable to your business processes, but it costs significantly less to implement, maintain, scale, and integrate with other apps.

Going Open Source is a smart move for any business because it might just be that competitive edge you need as a company. Junction Networks uses Open Source first because it does the job extremely well and second because it too gives them their competitive edge. “One of the benefits of using Open Source from a business perspective is that we avoid paying licensing fees. Instead, we get to pass these savings onto our customers which is why we’re the only hosted VoIP provider who does not charge a per seat/extension fee.” says John.

Why this is important to OS-VoIP

Most of us OS VoIP professionals spend the majority of our days trying to convince hard headed IT executives that Open Source VoIP solutions, like Asterisk, OpenSER, and FreeSwitch, ARE in fact ready for the enterprise. Whether you’re a sales person trying to sell OS VoIP to a CIO, or a Director of IT trying to convince a board, we all know what were up against – Open Source Racists!

An Open Source IP PBX (if implemented properly) will NOT be plagued with problems, it won’t crash constantly, you CAN get the same features-sometimes more, and NO Mr. CIO, you won’t lose your job…. infact if done properly, you just might get a bonus, a promotion, even a better job… plus I’ll write about you on OS-VoIP! It’s a win win.

The OS VoIP story we all know, a story also shared with thousands of small businesses, is that OS VoIP has penetrated a large portion of the SMB market as a cost effective, easy to use, and reliable IP PBX solution. You can’t look for a sub 50 seat phone system without finding something about Asterisk, Switchvox, Trixbox, Fonality, and some others. But…the story we don’t hear much about, and the story OS-VoIP is trying to tell is the use of Open Source telephony in the large enterprise market and OS VoIP’s ability to support user populations of 1K+.

The first hurdle here is that not everyone is willing to admit the use of Open Source telephony within their infrastructure, even if it works flawlessly. This is because Open Source for certain people still means ammeature, unrelaible, poorly supported, among other things. I know IT managers who use Open Source telephony within their infrastructure and won’t even tell their CIO because it’ll be met with the same prejudices that has plagued Open Source for the past decade. Many of these hurdles have been overcome by the 50%+ adoption rate of Linux within the enterprise which has done a world of good for OS in general, but OS telephony still requires a lot of extra legwork to convince executives on its ability to be reliable.

Open Source VoIP may not always be the best solution for a business, but its benefits, pros & cons should be evaluated against other large proprietary systems. OS VoIP is evolving so quickly that we’re beyond the point of viewing established Open Source applications as unreliable, the only “x” factor is in the reliable implementation of such a system but shame on you if you don’t evaluate an OS VoIP implementer just as throughouly as you would a proprietary VAR.

Junction Networks is one of the few phone companies who openly state that their telephony infrastructure is based on Open Source. This is why I’m writing about them. They’re a prime example of Open Source VoIP’s ability to be reliable as a carrier grade technology designed to support large user populations across disparate locations. Although Junction Networks won’t give me the number of extensions running on their system, they have 4,000 business customers. It would be safe to say that at bare minimum, Junction Networks must be running at least 10,000 extensions using Open Source which is a user population that rivals some of the largest corporate IP PBX systems.

And before anyone mentions it themselves, let me address call quality – All INTERNET BASED SIP PROVIDERS HAVE THE POTENTIAL TO HAVE POOR VOICE QUALITY, not because of their network but because you can’t guarantee quality of service (not to be confused with QoS) over the internet. Not to say there aren’t crappy VoIP networks out there, because there are, but you know what I mean… hopefully.

So coming back to Microsoft, my hope is that since the largest technology company in the world has partnered with Junction Networks, they in-turn, whether intended or not, put their trust in Open Source’s ability to provide reliable service to Response Point customers. You might say “well Response Point is for small businesses” but that’s not the point. The point is that Junction Networks is proof that large user populations can be supported using Open Source VoIP technologies.

Special thanks to Junction Networks CTO John Riordan and Robert Wolpov for their time.

In true OS-VoIP fashion, I’ll make the comparison between hosted VoIP and an Open Source IP PBX because there are some unique advantages an OS IP PBX has over hosted VoIP. I’ve seen plenty of companies decided that hosted VoIP made more sense than a proprietary premise based IP PBX but once you throw an OS IP PBX into the mix, the metrics can change quite dramatically.

Cost:

A customer premise OS IP PBX will require an upfront capital expense plus the additional cost of yearly maintenance which can vary in price depending on the level of maintenance required. Although an OS IP PBX does not hold the same costly licensing fees associated with proprietary systems, the capital expense is still not one to overlook. Depending on the type of OS IP PBX, you’ll probably discover a savings of 30%-50% compared to its proprietary counterpart.

By using a hosted VoIP service you’ll avoid this capital investment but may soon find that the operating expense usually exceeds the capital cost of a customer premise OS IP PBX in as little as 2 years. Because an OS IP PBX can be 50% lower in cost, your ROI time frame is half that of a proprietary system. It’s obviously much easier for a business to stomach a 2yr ROI than a 4-5yr ROI compared to what you would have paid for hosted services. The same reason why you wouldn’t rent a car for 2 years is exactly why you wouldn’t “rent” your IP PBX. There is still the cost of telecom services which are required with a premise based system but at least you have the flexibility of using whichever telco service you happen to find as the most cost effective and reliable- this may include POTS, T1, SIP, or some other internet based VoIP service (which I recommend against for a businesses).

I will argue that under most circumstances, a business under 25 seats will probably find more cost advantages for hosted VoIP than in purchasing an OS IP PBX simply because the investment required for a premise based IP PBX is usually more than a small business can afford. When it comes to larger organizations who focus largely on the TCO, a premise based OS IP PBX will win hands down in cost when you spread that cost out over a few years.

Hosted VoIP solutions are usually based on flat fees costing anywhere from $39 to $69 per month per station depending on the features required. Most hosted VoIP providers also require that you purchase your own IP phones which typically start at $100 for a decent device.

Hosted VoIP providers typically have a few pricing tiers which are based on features plus you may find an a la carte selection of advanced features you can purchase on a per/mth basis. Sometimes a business will be forced to purchase a higher tiered hosted VoIP product simply because they needed just 1 advanced feature which is not included in the lower tier. Some smaller hosted VoIP providers may be flexible enough to work out special pricing but this is not the norm.

OS IP PBX system on the other hand come standard with a full set of features that require no additional fee beyond the cost of the system itself. More advanced features and functionality may be an additional cost, but this cost need only be made once because there aren’t any ongoing licensing fees associated with an Open Source system. When you add a feature to a proprietary system, you may be required to pay a fee for every extension on that system, this is not the case for an OS IP PBX and therefore advanced features a far more affordable for larger systems.

Reliability:

Hosted VoIP providers treat reliability as a number 1 priority since they are responsible for the voice services of countless customers. The infrastructure on which your VoIP service operates are located in data centers with backup power, systems monitoring, high security, and much more. Since each hosted VoIP provider delivers VoIP service to thousands of customers on a single system, they can hardly afford an outage. That being said, with all the levels of redundancy in place at a hosting provider, the most common outage is typically not with the hosted VoIP infrastructure itself but with the services delivering voice calls from the hosting provider to your business location. VoIP delivery methods range from your existing internet connection, to dedicated T1’s or MetroE circuits. VoIP services delivered from most hosting providers run over a single circuit with zero redundancy. If for example, your T1 or cable goes down, so does all your voice service. This is one of the biggest problem hosting providers face when trying to deliver a reliable voice product to their clients. And because many hosting providers still rely on other Local Exchange Carriers such as Verizon for delivering these circuits, your business is now at the mercy of multiple telecommunication companies.

There is one vital flaw with hosted VoIP services and this flaw is that the majority of hosted VoIP providers rely on the internet to transmit an IP voice call. Unlike traditional phone service which utilizes a carrier’s network to send and receive calls over the PSTN, the quality of a hosted VoIP call can be largely dependent on the “weather” of the internet. Sometimes packets are sent from A to B without any loss but sometimes if the Internet is having a bad day, you’ll lose some precious data packets which will result in a choppy voice call or even worse, a dropped call. This is why most hosted VoIP providers don’t offer SLA’s since there are too many factors and external parties involved in delivering a stable voice call. This isn’t to say that VoIP over the internet doesn’t work, because it does, and it works well much of the time. It’s just not possible to guarantee the same results to every customer, and if your business needs its phone service to work 100% of the time then using an internet based hosted VoIP provider is not my recommendation.

Instead, the most reliable way to go hosted VoIP is from a carrier who owns their own network. Companies like M5 Networks will sell you a dedicated T1 over which your VoIP service is delivered. This T1 does not hit the Internet and instead carries a call from your desktop, over the T1, then directly into the carriers network, their PBX (which gives you features), and out through to the PSTN. There’s no need to hit the internet which dramatically reduces the chances of packet loss and poor voice quality.

It would be hypocritical to say that a premise based IP PBX isn’t susceptible to such service outages since often telecom services to an IP PBX are delivered over the very same T1’s, PRI’s, and MPLS circuits as hosted VoIP solutions. The reason why a premise based IP PBX can be more reliable than hosted VoIP is because a premise system can utilize redundant services such as the 100 year old technology called POTS (copper lines) which to this date is still one of the most reliable. “Should” a primary voice circuit fail, the PBX will automatically route calls over POTS thus maintaining full system functionality during a circuit outage. The cost of maintaining a few POTS for redundancy is something most small businesses can afford and certainly worth while should a primary voice circuit fail.

One great thing about hosted VoIP is that you are never responsible for its overall health and availability. Just set it and forget it! A premise based IP PBX is a piece of expensive equipment that you’re responsible for. The unique advantage of many Open Source IP PBX systems is that they’re built on the very same hardware that your IT staff are already familiar with. I would recommend that every IP PBX be supported by a vendors maintenance plan, but simple things like replacing a hardrive can be quickly done by any average IT employee.

Scalability:

Hosting providers are in the business of scaling a system infinitely. If they couldn’t, they wouldn’t be able to add additional customers. As your business grows, so can your hosted solution. What one must not forget is that for every user added to a system, there is a direct linear cost associated with the number of users added. Most OS IP PBX systems have the ability to scale significantly but instead of sharing system resources with thousands of hosted customers, a premise based IP PBX is all yours.

Hosted VoIP scales easily- just place the order and you’re good to go. This does mean that you’re entirely reliant on the schedule of your hosted VoIP provider. If you have purchased a support plan with your OS IP PBX, typically you’ll find that most vendors have a 30-60 minute response time to MAC requests which can be done remotely and usually accomplished quicker than the amount of time it would have taken with a hosted VoIP provider. There are obviously hosted VoIP providers who can be quick and IP PBX vendors that take forever so just make sure that each company has an SLA that meets your needs.

Functionality:

Functionality and features vary from hosting provider to hosting provider, but it’s safe to say that most will
deliver many of the standard features businesses require such as voice-mail, call transfer, call forward,
auto-attendant, and more. In some cases, hosted VoIP may come with very similar features to those included standard with an OS IP PBX.

The downside is that the features available with hosted VoIP solutions are limited to a finite a la carte menu. Some may come standard, where others may have an additional monthly fee, and some advanced features aren’t even an option. One of the most popular uses of VoIP for mid sized businesses is having the ability to integrate with other 3rd party internal applications for more efficient business processes. Hosted VoIP rarely has this level of versatility simply because the customer does not have the necessary access to their communications system in order to achieve integrations of this type.

Whether you need it today or not, having the ability to implement 3rd party integrations, unified communications, collaboration, call center applications, and many others is the reason why many organizations choose a premise based OS IP PBX. The Open Source nature of an IP PBX gives an organization a lot more flexibility in how quickly and easily an integration between the IP PBX and 3rd party application can be achieved.

This may change in the near future as I expect many hosted VoIP providers will begin partnering with 3rd party application developers. Already companies like Ribbit have integrated their VoIP services with Salesforce.com’s CRM package.

Tips:

The advent of VoIP as a mainstream business product has prompted thousands of hosted VoIP companies and Open Source IP PBX vendors to open up shop. I think it is almost too easy to get into the hosted VoIP business and companies like Fonality will recruit anyone as a re-seller of their phone systems. My recommendation to any small business owner is to use well established companies with existing customers and a good track record. If you’re going with a hosted VoIP provider, make sure you research whether they’ve had many network outages and for how long. Network outages happen, but they should never be frequent nor should they be for long durations….obviously. I would advise against using an internet based VoIP service if call quality is of high importance but maybe I’m just being a perfectionist since this is what most people do.

If looking at an OS IP PBX, make sure you use a company who fully understands the inner workings of the product they’re selling. Open Source systems require a more in-depth technological understanding than a simple plug and play proprietary system. Many OS IP PBX systems have easy to use admin interfaces but If you ever run into a complication with your IP PBX, you’ll be very happy to have a vendor who has the right engineering talent to solve the problem quickly.