OS-VoIP | Open Source VoIP » Hardware

Open Source VoIP in the carrier space : A look at Bandwidth.com

Aaron Rosenthal — Fri, 31 Oct 2008 21:58:49 +0000

We’ve talked a lot about enterprise adoption of OS VoIP but businesses are not the only users of this great technology, in fact there’s an untold story about Open Source VoIP and that’s its use within the carrier space. What too many people not in this field don’t know is that carriers are some of the largest users of Open Source VoIP technologies although few carriers will ever admit their use of open source. The reason why many don’t admit its use is the same reason why OS VoIP is slow to penetrate the large enterprise market; that reason being that OS VoIP is still perceived by an uneducated many that Open Source will always be the domain of basement dwelling techno nerds and hobbyists.

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.

Polycom SoundPoint IP 670 – now available for order

Aaron Rosenthal — Thu, 26 Jun 2008 23:41:25 +0000

So I just heard from my Polycom guy today that as of yesterday the Polycom IP670 phone is available for order. I had a few things to say about this phone in a previous blog here, but now that it’s available I’m sure it’s only a matter of time before we get one to play with at which point you’ll see a full review.

I’m quite surprised that the release of this phone has gone by almost entirely unnoticed. The fact that I’ve heard very little about this phone leads me to believe that there isn’t whole lot to this phone other than a color display. It seems like Polycom is really pushing theirApplications for their SoundPoint IP line which I haven’t found to be all that groundbreaking other than providing a more intuitive phone based interface for things like conferencing… nothing groundbreaking.

This is what I call groundbreaking interface>

not this>

You’ll probably see me write a blog reviewing these new Applications as there’s a lot more to talk about here than the IP670. Has anyone been working much with the Polycom SoundPoint Applications? Would love to hear your thoughts.

I hope Polycom comes out with many more pictures of this IP 670 because the only picture I could find on their website is EXTREMELY lackluster. I don’t get why they released their first SoundPoint color phone with a picture of a color screen that looks more like a DOS prompt. I’ve seen the color screen on the Polycom Microsoft Communicator phones which is quite sexy and assuming Polycom is using the same LCD screen, it should look pretty nice on the IP670. I honestly think Polycom is overdue for a re-design of the SoundPoint IP line.

I really hope the Polycom CX700 Microsoft Optimized Device will ultimately adopt the same open’ness of the SoundPoint phones once Microsoft releases their exclusive rights stranglehold on Polycom. I certainly hope that’s the case cause these phones are HOT!

Regardless of the IP670 not hitting the news hard, it should still be a great piece of hardware. The SoundPoint IP phones are rock solid and work swell with Asterisk. Polycom still has a few things to sort out with Asterisk (call barge anyone?) but overall these phones have proven more stable than other SIP phones we’ve tested like Grandstream, Snom and Aastra.

The color expansion module is a nice touch too. But seriously!!.. who took these pictures?? Polycom almost has me convinced that these color displays are only green. Something tells me they haven’t even photographed the new phones yet and just had someone photoshop some color.. aka green.. onto the IP650.

fring™ – a mobile VoIP client with SIP… among other things

Aaron Rosenthal — Fri, 20 Jun 2008 21:19:47 +0000

The eventual evolution of VoIP is going to break into the mobile market and go mainstream sometime very soon… unless networks start banning VoIP but that’s just stupid. The idea behind mobile VoIP is to make and receive voice calls over your data plan. With unlimited data plans standard with most carriers, this means unlimited calling without using up your minutes. This is a great concept but at some point most carriers will offer, just like Sprint and Helio, an unlimited anytime calling plan that is very cost competitive ~$99/month. So what then is the next great thing about mobile VoIP? I think it has everything to do with seamlessly integrating corporate IP PBX systems with business cell phones… and as mobile applications are more widely used I’m sure we’ll see some VoIP integration with mobile apps.

I just wanted to plant some ideas here but also want to discuss fring™which at the moment looks like the best mVoIP company out there. fring™ is a mobile internet service & community that enables you to access & interact with your social networks on-the-go, make free calls and live chat with all your fring, Skype®, MSN® Messenger, Google Talk™, ICQ, SIP, Twitter, Yahoo!™ and AIM®* friends using your handset’s internet connection rather than costly cellular airtime minutes.

There are other mVoIP companies like jajah but fring doesn’t charge anything for their service. As far as I’m concerned, my favorite thing about fring is that I can call my dad in Australia using the skype client from my mobile phone and it doesn’t cost me a penny!

What I really want to see is integration with fring’s SIP client and Asterisk. This would be a huge feature add for a large company using an Asterisk based IP PBX. I just sold myself. Maybe we’ll make this part of our Asterisk based IP PBX solutions…. let me get some engineers working on this pronto!