OS-VoIP | Open Source VoIP » junction networks

Asterisk and FreeSWITCH

Aaron Rosenthal — Fri, 01 Aug 2008 20:50:11 +0000

This week I’m using some link bait which discusses a few of the differences between Asterisk and FreeSWITCH. David Greenfield wrote a short blog post discussing one particular case study where FreeSWITCH was used over Asterisk. I wouldn’t say I’m all that crazy-go-nuts over the post but the topic is worth additional discussion since FreeSWITCH and Asterisk are both fantastic pieces of OS telephony software each of which are strong in their own right. This is in no way a comprehensive comparison between the two, but it’s a start.

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.

Junction Networks helps Microsoft be a little more “Open”

Aaron Rosenthal — Tue, 15 Jul 2008 16:38:21 +0000

Well… the title of this post is a little deceiving, Microsoft isn’t really “being Open”, but they’re openly (and officially) working with people who are, like Junction Networks… open by association Microsoft is!

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.

Junction Networks launches partner program

Aaron Rosenthal — Wed, 02 Jul 2008 19:29:14 +0000

Junction Networks is a better than average hosted VoIP provider and yesterday launched a new partner program which warrants a mention at OS-VoIP since many of our readers just might be interested.

Junction Networks’ OnSIP hosted VoIP product is almost entirely engineered using open source software like Asterisk. This is a testament to Asterisk and Open Source’s ability to reliably support over 4,000 users dispersed across hundreds of locations. Big enterprises take note – Asterisk IS suitable for supporting large user populations if engineered properly.

There is some criteria to becoming a Junction Networks agent. I just hope it’s more difficult than becoming a Fonality reseller.

Those interested in becoming authorized agents for OnSIP must demonstrate technical competence and an ability to sell, implement and manage Internet-based services.

Junction Networks is one of the few hosted VoIP providers who do not charge per seat or extension. Instead they have various plans ranging from $39.95 – $199.95/month which includes unlimited extensions, users, and inter-office calling. The trade off is that although you can have unlimited users, you’re still going to pay 2.9cents for every minute on the phone.