Posts tagged freeswitch
Open Source – What really drives it? is the desire to change and create something new? is it a firm belief in the idea that knowledge wants to be free and that software should roam the world? or when you boil down – is it just plain Ego?
I’ve been an Open Source advocate and evangelist for the better part of the last 20 years. I’ve started my days with Slackware Linux, moved to RedHat, then to Mandrake, then over to CentOS – which is now my choice of OS for the desktop and server. During these 20 years, I’ve seen various project come and go, companies rise and fall, technologies adopted and abandoned. A recent post on facebook from Dovid Bender got me thinking about this issue again:
Now, let’s put aside the grand discussion on the way the OpenSIPS project came about, their domain hijacking tactics, their overall confusion in the initial stages in regards to the difference between OpenSIP and OpenSER/Kamailio – let’s just put these apart for a second. Honestly, I can’t really tell the two apart, they use the same general configuration syntax and in most cases (over 95%), you can use the same configuration on both and it would work exactly the same. So, what does it boils down to? it boils down to Ego. Do I want to be considered traditional and stable and work with Kamailio, or would I like to be perceived as cutting-edge and work with OpenSIPS (although that isn’t true at all).
The same issue can be attributed to the ever growing battle between Asterisk and FreeSwitch. Now, each one was built for a totally different class of operation (although, Asterisk 12 does introduce new functionality that makes it shine much harder than FreeSwitch). People repeat the old “You’re melting our switches” FreeSwitch urban myth, but again, I still hadn’t seen one installation that truly did everything with FreeSwitch and is truly focused on using FreeSwitch to leverage something else. So, if FreeSwitch is only used as a media/application server, then I see no difference between it and Asterisk in that regard. More than that, if the added value of using FreeSwitch is just a mere 5-10% increase in performance, it just isn’t worth my time to do so. Now, I’ve used FreeSwitch in the past, don’t get me wrong – it’s a wonderful tool in that respect, and for Class-4 switching it is a massive tool. But when it comes to Class-5 and high-level services, sorry to say, Asterisk will always be my choice – not because it is better, not because it’s support and community is far more experienced, not because it out-performs FreeSwitch – it will always be due to one simple reason – it is the one I know will require the less amount of ongoing support and maintenance and will bring me to my target much faster than FreeSwitch.
A few weeks ago, I put the following status on facebook:
Now, the two have direct correlation – When a CTO/VP R&D isn’t a telecom’s guy – and he takes decisions for development of the platform – simply based upon the writings of others on the net – which is purely influenced by a religious war – he is incapable of making the right decision. Take Jajah for example, when Roman and Daniel started Jajah, they only tool they used back then with Asterisk@Home – because that’s what they had. When the company grew, they could have easily moved to new grounds – FreeSwitch was already around. Why didn’t they? Why did Jajah remain with Asterisk – adding OpenSER/Kamailio into the mix later on? Why didn’t they move to a new platform? was it because they have loads of code developed? companies throw away code like dirty socks every other day – they had the resources. Fact remains, the service was alive for a long time, the company was bought out by Telefonica Digital at a price of $215 Million.
On the other hand, let’s take a company like Truphone (and pardon me James, I know you’re gonna kick my ass next time we meet). Truphone had changed technologies over the course of times many times. Each time, abandoning the previous tech and going for a new one. So did companies like Rebtel, Spikko, Skuku and others. Amazingly enough, none of them could be considered a massive success. Word on the market currently says that Truphone is looking for additional investors, as their existing ones aren’t willing to put in more cash. Spikko’s original model is totally gone and the company literally caved-in on itself – and same applies to many others.
So, what does it boil down to? is Asterisk better? is OpenSIPS better? – these are the wrong questions. The questions should be:
- Is your R&D lead actually knows the arena he’s treading in?
- Are your decisions based on actual investigation or just by whim?
- Are you completely aware of the various obstacles and challenges you’ll meet?
- Are you building your development and product on rapidly changing technology?
- Who is backing your choice? a proper business entity? or a mere group of people with an idea?
When it comes to choosing between Asterisk and FreeSwitch, here are my reasons for choosing Asterisk over FreeSwtich any day:
- The ability to rapidly prototype any application is 5 times faster and 2 times more economical than FreeSwitch
- The installation path for FreeSwitch is much more complex and convoluted than Asterisk, making future maintenance a nightmare
- Digium is indeed a young company, but it sticks by its products and makes all efforts to make it the best it can – I always have someone to talk to
- Barracuda Networks is a well established company in the Storage/Security market – if you go to their website, their support for FreeSwitch (CudaTel) isn’t there at all – does that mean something?
- Asterisk is a very reliable, dependable, predictable piece of code – it is something I can put my money on and know exactly what I’ll get, FreeSwitch still isn’t
Ok, before all of you jump at my throat for not posting for a long time – I’m sorry. I can’t believe the last time I posted was around June, much has changed since then. So, let’s start with some updates… well, there are no big news updates. Since I left Humbug, I’ve been doing my best to keep busy with GreenfieldTech projects. We’ve successfully completed 10 different custom development projects since June and we’ve started a brand new services branch at GreenfieldTech – the VoIP Security Audits branch – but that’s a different post altogether.
So, I spoke at this years’ Astricon, which took place in Atlanta, Georgia!. This is my first time to the southern parts of the USA and might I add, Southern Hospitality is something you need to experience (I’ll write about that later). Much had changed at Digium in the past 2 years. Many of the long time project members had left to new grounds, Asterisk-SCF is now officially no longer in active development, in other words – this years’ Astricon was a fresh breeze of wind, specifically with new people at the driver seat and new ideas springing about.
So, let’s talk PRI cards. Once every often a company approaches me to evaluate their products. I do my best to be as impartial as I can – after all, Digium products are my favorite. However, I’m always happy to see a product that can compete nicely with Digium, simply because I believe it will make Digium products better and stronger. So, a couple of months ago, Allo.com approached me to evaluate their PRI card. I agreed, and they’ve sent me their PCI-e version of a quad span E1/T1 card.
So, let’s start with what I didn’t like about the card. People, it’s 2012, technology had progressed a great deal since the old Zaptel days, not to talk about the old Xilinx Spartan chips – comm’on, even the crappy Chinese boards don’t use that anymore – move on. Ok, let’s put aside the issue of the actual chips being used on the board – can someone PLEASE explain to me why I need to patch my DAHDI modules to support these cards? How shall I put it, patching DAHDI/ZAPTEL is so 2004. Make your card fully compatible with DAHDI, no patching, stream line the card with the DAHDI stock kernel module – OpenVOX did it, Yeastar did it (do an extent) – you want me to use your hardware, make it easy to install and simple to update and maintain.
OK, regardless of my somewhat reluctant feelings regarding drivers compatibility, I had the unit installed in a test gateway. It performed as I expected from a low-cost compatible. It held up nicely with normal traffic, but when I tried pushing 30 call initiations per second on the card, it heated up slightly and CPU spikes could be observed here and there. Now, in an office scenario – sure this card will do nicely – in a service provider scenario – I’ll think twice. Now, in the past I’ve received similar performance from other clone cards, so my estimate is that there is a group of engineers passing from one company to another, coming in with the know-how for a single design and they wrap that into a card.
Final word regarding the Allo.com card – not my favorite, but definitely a possibility for office environments. I have no idea how their other equipment holds up, but I hope that it holds as well, so that the office/smb market has a new option to choose from.
I will post my Astricon update later on.
Well, it’s been almost a month since I’ve started writing about the humbug project. Now, it’s time to actually get you people involved, at least in the initial levels. We are looking to add 10 additional members into the humbug call analytics suite. Currently available analytics during the alpha testing is inbound call analytics.
Our aim is to gather as much information as we can and as much user requests as we can, humbug is a community oriented project, thus it relies on community oriented input and feature requests. Participating members will be granted access to the humbug analytics portal, allowing them to gather statistical information regarding their inbound call hits and their top ten DID numbers – we are working on additional statistics. As new stats will become available, we’ll role those out into the service as soon as possible.
In order to participate in the closed alpha testing, please send an email to alphatest at humbuglabs.org, and we’ll send you a short piece of dialplan code to insert into your Asterisk server. Technically speaking, we’ll send you a short AGI command that looks like this:
exten => _X.,n,AGI(agi://somehost/DataReceiver,some_unique_ident)
The above line needs to be inserted into any place you would like to generate call analytics from. We’ll also enclose configuration steps for FreePBX (and other FreePBX compatible distributions). We are hard at work for creating a FreePBX integrated module, so you can do a one-click install.
When most of us think about PBX systems, we usually associate these with cumbersome usage, confusing dialing codes and in most cases – a PBX system is automatically associated with the annoying task of transferring a call from one handset to another. Lately, I’ve been thinking deeply about how people use PBX systems, is this really the only way to use a PBX system? is there something else to the mix? can we really enrich one of the oldest operational paradigms in the world? – and for that matter, can the public be re-educated to assimilate a new breed of PBX systems or services?
As to answering the question of re-educating the public, I guess I’ll have to leave that question to the head shrinks. As to answering the latter, enriching the PBX experience is both achievable and advisable. When I say enriching, I mainly talk about your ability to bring to the IP phone functionality usually not associated with it. Imagine to have the ability to receive a stock exchange RSS feed to your phones idle screen, notice that you stock is either rising or falling, and by the flick of a button – either sell or buy. We’ve all come accustomed to IP phones that look like the one of the right. A whole bunch of buttons, that in most cases have no direct use when our phone is utilized using a single account. However, these buttons can be externally re-assigned and re-programmed to achieve greater functionality – surpassing the normal behavior of just making phone calls.
The technology involved exists on almost every high-end IP phone on the market (well, at least those made by SNOM, Aastra, Cisco and Polycom – most of the Chinese makers don’t have this) – it’s called a Mini Browsers. Mini Browsers are exactly what they are called, these are simplified versions of your typical Internet browser. Some vendors had produced their own XML based Mini browser markup language (SNOM, Cisco, Aastra) while others had decided to provide a sub-set of XHTML (Polycom). The variations between the vendors are at the neck deep of the problems of using Mini Browsers, and that is that the formats are considerably different. Sure, SNOM had more or less adopted Cisco’s general structure, however, it still varies.
Through the utilization of this technology, it is possible to create phone based browser applications, that seem native to the phone user, as the general interface resembles the native phone interface. It is now the developers job to make the web interface displayed to the user as seamless and as native as possible, keeping in mind that the developer must remain agnostic to the information retrieval layer. Most companies leave their phone systems and these tasks to their system administrators and infrastructure team, however, this task is far beyond their capabilities and skill set. Creating an agnostic IP phone minibrowser dislplay layer, capable of utilizing multiple vendors and models, is a question of content management and content rendering, very must similar to the content transcoding problem that is common to the mobile content world – in other words, a sys-admin will create an ad-hoc solution, a programmer will create a proper, well structured, well designed solution that carry the enterprise beyond its initial needs and requirements.
A short example of how these interfaces work can be found here – on my company blog.
I keep a vigilant eye on Google Alerts. To be more exact, I’ve got a few alerts defined, in accordance to subjects that interest me. Two of these subjects are Asterisk and FreeSwitch. Recently, the following had been posted on a personal blog:
Asterisk just seems to come from a different world than what I am used to. Inflexible and problematic. Very long configuration files with ancient syntax. Now I’m far from pretending I’ve used Asterisk enough to understand it pros and cons well, but I have a decent software development experience and I can tell when something “smells.” I didn’t want to build our system (that I want to be flexible and scale well) on some old technology that is only living from its past.
And (most importantly) there are many people experienced with both Asterisk and Freeswitch favoring the latter: Anders Brownworth, Jonathan Palley (creator of Telegraph, a Rails plug-in that lets you talk to Freeswitch), or of course the creator of Freeswitch (and former Asterisk developer!) Anthony Minessale himself.
Reading the above simply flames me UP – WHAT A LOAD OF BULL!
The writer admits that: “Now I’m far from pretending I’ve used Asterisk enough to understand it pros and cons well, but I have a decent software development experience and I can tell when something “smells.”” – If he’s such a worthy developer, the above simply proves that he’s simply an RTFM!
Asterisk provides a multitude of methodologies for configuration management, dialplan logic programming, programmatic API structures – basically, all the tools you need to go about and create your application. While I admit that FreeSwitch is slightly better at the “Core Switching” environment, implementing a near true Soft Switching architecture – Asterisk is a totally different thing.
A while back I decided to develop a couple of my better systems on FreeSwitch, as a test of my abilities. To see if I can be as proficient with FreeSwitch as I am with Asterisk. I re-developed the applications on FreeSwitch from scratch, however, FreeSwitch’s somewhat annoying XML configuration structures made it a hell for me to develop something that I would ask my customers to put their money on.
I started working with Asterisk in 2002, and starting 2003 I tried to have people in Israel start using Asterisk. I can honestly say that the first ever true adaptation of Asterisk in Israel came about only late 2005, and even that wasn’t for mission critical applications. Asterisk had achieved a point in which it is now considered a valid replacement for classic IN and NGN applications and engines, such as SigValue. I suspect that FreeSwitch will become a more dominant contender in that part within a period of 18 to 24 months, as its developer community will grow – time will tell.
In my large installations, I like using Asterisk as my IVR/Application engine envrionment and using FreeSwitch as my routing and LCR environment, simply because it’s capable of handling slightly more SIP-to-SIP connections than Asterisk – but that is the only reason, all the rest is Asterisk.
Both projects are wonderful tools, each one with its own set of strengths and weaknesses. Learn how to balance the use of the two, utilize the best of both world – and stop being a bloody purist, that’s exactly what leads to poor technologies such as Microsoft and poor life practices, like wars and terror.