Don’t Replicate – Federate

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For many years, the question of high availability had always circled the same old subject of replication – how do we replicate data across nodes? how do we replicate the configuration to stay unified across nodes? Is active-active truly better than active-passive? and most importantly, what happens beyond the two node scenario?

Since the inception of the Linux-HA project (and I do believe it’s been around for years now – over 15 years), it has been the pivotal tool for creating Linux based high-availability clusters. Heartbeat, Stonith and Mon will take care of floating the IP numbers and services across – no biggy there, making sure the data is consistent across the board, that’s something completely different. Recently, one of the better known Asterisk Commercial offerings had launched an Asterisk-HA solution – it’s been long due – it’s just a shame it’s a commercial offering without an Open Source derivative, after all, it is Open Source based (I hope).

However, being a high availability solution on one hand, doesn’t mean you are truly a clustered solution – it is an active-passive solution, with a major caveat (at least as I see it), that if your data sync fails for some reason, you end up with a split-brain issue – and your entire solution is now made moot. Don’t get me wrong here, I think that for now, the solution is the next best thing to sliced bread, simply because there is no other solution out there. However, the fact this is the only solution, doesn’t make it the right solution.

What does federating mean in this respect? it means that data doesn’t need to be replicated across the board, it is automatically trickled across the network, making sure all nodes in the network have clear visibility for it. If a node fails inside the cluster, client automatically redirect themselves to a new node, no need for floating IP numbers. Call routing is automatically determined upon request and are never preset for the entire platform. And most importantly, the amount of data traversed between the nodes is as minimal as possible, preventing excessive usage of network resources and I/O.

What would it mean to federate the configuration of a PBX system? first of all, make sure each unit is capable of working on its own, information should be trickled across the nodes via two methodologies: A multicast/broadcast mechanism (for local LAN connected nodes) and a Published/Subscriber relation (for externally connected nodes). When a change is made to any of the systems, that change is then replicated to all the systems. The configuration is never fully transmitted between nodes (apart from a new node joining the cluster). Routing decisions are dynamically made across the network, they are not predetermined or preconfigured. There is no need to keep the cluster nodes in perfect physical alignment, mixing hardware specifications should be considered the norm. External devices should be able to “speak” to the cluster, without being aware of its existence.

Once we achieve all of the above, we’ll truly get to a point where we’ve clustered Asterisk (or another open source project) the right way.

Federating Asterisk – truth or myth?

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During this years’ Asterisk Developers’ Conference, one of the subjects I’ve raised an issue for Asterisk is: “Federating Multiple Asterisk Instances”. Now, for the seasoned Asterisk user/developer, the answer would be simple – use Kamailio/OpenSIPS for that scalability, and use Asterisk as a Media Gateway or application server.

But I ask the following: “What if we could federate Asterisk without the need for an external component? What if we could federate Asterisk in such a way where our users aren’t event aware of the federation process, and it’s fully autonomous? What would actually be required in order to do that?”

I’m normally confronted with these questions on a day to day basis, looking at the problem from different angles – thinking to myself: “Ok, I know the normal box here – but where are the outer limits? what can I do to make it more robust on one hand, without truly making a mess out of it.”

A federated database is defined as: “A federated database system is a type of meta-database management system (DBMS), which transparently maps multiple autonomous database systems into a single federated database. The constituent databases are interconnected via a computer network and may be geographically decentralized. Since the constituent database systems remain autonomous, a federated database system is a contrastable alternative to the (sometimes daunting) task of merging several disparate databases. A federated database, or virtual database, is a composite of all constituent databases in a federated database system. There is no actual data integration in the constituent disparate databases as a result of data federation.” – http://en.wikipedia.org/wiki/Federated_database_system

So, we would like to virtually create a “map-reduce” functionality for Asterisk? can we truly create a map-reduce’ish functionality for Asterisk? should it be internal? should it be external?

In order to accomplish this, we are required to create a federator – a device capable of handling the information regarding each users, device, trunk, provider and other wise SIP/IAX2 entity connected to our system. The federator for all practical purposes is a data store, be it a key-value store, a database, a shared memory environment or some other form of data distribution layer.

Here are some key issues that true federation may be required to tackle:

  1. Geo-Position Agnostic – A truly federated system should render services identically across the board, regardless of where the user is located.
  2. Services Agnostic – A truly federated system doesn’t care if the user is connected to an Asterisk server version 12 or 13, it should behave identically.
  3. Version Agnostic – A truly federated infrastructure can leverage older version and even other software, without changing the underlying federation layer.
  4. Predictable Scalability – A truly federated infrastructure will allow for growth to be planned linearly, with discrete measure methods.

So, you want a tip on how to start federating your systems? here’s step number 1 – there is no central registry, there is no SIP proxy, there is only the cloud and the services it renders. Start thinking from this point and see where you go.

Asterisk ARI – What AGI/AMI should have been

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Asterisk ARI – for a seasoned AGI/AMI developer like myself, ARI is a serious mind warp. Why is it a mind warp? simple, it’s all the things we wanted AGI to be, and the reliability we wanted AMI to have, minus all the work around we needed to do – in order to get similar functionality in the past.

So, is ARI truly a replacement for AGI/AMI? well… I think the true answer will be NO. Is a replacement for the Asterisk dialplan? well… I think the answer to that is NO as well. “Say, are you messed in the head? first you say “What AGI/AMI should have been”, and then you say it’s not a replacement? – are you mental?” – well, there are a few reasons why I claim it’s not a direct replacement, and I’ll detail these here.

In order to explain, I’ll give a few examples, using the “in-development” PHP ARI wireframe that I’m developing, called Stanley.

Synchronous vs. Asynchronous

ARI by definition is asynchronous. Keeping that in mind, in means that that any command you give it will get queued or spooled in some manner, and return back an immediate result. Just to illustrate it, let’s examine the following code segment:

$this->stasisLogger->notice("Stasis Start");
$lastResult = $this->channels->channel_playback($this->ari_endpoint, $messageData->channel->id, "sound:hello-world");
$this->stasisLogger->notice("Last result: " . $lastResult);
$lastResult = $this->channels->channel_playback($this->ari_endpoint, $messageData->channel->id, "sound:demo-congrats");
$this->stasisLogger->notice("Last result: " . $lastResult);

For all practical purposes, you should regard $this->stasisLogger as a simple logging object, and $this->channels as a model to initiate ARI Channel requests. If you use the above the code, and activate it from with a Stasis application, you would listen to the “hello-world” and “demo-congrats” segments. Now, let us examine the following code segment:

$this->stasisLogger->notice("Stasis Start");
$lastResult = $this->channels->channel_playback($this->ari_endpoint, $messageData->channel->id, "sound:hello-world");
$this->stasisLogger->notice("Last result: " . $lastResult);
$lastResult = $this->channels->channel_playback($this->ari_endpoint, $messageData->channel->id, "sound:demo-congrats");
$this->stasisLogger->notice("Last result: " . $lastResult);
$this->channels->channel_delete($this->ari_endpoint, $messageData->channel->id);

The only difference here is the last line. If you activate this code, you will hear the world “Hello”, immediately followed by a disconnect. “Wait a minute, what just happened? – wasn’t I supposed to hear everything?” – that’s exactly the point, the answer is NO! The asynch nature of ARI will simply queue the first 2 playback requests, while the hangup is performed almost immediately – the playback simply never get to be executed.

In other words, if you need something to be synchronous within the dialplan, you may need to work differently about it. If you are familiar with the Node.JS framework, you are fairly familiar with this issue.

ARI is for writing applications, not IVRs

When the Asterisk team created ARI, their idea was simple: “Don’t manage the queue application, simply write your own”. Same applies for managing multi party conference calls, call origination, etc. In 2009 I wrote a book about AGI programming, where I’ve explained the methodology for “Atomic AGI development“. The concept behind Atomic AGI was to contain small logic units in AGI scripts, and leave most of the heavy lifting to the dialplan. This methodology enables to create scaleable Asterisk platforms at fair ease, and introduce additional technologies, without going about and adding odd things into Asterisk.

ARI is meant to do something similar, in the form where you can go about and create your own logic, contain it into a singular application and activate when you require – for example, rewriting the queue application. One of the first applications that I’ve decided to re-write using ARI was a Radio broadcasting system that I’ve developed in 2006. The problem with that application was that I need to hold about 600 callers in a single queue, and attach them over to the broadcasting booth as required. Of course I needed to enable full call control, caller management, UI and more. Initially, I used MeetMe, MySQL, and AMI to do this. Later on it changed to MeetMe, Redis, AstManProxy and some other tools – but it never seemed to please me. The fact that I needed to maintain 2 MeetMe bridges, one for holding people and one for the actual broadcasting really bugged me. Yes, when Asterisk 1.8 came out I migrated to the Bridge application and yes, I updated bits and pieces here and there, but it was never what I wanted it to be.

When I started playing around with ARI, I said to myself – this is the perfect application to migrate to ARI. The only thing I needed was a simple Stasis application to read my state correctly, and that would be activated once the called is put into the waiting area – so in terms, I’ve developed a very simple queue application.

IVR heavy lifting was done using dialplan, but the actual service was done with ARI.

Blades and Bleeding Edge

Now, before you go about migrating all your existing code to ARI – you must remember this: If you walk on the bleeding edge, expect the blade to cut you here and there. Currently, I hadn’t yet seen any proper ARI wireframe available. I’ve seen some work done with Node.JS and Ruby, but I can’t say that I’ve taken a fancy to any of those. Honestly, my comfort zone is very much PHP and C/C++, what can I say, I’m old school.

When I started building the Stanley wireframe, it was fairly frustrating – simply because not everything was that much clear and clean. In addition, as Asterisk advances, ARI will change and advance as well. What ever you write, make sure it’s modular enough so you can change it as required.

 

Mobile VoIP OTT is Dead! – Long Live Mobile VoIP OTT!

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What do the following have in common: Skype, Viber, Whatsapp, Line2, Tango and Kakao? Yes, there are all OTT apps for your mobile phone that enable you to communicate with your peers. Skype, Viber, Line2, Tango and Kakao actually enable you to call one another. Each one dominates a section of the world, where Kakao and Line2 are dominant in the far east, Viber dominates Japan and Eastern Europe and Skype kind’a says: “Look at me bit**es, I’m all of you combined”.

What do the following have in common: VoipDiscount, Nymgo, WiCall, VoIPstunt, Vox Mobile, Cloud Roam, Skuku? All of these are VoIP Mobile OTT apps, similar to the above and yet – no one truly heard about these or is using them. Each one of the above is more or less a replica of the previous one, maybe with one or more added features – but all in general are the same pitch and bit**, make cheap calls over VoIP via our service.

So, what does it all mean? it means one simple thing, no one truly cracked the formula to make money on the Mobile VoIP OTT business – everybody is still looking for the killer business model/VoIP OTT Application. What is the right way? providing low cost calls? providing business oriented services? providing simple roaming solutions? maybe bundling roaming data plans and SIM cards? or maybe, all of these are sooooooo passe that the world just says: “Stop fu**ing about and create some truly new, change how think and how we work completely. Paying 1 or 2 dollars more per month, I’m not gonna change my service for that – it’s pointless.”

So, what are the true killer apps that will truly say: “this is a game changer, from this point onward, VoIP OTT will no longer be the same!” – Here is a list that I believe will make the difference:

1. Make calls completely social – Phone numbers are so 18th century, they are pointless

2. Make your phone aware – Presence and availability is key

3. Drop the stupid things – call recording, visual voicemail, funny sounds, funky tones – stop the bullshit, give me proper services than stupid features

4. Make your service reliable – stop behaving like a website operator and thing like Ebay, every minute your service is down or affected by bad service you are loosing money

5. Make work, then make pretty – application design is important, product design is important, but not more than the product itself

6. Invest in support and monitoring – relying on your suppliers to do it for you is stupid and childish

7. Only blame yourself! – when something fu**s up, it means that you did your job wrong and you cut corners. Don’t start blaming your colleagues or your contractors, they are only doing what you asked them to do

And most importantly, remember the following statement: “I’ve seen the furthest, because I sat on the shoulders of giants.” – don’t tell the world how you’re going to obliterate Whatsapp and Skype, look at them, strive to be them, and then do it better.

I wish all of you good luck.

Technology? Religion? or just pure Ego?

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:

  1. Is your R&D lead actually knows the arena he’s treading in?
  2. Are your decisions based on actual investigation or just by whim?
  3. Are you completely aware of the various obstacles and challenges you’ll meet?
  4. Are you building your development and product on rapidly changing technology?
  5. 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:

  1. The ability to rapidly prototype any application is 5 times faster and 2 times more economical than FreeSWITCH
  2. The installation path for FreeSWITCH is much more complex and convoluted than Asterisk, making future maintenance a nightmare
  3. 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
  4. 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?
  5. 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