Saturday, May 7, 2011

Two technologies striking back in VoIP - part II

Do you remember VoIP/Telephony shows back in 2005-2007? It was IMS (Internet Multimedia Subsystem) all over. Then silence.
What killed the trend? Did everyone deploy it and this was it? No, there were many deploying pre-IMS, half-IMS or IMS-ready solutions (most of them being just some re-labeled existing system by the marketing/sales departments of some vendors) for astonishing amount of money, bringing back to them no other benefit that a SIP server with registrar, location and proxy services could deliver.

Many of us, including me, I was reticent about IMS. It was not because things like being a bad idea, but seeing who is behind the movement (read old telco vendors), organizations with few understanding of new communication demands over IP. They couldn't have it pushed beyond voice. Remember, IMS concept promised a framework to offer voice and new real time applications to mobile networks.

But voice was there, it had a dedicated channel (no IP) even in 3G networks. So an IMS with nothing more than voice brought zero add-value to services. There was no real return of investment for most of those deploying x-IMS.

Things are changing, demands for data over mobile networks is increasing, the operators are forced to go to 4G or what so ever will be next. Here we talk about technologies like LTE. With such technologies, there is no longer a dedicated channel for voice. Everything is IP.
Therefore the operators need a system that has to route the voice over LTE. One of the options, the natural one, is IMS.

IMS is a lot about defining standard interfaces, to handle services such as: interconnect, roaming, billing, access policies a.s.o. Not bad at all. Many concepts map directly, entirely or partially, over the standard SIP concepts, e.g., HSS (Home Subscriber Server) and SIP Registrar/Location Server.

It is almost mid of 2011, for few years there was a gap in (public and high) interest regarding IMS. But now is different. Starting with the end of last year, more and more people showed real interest in bringing IMS extensions in Kamailio. They are in the source code repository for some time, still under heavy work, but you can run it -- see the tutorial pointed at this web page.

First the big advantage for companies looking to use it is trying it at very low cost. Although the IMS extensions were available for some long time for free through OpenIMSCore project (which is part of our SIP-Router.org development eco-system), this time there are made available on top of a very stable VoIP/Telephony engine. All the IMS extensions are modules, so the stability of the other extensions and core is not affected at all.

More over, apart of voice, Kamailio (being the core of the IMS solution) offers add-on services such as presence and instant messaging out of the box. Furthermore, existing embedded programming language interfaces to Lua, Perl or Python offer a straightforwards and easy way to implement RTC goodies (e.g., see some slides with a tutorial for sending notifications to Twitter on missed calls).

I summary, with the IMS extensions on top of Kamailio:
  • free-as-in-beer system that anyone can try before deciding to pay integrators for professional installation
  • rock solid underneath system
  • access to other hundreds of RTC extensions
  • services beyond voice
  • embedded scripting languages
  • open source, vendor-unlocked environment, with many companies able to offer professional consultancy services
Forced by the need of a new voice system for LTE, backed up by cost effective and reach in extra features solutions like you can get with Kamailio, IMS has a good chance to succeed this time.

First part of writing about technologies striking back in VoIP was about IPv6. Same like there, things seems to get serious now (repeating the disclaimer from first part: the statements are made based on the activity inside our mailing lists, looking back at the discussions about these subjects in the past 10 years).

On our mailing list, Jason Penton (new developer to work on IMS extensions), revealed his company plans to deploy IMS in several countries across Africa (Uganda, Tanzania, Nigeria & DRC -- click to see the message).

The development of the IMS extensions is coordinated by Carsten Bock, long time developer and member of management team of Kamailio.

If you are interested in IMS, Carsten (along with other developers) will be present at Kamailio booth during LinuxTag 2011, in Berlin. We will be glad to talk with you, just drop by. We have several visitor passes that we can give away, write me an email if you are interested. Of course, flyers and giveaways will be on site, more details about developers availability at:

Friday, May 6, 2011

The Others - We 'Switched' the Seats

The other day I saw on the web space the launch of a new mailing list about IPv6 by SIP Forum. SIP Forum seems to be an exclusive club of Telco vendors/equipment manufacturers and operators (annual membership fee $7500). Claiming to some extent they help promoting adoption of SIP, the group is well known, also maybe due to organization of various SIP related events such as SIPit (where I participated several times).

Do they help with SIP adoption? I don't think so. Why?

Just short details about what triggered this post now, but the facts are actual since many years. So I subscribed to the new IPv6 mailing list using my address used everywhere on all the mailing lists I am member of. It is from one of the very popular free emails providers. And I got rejected, with the following reason:

Your request to the IPv6 mailing list
Subscription request has been rejected by the list moderator. The moderator gave the following reason for rejecting your request: "As a SPAM control measure, all SIP Forum WG mailing list subscriptions must be from an address (domain name) that has an obvious association with SIP and IP communications technology. Please re-subscribe under such an email address, or send email to describing the organizational affiliation with SIP and your interest in participating in the WG mailing list. I hope you understand and appreciate our desire to keep spam from the list, and will bear with this inconvenience." Any questions or comments should be directed to the list administrator at: [email address]
To clarify, I don't claim my name or address is something well know in the SIP world, that is the most irrelevant aspect in this context. For the record and the reason I'll point out later, in summary: I have 10 years working with SIP and only SIP, spoke every year at least 3-4 times at SIP related events, participated to many SIP Forum events (including SIPit in Stockholm where I helped setting up the IPv6 SIP testbed, providing several Kamailio instances on IPv6 to be used by all the others for various use cases testing). It was not even SIPv2.0 when I started (RFc3261, June 2002), since then managing and developing SIP Express Router (SER) and/or Kamailio (OpenSER) SIP server projects. The IPv6 support is in our project (May 2002) also before SIPv2.0 was out. My interest was not in learning anything, but helping others to figure out the way to SIP IPv6. We did our work back in 2002.

The issue here is something that's being a reality for many year, nobody is speaking about it. SIP has been hijacked from the purpose of innovating the real time communications over IP networks. It is going to be killed if the situation goes on like this.

Back in early 2000, people working with SIP were open minded, the specification written at that time were for IP networks and new services. It was small companies, enthusiasts, academic institutes and open sources projects really pushing SIP ahead. It was cool, it was fun. Just as an example: KPhone, a very early implementation of SIP softphone, it works even now with audio, video and, surprise, PRESENCE, based on initial specifications (end to end presence) -- and see author's picture, Billy Biggs, when he did it in early 2000.

Now, all organizations that eventually coordinate SIP independent activities, mainly including SIP specifications, are controlled by the old Telco vendors/operators and polluted with Telco mentality. If you are not one of us, fuck off. Besides that, basic concepts of real time communication became nightmares in specifications (did I say SIMPLE?), events and conferences have prohibitive fees, and I can continue...

Hey guys, you do have to retire! With this way of thinking you stopped innovation in SIP, you turned SIP in a protocol to put PSTN architecture on IP. It is not going to work, trust me. We do not longer need PSTN-type of communications, we need new ways of communication. You forced many ideas of services that could have been done very easy via SIP to hack-around solutions through web technologies or similar. Just because it was not a "heavy, well know guy in the gang" doing it.

I may be nobody or somebody in the SIP world now (provided the summary above) and can send an email justifying my interest in SIP and IPv6, that I am not a spammer and beg for acceptance. I'll eventually get it. But if I was just an young guy with good ideas with no experience in the field, just knowing what would make my life easier with VoIP? Should I fax photo ID with SSN and last bill to my Telco?

You, Telco guys, are here for ages. What you brought us lately? Nothing. All SIP desk phones, TDM phones and mobile phones (up to very few years ago) have more or less the interface from 1960 - 10 digits dialpad. You always claimed it is what people want. Right now, you, the vendors, don't how how to get out quickly your touchscreen devices (concept which was brought to the market by a non-Telco-related company) or, you, the operators, how to start social interaction services. Cheaper rates and any kind of innovation in telecommunications lately were because of open source, better and feature rich VoIP switches, plus the new alternative communication streams, don't even think to claim any of those.

You are not capable to change the trend of becoming a bunch of wires and antennas. Soon you won't be even useful for that, as major Internet companies are laying down fiber and power up wifi and wimax networks. I travel a lot, I rather pay 5 bucks for 2 hours internet, check the email, do some voip, video and messaging, than paying you 1 dollar roaming fee per minute.

Your end is coming unless you open your minds and let new, young and enthusiast people coming in and you listen to them. They know what they need, they build their future right now.

The other day experience shows you consider everyone else irrelevant, a bad guy or potential enemy. You could have banned me with my first spam message. You could mark on mailing list settings first (or all) post by 'untrusted' members to be moderated. No, my email was not known to moderator, I am high risk spammer for a 2-hours-ago started list. How many were registered? 10, 20? It was only the welcome message in the archive at that time.

I am not going to try joining again, with less relevance for the benefits of that mailing list. But how many brilliant minds were kicked out from Telco environment because the old dinosaurs in the key positions spending investors and share holders money love their commission with expired vendors? You stay on your private yachts now and need just to call secretary to ship cold champagne, thus, by now smaller, bricks with 10 keys dialpad are more than enough, it would have been even better with ONE KEY. But the rest need to properly interact and communicate.

Fortunately there were and still are THE OTHERS, those rejected with stupid reasons that made no point to retry in the same direction. And they succeeded, my SMS is now Twitter, Facebook & co, my voice (and video and presence which are yet years away to your offerings) is now my own SIP server most of the time, for the rest I have cool options such as Rebtel, Truphone, Gtalk or Skype.

I no longer want to be +123456789, I have a name and cool email address everyone important to me knows and it is easy to remember for them. Btw, if Enum would have made it out, being the unique 'contact address' for all communication meanings, were you able to detect faster that my Enum number belongs to a person with real interest in SIP and IPv6? Would you have been capable to do copy of the number and DNS NAPTR lookup faster than a copy of the email address and web search? Pathetic.

Guess what Telco guys, we were the others, those isolated, but now it is your turn to be. And we will make sure that you change for better or vanish!

PS. The guys behind IPv6 mailing list at SIP Forum might be good people, this is not something personal to anyone. The debate here was about the policies of such organizations, about the management thinking: a walled garden environment, always killing evolution, thing which is present in most of telecom related organizations.

Monday, May 2, 2011

SIP:Provider CE v2.2 RC1 is out

SIP:Provider CE, a complete open source VoIP provider servicing platform, released v2.2 rc1. It uses Kamailio v3.1 for core SIP routing, Asterisk for voicemail and SEMS for B2BUA applications. See full release notes at:

This version runs two instances of Kamailio, one acting as load balancer and the second for SIP registrar and proxy services.

Among SIP:Provider CE features: user web management interface, administration web interface and monitoring, call forwarding, click to dial, call blocking, speed dial, postpaid billing engine with individual billing profiles, peering, least cost routing, multi-domain, voicemail, IVR and topology hiding. See more details at:

Tuesday, April 26, 2011

San Francisco, Silicon Valley, Trip

I am going to spend few days in San Francisco, during May 06-11, 2011, visiting businesses and friends around.

If you want to meet and chat about Kamailio, SIP Express Router, SIP:Provider Platform or SIP and VoIP in general, drop me an email at
  • < miconda [at] gmail.com >.

Monday, April 25, 2011

Two technologies striking back in VoIP - part I

Living a Deja-Vu?!? Or maybe this time the dreams (of IETF/ITU/ETSI specs writers) come true ... who knows. This is a two parts story, about two technologies which many thought went dead long time ago but they are, again, actual in VoIP: IPv6 and IMS (Internet Multimedia Subsystem).

IPv6, the old story is on first pages again. Back in 2002 I worked at FhG Fokus Research Institute in Berlin, developing SIP Express Router (aka SER), within a majority of European Commission funded projects focusing on IPv6.

One may think this is not strictly related to VoIP, well, it is then at least very related to SIP, the leading routing protocol in VoIP these days. Why? Because SIP was actually designed for IPv6 networks. Why? Don't ask me.

Just slightly off-topic note, to refresh, for the past 10 years we had to invent, smash, re-invent solutions and do workarounds to get VoIP going through IPv4 networks -- or more specific saying, through NAT routers, since NAT was not considered in SIP specifications at all. Mixing STUN, TURN, COMEDIA and RTP relays, slipping on ICE, mangling headers and SDP, we got it working. Not few times, friends know that well, it had to be very specific hack per user agent or even firmware.

Back to IPv6, in Kamailio it has been IPv6 support to the core and main modules since May 2002, during the age of SIP Express Router v0.7.y, that's 9 years ago. Looking at the mailing list archive, it is now obvious the interest in the IPv6 is for the first time serious. Of course, the engine behind is the IPv4 address space shortage. Still there is no significant move towards IPv6 as reported recently. There are many (of us) not that happy to remember hex-based IP addresses, which may not be short at all, by the way.

If I hate (phone, digit-based) numbers, preferring instead SIP addresses (username@domain), think about the love I care for hex-numbers.

Moreover, fixing SIP for IPv4 somehow broke SIP for IPv6. Why? Because now we have to support all the nasty tricks specified to make SIP work for IPv4.

The IPv6 was started about a dozen years ago, initial battle was lost, now it is another one. Will it be the final battle? If IPv6 doesn't win this time, then the war is over for it.

To win, it requires fast and smart moves:
  • first and most important, is the adoption of IPv6 at the access level - core networks and home routes
  • then SIP devices (hard and softphones) with proper IPv6 support
If these two are not solved quickly, IPv4 shortage might be workarounded.

The last idea has a strong reason, non-technology related, but at the same time the biggest enemy for IPv6: the cost to switch to it. Internet is way bigger than 10 years ago, replacing/upgrading every piece running IPv4 now is going to be amazing expensive. Time was not the ally for IPv6 at all.

In the next post in this series it will be about IMS, yet another hot topic these days, might not be for VoIP, but surely is for VoLTE...

Maybe in the future we will get to the same situation for Enum? What about mobile IP?!?

Thursday, April 21, 2011

Kamailio Booth at LinuxTag 2011

Kamailio - the Open Source SIP Server project has a booth at LinuxTag, Berlin, Germany, May 11-14, 2011.

Location is Hall 7.2b, booth 112 (near to the Mozilla project).

Come and meet some of us, we will be glad to chat about Kamailio, SIP Router, SEMS, SIP, IMS and VoIP in general. Depending on the day, you can meet at stand:

We will have a demo running latest version and nice flyers to show the capabilities of Kamailio based platforms.

Monday, April 4, 2011

Kamailio v3.1.3 Released

Kamailio SIP Server v3.1.3 stable is out – a minor release including fixes in code and documentation since v3.1.2 – configuration file and database compatibility is preserved.

Kamailio (former OpenSER) 3.1.3 is based on the latest version of GIT branch 3.1, therefore those running 3.1.0, 3.1.1 or 3.1.2 are advised to upgrade. There is no change done to database structure. On 64b operating systems you may need to update the path to loaded modules in configuration file — see the bottom note for more details.

Resources for Kamailio version 3.1.3

Source tarballs are available at:

http://www.kamailio.org/pub/kamailio/3.1.3/src/

Detailed changelog:

http://www.kamailio.org/pub/kamailio/3.1.3/ChangeLog

Download via GIT:

# git clone –depth 1 git://git.sip-router.org/sip-router kamailio
# cd kamailio
# git checkout -b 3.1 origin/3.1
# make FLAVOUR=kamailio cfg

Binaries and packages will be uploaded at:
http://www.kamailio.org/pub/kamailio/3.1.3/

Modules’ documentation:

http://www.kamailio.org/docs/modules/3.1.x/

What is new in 3.1.x release series is summarized in the announcement of v3.1.0:

http://www.kamailio.org/w/kamailio-openser-v3.1.0-release-notes/

Note: on 64b operating systems, the internal libraries and modules are installed under ‘lib64′ folder (e.g., /usr/local/lib64/kamailio/modules/). If you want to install under ‘lib’ directory, then use ‘LIBDIR=lib’ when compiling and installing (e.g., ‘make FLAVOUR=kamailio LIBDIR=lib install’). If you keep the default lib64, then you may need to adjust ‘mpath’ (or ‘loadpath’) parameter in your existing configuration files and change ‘lib’ to ‘lib64′.

If you want to see what is new in development version (to become the future major release v3.2.0), visit the web page: