Showing posts with label freeswitch. Show all posts
Showing posts with label freeswitch. Show all posts

Monday, November 6, 2017

ClueCon Weekly With Fred Posner

FreeSwitch project is running a weekly video conferencing call for many years now, named ClueCon Weekly. Kamailio project has been present several times in the past.
The edition from Wednesday, Nov 8, 2017, has Fred Posner as a guest. Long time Kamailio community member and advocate, Fred invites you to attend the session and be part of the discussions about Kamailio and FreeSwitch, how to use them together for building modern real time communication systems.
How to connect to the conference call is detailed at:
For video you need an WebRTC capable browser, but there are good options to connect only for audio, via SIP or even from PSTN or mobile phone — very convenient ways to listen while still working.
Thanks for flying Kamailio!

Monday, July 24, 2017

ClueCon 2017

As every August for more than a decade, the ClueCon conference takes place again in Chicago, USA, during August 7-10, 2017. Mainly organised by the people behind FreeSwitch project, ClueCon is among those events where you can meet a large group of experienced people with open source realtime communication technologies.
Kamailio will be present with a talk by Fred Posner, one of our very active advocates of the project in North America and not only. His presentation is scheduled for Wednesday, August 9, at 10:45am:
Besides Fred, you will be able to meet other relevant members of Kamailio project or friends from the VoIP world! If you haven’t checked the event so far, do it very soon and consider to register, no much time left and ClueCon is always worth attending! Read moreabout the event at:
Enjoy!
Thank you for flying Kamailio!

Tuesday, October 20, 2015

Kamailio and FreeSwitch – Cluecon Weekly Conference Call

On Wednesday, October 21, 2015, at 17:00GMT (12:00CT, 18:00 London, 19:00 Berlin), the Cluecon weekly conference call will focus on Kamailio and FreeSwitch. I will be joining the call, answering the questions about Kamailio and its options to integrate with FreeSwitch. Expect the FreeSwitch core developers to be around to handle the questions about their project.
Participation is open for anyone, you can dial in for audio or video sessions using a SIP phone or webrtc capable browser:
More dial in options (PSTN) are presented at:
Enjoy!

Friday, September 26, 2014

KazooCon 2014

Kazoo, the open-source, scalable, distributed, API-driven, cloud-based telephony platform using Kamailio as SIP routing engine, is having its annual conference in San Francisco, USA, during Oct 6-7, 2014 – KazooCon.
It is one of the conferences that must be attended if you are looking to start your own telephony services in short time, based on top and modern technologies.
Upcoming major version of Kamailio, 4.2.0, expected around mid of October, includes the Kazoo connector module, making deployments of Kazoo easier to synchronize with upstream Kamailio packages.

Friday, March 23, 2012

ITSPA Awards 2012 – Open Source VoIP Projects

ITSPA UK has unveiled the winners of its 4th annual Awards, an event designed to celebrate innovation and best practice in the VoIP industry.
The event took place at the House of Commons Members Dining Room, Palace of Westminster, London, on 21st of March 2012, hosted by Dr Julian Huppert MP, Vice Chair of the Parliamentary Internet Communications and Technology Forum.

With main focus on awarding IP Telephony businesses in UK, this year they introduced a new category, “Members’ Pick Award”, to endorse something or someone that has provided real value to VoIP Industry. Open Source VoIP Projects as a group was introduced in this category, made it do the final and ultimately won the category.

I (Daniel-Constantin Mierla, as co-founder and member of management board of Kamailio SIP Server project) attended the event and was selected to pick up the award.

The members of ITSPA acknowledged the major role of open source VoIP projects for their businesses, many of them would have not existed without these projects. Awarding Open Source VoIP Projects as a group was decided because most of the deployments combine several projects to build a complete IP telephony platform. It is a common practice to mix applications such as Kamailio/SER, Asterisk or FreeSwitch to build large VoIP systems and provide a broader range of services.
It is yet another confirmation of reliability and quality solutions provided by Open Source environment for real time communications.

This is also an opportunity to send best wishes and regards to all the people behind Open Source VoIP Projects, developers or community members, that dedicate work and time to develop and improve the quality of the applications and act in the true spirit of Open Source: freedom and fairness!

Wednesday, June 1, 2011

The race to the new and world wide telco

Globalization is everywhere. Still one of the most crumbled markets is the telecom world (classic telephony or mobile). But we are tired of changing SIM cards as we land, we are tired of remembering to redirect numbers when we leave the office, we are tired to memorize all our phone numbers, someone's gonna fix it ... lots of money still there and the seat of a world wide telco is yet free.

The race (could be called war as well) started: Apple, Facebook, Google and Microsoft moved troops, stroke and changed the strategies. We know who is top search engine, top gadget maker, top social networking platform or top operating system vendor, but who is going to win the top world-wide telco crown? Time will decide, first for now just a quick look at top fighters.

Microsoft

Being still hot, let's look first at Microsoft and its acquisition Skype.

Deal was agreed, but it is going to take some time to get approved by authorities. This period is a big waste of time. Besides that, the whole eco-system is a mixture, fitting pieces (i.e., Microsoft applications with Skype communication) together to build a puzzle that was not designed for this goal from the beginning it is going to cost more time and quite some money in development and deployment.

Another weak point for this team is poor presence among end user mobile terminals. Windows 7 for mobiles is at its very young age, iOS and Android have far more third addons that significantly balance the end user decisions of what to buy. Moreover Microsoft does not have a branded hardware yet that attracts customers. Buying Nokia can bring them the infrastructure, the knowledge to build mobile hardware and more distribution channels, still the 'device' is not there since Nokia does not have it either. Therefore more lost time.

Many presented their friendly relation with Telecoms as a strong point, I see it completely different, we talk here about taking parts of telecom cake, so it can turn against rather than help them. I expect Telecoms reducing Microsoft's revenue on other channels (desktop and server OS, productivity applications, a.s.o.) once it starts direct competition.

Apple

The company is lone rider. Does cool stuff, good looking devices and rich user experience that everyone want, but for the other companies is hard to do business with them. Apple has a 'good' reputation of changing the rules in the middle of the game as they like.

They are well positioned regarding mobile devices with iPhone and I would count even iPad and iPod Touch. The upcoming iCloud can provide the needed infrastructure to run telecom service.

Facetime service looked interesting, still none of my close contacts switched to use it as primary communication channel. I don't have figures about their subscriber base, but can be the seed for the telecom plans.

Google

By using Gmail ID Google solves quickly the addressing space with unique user ID in a convenient way. The subscriber base is big enough to be very appealing to use the service or interconnect with.

With GPhone and especially with Android OS for mobile phones and pads, Google is extremely well positioned at this moment, recently outselling iPhone.

Google offers GTalk and Google Voice products for voice communication, too separated so far in my opinion.

Facebook

The famous social networking relies primarily on its user base. They realized that lack of a public Facebook unique ID is working against them, so recently they added own email service and try to force users to choose their Facebook address.

End user controlled device is missing completely right now, there were rumors about a Facebook phone, nothing for sale so far. Their messaging system is open for federating through XMPP, an open standard, but internally it is different.

My thoughts

Money is not a big issue for any of these companies, what matter is the time and the immediate gain of taken decisions.

What I would like to see from the new telecom model? Freedom in communication and mobility for users, plus a proper mapping of the service on the Internet architecture and use what drives the Internet and made it famous (the DNS). What I expect?
  • be able to choose my contact addresses (what used to be phone numbers), where people can call me. I am Daniel, not the only Daniel in this world, but I am the one at Kamailio project, so daniel@kamailio.org can uniquely point to me.
  • be able to interconnect easily with my own brewed telco system. Look at how email is working - Internet domains (DNS) for routing.
  • be able to migrate easily my own brewed service to world wide telco's infrastructure (porting my addresses) and the other way around. DNS is again straightforward the solution
  • be free in the content of the communication, no restrictions on media type imposed by core network
At this moment I see SIP, the open standard protocol very popular in Voice over IP, as the right direction. Yes, I am a SIP guy, but it is hard to argue against the next facts:
  • SIP has a very large end user equipment base in place - SIP is deployed by many operators and present in form of hardware or software to hundreds of millions of people, probably more than Skype has as Tim Paton stated. That means if one of the companies starts a very appealing SIP service, it costs nothing to all those owners of SIP devices to join the club - a huge market already prepared.
  • Interconnection still drives a lot of money in telecom - a trusted world wide company that can offer direct SIP-to-SIP interconnect at cheap rates can secure good revenue from existing SIP-based VoIP operators just by offering a bypass of PSTN.
  • SIP was designed with IP networks in mind, therefore it has email-like addresses and DNS in the core of the protocol
  • Native extensions for end-to-end (proxy model, see below) Text Messaging and Presence
  • It works with any real-time media streaming communication - voice, video, desktop sharing, a.s.o.
Using SIP is not easy though, has to be done carefully, the most important aspect is to forget what a classic Telco does and design a new telecom model:
  • forget about implementing the intelligence in the core network - back-to-back user agent model does not scale to world-wide telco target and imposes restrictions on type of communication. Proxy model scales much more better and requires only the basics of SIP - just interconnect people and operators.
  • stick to basics of SIP - specifications that worked always and they are very simple to implement. If your engineers cannot design the service using less than 10 SIP related RFCs (3 being the core of the protocol), fire them, is going to be too much of classic telecom.
  • let the end device be in charge of communication - people like smartphones, they pay a lot for such devices and then want to use them to full features set
  • charge based on time or volume of data, not type of communication - I may have my new SIP application doing 3D holograms, don't hunt the content type to rise the price, it is a set of bits flowing around for anything, period
  • TLS as primary transport layer - protect people against ISPs blocking VoIP, ensures encryption of content (texting and presence) and privacy (who is calling who). Moreover, for Interconnect services will provide good authentication of peers, without the overhead of setting new access rules for any new server a company is powering up, removing as well the risk of SPIT
  • no allocation of random numbers - provide the option for people to choose their contact ID, email-like addresses is something people are familiar with and remember easily
Regarding the technology, if there is nothing developed in-house for the infrastructure part, using open source would bring real advantages. From the applications I am familiar with:
  • Kamailio SIP server for proxy - hard to beat scalability for routing SIP, including secure TLS communication from end user to core network or for interconnect, plus SCTP as a good choice for within core network communication
  • a good range of software for dedicated Voice application servers, e.g., Asterisk, Freeswitch or SEMS
  • gazillions of extensions already in place for add-on services and interaction with social networking or other IP services
  • access to source code to develop further enhancements in-house for new needs and be able to integrate with the other services offered by third parties or the company itself
How about you? What would you like to get from a new world wide telco? Which technology would be today the best to choose? Do you think of any other company able to run in the race?

Wednesday, May 18, 2011

Remarks about LinuxTag 2011

This year I attended the last two days of LinuxTag show in Berlin. Kamailio project had a booth at exhibition hall for the four days, but I arrived late for it from the Silicon Valley trip.

However, it was just in time to meet with most of the people that took care of project's booth. In a rare moment, due to spread across the world, there were 5 of the 11 members of Kamailio management team.
From left to right: myself, Elena-Ramona Modroiu (Asipto), Henning Westerholt (1&1), Raphael Coeffic (Tekelec, developer of SEMS project), Andreas Granig (Sipwise) and Carsten Bock (Telefonica/O2).

Henning had a presentation about Linux at 1&1, including remarks about usage of Kamailio. Carsten did the fast track, two slides presentation of Kamailio project.

The number of visitors was balanced during my two days there, Friday and Saturday, with waves of crowds. As usual, many of the well know open source projects were there, such as famous Linux and BSD distributions, CMS projects such as Drupal or Typo3, Mozilla, a.s.o.

Very interesting for us, we learned that the German Federal Office for Information Security (BSI) sponsors an integration project that aims to provide a secure, scalable and easy to use open source PBX platform when they presented the upcoming "Gemeinschaft" PBX version 4 that will use Kamailio and Freeswitch.

Yet another confirmation that secure communications, including SIP over TLS, will have a relevant boost this year in VoIP.

Inside our booth, we offered demos and presentation of various Kamailio use cases:

Tuesday, May 10, 2011

SIP for Skype

Let's put it straight. Skype was the simplest telephony service ever. Period. And that's perfectly ok.

Frankly, building a Skype-like service is damn easy. Hopefully, now since it is gone and not anymore a potential easy-to-buy target, it will wake up some people in decision positions from the major telephony operators.

It's just lately Skype added video group chat, but what was the rest?
  • audio conversations directly between two people
  • instant messaging between two people
  • audio group chat (audio conferencing)
  • multi-user chant (group chat)
  • desktop sharing
  • contacts list (buddy list) with presence states (offline, online with sub-statuses like away, do not disturb, a.s.o.)
  • secure communication - encryption for all cases above
  • interconnect to other networks (PSTN and SIP)
In the core network were just the group of authentication servers with the storage for accounts data (user profiles) and the gateways to break out to PSTN and SIP. There was nothing else done in the Skype core network, everything was done in the client side, offering peer to peer communication model. To localize and make possible to have the conversation with the called person, some instances of the client applications were acting as supernodes, guiding the signaling and media streams.


That's all folks!

Now, let's look at the SIP side. It was designed to be able to establish peer to peer real time communication (RTC) sessions over IP networks. Here is the classic SIP trapezoid:



The difference to Skype implementation is in essence that the core network servers do location services along with authentication and storing the user profiles. But the rest is practically identical as concept. Simple and straight.

What it takes to build a service such as Skype with SIP?

Here is the same bullet list saying if it is possible and how, exemplifying with open source applications (therefore very cost effective to design the service and deploy the components):
  • audio conversations directly between two people - yes, the very basic concept in SIP. Client applications, lot of them, e.g.,: Jitsi, Ekiga, Linphone, ... Server applications: Kamailio, Asterisk, FreeSwitch, SEMS
  • instant messaging between two people - yes, the end to end model using SIP MESSAGE request, widely supported. Client applications: Jitsi, Ekiga, Linphone. Server applications: Kamailio, Asterisk, FreeSwitch
  • audio group chat (audio conferencing) - yes, most client applications can do 3-way conferencing (small conference of 3 people), then Jitsi can do multi-user conferencing. In addition, instances of applications such as Asterisk, SEMS or FreeSwitch can be used as dedicated conference bridges, where anyone with a SIP capable phone can dial in and join group conversations.
  • multi-user chat (group chat) - yes, server side with Kamailio in a not standardized way, just reusing the SIP MESSAGE to carry the chat content and imc module to control the members of the chat room. Client applications can be also the host of multi-chat room.
  • desktop sharing - yes, use Jitsi as client and Kamailio as server
  • contacts list (buddy list) with presence states (offline, online with sub-statuses like away, do not disturb, a.s.o.) - yes, client application: Jitsi. Skype model is SIP end-to-end presence model which is very simple from server side application point of view. The SIP end-to-end presence is implemented in many other SIP client applications. Server side: Kamailio. Contact list can be managed locally or via XCAP. Jitsi is a very good client application for both. When XCAP is used, in server side you can use Kamailio with embedded XCAP server.
  • secure communication - encryption for all cases above - yes, via TLS for SIP and SRTP for media stream. Using Jitsi and Kamailio can be an example of doing such communication. Servers can run on different ports that will help going through firewalls and the TLS will ensure nobody can detect it is SIP.
  • interconnect to other networks (PSTN and SIP) - yes, numerous SIP-to-PSTN gateways or termination providers
Actually, looking at the points above, SIP as it was at its early stage of specifications (call setup, instant messaging and presence) is far more closer to Skype communication model than the latest IETF/ITU published specs in regard to SIP.

Where is the problem in IP telephony? Why no big telecom which had the infrastructure and the financial resources could do any revolutionary step in IP networks? Because they don't think simple and don't look at what is important for the most of the users nowadays. All that matters so far for operators (based on my 10 years in the area) are the roaming/inter-connect fees and the set of 500 old PBX features, which are not used more than 2 percent anyhow in a daily basis, and 95% were never used probably.

The change for a proper evolution would be very simple technically, practically is the hardest: because that's about the change of mentality and concepts.

First, stop pushing new SIP specifications with very complex architecture! They don't help neither implementers nor operators. Roll back to the basics of the SIP, it is really a good protocol for RTC. Forget about compressing signaling, bandwidth issues, gran'ma love to 10 digits dialpad, battery life, a.s.o., they are the false problems for you right now (btw, gran'ma loves to see gran'kids on the screen). Get out simple to use services, bring on board new customers with that. Start listening to young and enthusiastic minds, they know what they like to make their life easier, they have the future ahead. Just as that: build the services for your kids.

If you keep talking only with the classic telephony vendors, then the chances to evolve and develop new ways of communication through your companies are close to zero. It could be the Skype protocol "the one" for the future of telephony, it could be XMPP, or SIP can stay in for that, but that is the least relevant aspect.

The winner will be the proper model, not the technology behind it. Definitely the classic telephony is not that!

Tuesday, November 30, 2010

FreeSwitch as Media Server and SBC for Kamailio 3.1

There is a new version of step by step tutorial about using FreeSWITCH and Kamailio together for large VoIP platforms.

Besides upgrade to use latest Kamailio major stable release, v3.1.0, there are couple of new features added in the architecture of the VoIP platform:

  • along with providing media services (voicemail, conferencing, a.s.o.), FreeSwitch is used now also as SBC for topology hiding and media relay (this helps for transconding needs, playing audio messages during the early session or NAT traversal)
  • Kamailio config has added features to detect DoS/DDoS and scanning attacks, secure SIP communication over TLS, IP authentication and a bunch of neat things that help for an easier maintenance and update of parameters such as database connectivity details, local IPs, etc.

The link to tutorial is:

Tuesday, June 15, 2010

Presentation at ClueCon 2010

I will give will give a presentation about using Kamailio and FreeSWITCH together to build large unified communication platforms at ClueCon 2010. While the primary communication channel these days is voice, instant messaging and presence gain more users day by day.

ClueCon is an annual 3-Day Telephony User and Developer Conference bringing together the entire spectrum of Telephony from TDM circuits to VoIP and everything in between. The event takes place in Chicago, IL, USA, during August 3-5, 2010.

See more details at:

Looking forward to meeting you there!

Saturday, May 29, 2010

Kamailio and Freeswitch Integration, Jun 2, 2010

On the 2nd of June, 2010, I will speak at FreeSWITCH Weekly Conference about integration of Kamailio and FreeSWITCH.




The draft of agenda is:

  • Goals of Kamailio, how it differentiates from FreeSWITCH and why using them together creates a very powerful framework to build large VoIP systems.
  • Kamailio for sip routing and FreeSWITCH as media server (conferencing, voicemail, IVR, announcements)
  • FreeSWITCH as a B2BUA for Kamailio (topology hiding, transcoding, call interrupt detection)
  • FreeSWITCH as prepaid engine for Kamailio
  • Load balancing FreeSWITCH servers with Kamailio
  • Q & A Session

You can find more details at:

Details to connect to audio conference (time: 1800 UTC / 1200 CST):
  • Dial: sip:888@conference.freeswitch.org
  • Dial: +1-919-386-9900
  • Click on this link
  • Skype call to: "skypiax5" - you'll be automatically connected (max 20 concurrent users).

Tuesday, April 6, 2010

FreeSWITCH added licensed G729

Expected for some time, FreeSWITCH announced the availability of G729 codec. It was an important piece missing, although everyone looks at wideband codecs for the future, still most of PSTN gateways like the old narrowband G729.

Coming on 1st of April, some were reticent not to be a "specific story" for fools' day. But it is real, at a fair price of 10USD per channel.

More on FreeSwitch site:

Saturday, January 16, 2010

Kamailio and FreeSWITCH - LCR and SBC

Several days ago I came across this interesting article describing how to use Kamailio as carrier grade Least Cost Routing engine and FreeSWITCH as SBC, it is almost one year old, but very well maintained.

First it shows how easy is to integrate both applications to solve demands. Then, although some features offered by those applications overlap, their main target differ, therefore they complete rather than compete.

I am using FreeSwitch pretty much for any new b2bua and voice related application I have to add to my SIP servicing solutions, but still when comes to SIP signaling handling, Kamailio is the king, no mater is about SIP packets mangling, registrar and user location, load balancing or least cost routing.

Overall, it is important to value the blending of open source applications, no matter is Asterisk, FreeSwitch or Kamailio, to build outstanding services and solution. I do not believe in "one size fits all" and most troubles reported out there to each of those projects are due to misuage - trying to do everything with one application, even for functionalities they were not designed for.

Monday, July 27, 2009

SIREMIS v0.9.3 Released

A new version of SIREMIS Web Management Interface for Kamailio (OpenSER) and SIP-Router.org is available as v0.9.3. Among new features:
  • communication with SIP-Router.org and SIP Express Router (SER) via XMLRPC (RPC command interface)
    • pear package XML_RPC must be installed
  • communication with FreeSwitch via event socket
    • no extra library required, implemented as library using TCP sockets
  • support to print rich html formatting of MI, XMLRPC and FreeSwitch command response (config option)
    • makes use of rich text editor features, output can be formated using the HTML languages (previous version displayed the output as text plain in a textarea)
  • stand-alone PHP classes for MI, XMLRPC and FS Event Socket, making further development easier
    • the code for communication with SIP applications is now separate of the view php code
    • php classes can be easy pulled and used in other PHP applications
  • output of commands can be filtered for nicer display (e.g, see FS help, xmlrpc MI commands)
    • current architecture makes easier to highlight parts of the command output, now just couple of commands benefit of pretty formating, the others print raw format (e.g., xmlrpc response)
  • MI, XMLRPC and FS Event Socket groupped under Cmds Panel tab
    • dedicated tab for external commands
  • updated phpopenbiz core
  • fix for re groups db table
  • link referencing on username between group view, db aliases view and subscriber view
    • click on an username value jumps to subscriber view matching that user
  • more options to sort the views using column names
    • more column names become clickable to sort the records
  • auto-suggestion for group name when adding a new record
    • auto-completion as you type the group name

Siremis Project homepage:

http://siremis.asipto.com

Download and installation steps:

http://siremis.asipto.com/install/

View screenshots:

http://www.asipto.com/gallery/v/siremis/

Demo site (it works on a database with random data, username: admin, password: admin):

http://siremis.asipto.com/demo/

Tuesday, July 21, 2009

Blog by MiConDa

I started this new blog by importing the stories of one of my active blogs:
http://openser.blogspot.com

The plan is to unify my blogs here, therefore shortly I will import entries from:
http://miconda.wordpress.com

OpenSER blog is no longer appropriate as the project changed the name in Kamailio, last year, July 28. More over, development of the SIP server project is hosted by sip-router.org , making possible to run Kamailio (OpenSER) and SIP Express Router (SER) extensions on the same SIP server instance. Kamailio 3.0 (planned for late September this year) will be a boost in performances and features.

The other blog hosted by wordpress.com was kind of mirror for most important news about the SIP server projects and stories about project environment, linux and life. Not much active and pretty unsync'ed. So I decided to unite them.

Why "by-miconda"? Well, miconda is taken (not active though) on blogger.com, but is the nickname I used so far. Blogger offers the export blog functionality, thus I could import entries from OpenSER blog in a minute. Importing them in wordpress.com would have taken ages. On the other hand, the stories to be imported from wordpress.com are few and can be done with copy&paste.

blogger.com offers option to rename the blog, but openser.blogspot.com worth to stay around a while.

These being said, expect here: