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Edvina launches a new training class for all students of the Asterisk SIP Masterclass and the new Edvina SIP Masterclass – the SIP Update 2013!
This new class will update you on the latest features in the coming Kamailio 4.0 as well as some new features in the SIP family of protocols – like MSRP, SIP Outbound, GIN and much more. The class is a three day class with labs that quickly gets you up to date with Kamailio and gives you input on how to improve your own platforms.
Teacher is Olle E. Johansson, with over ten years of experience of Kamailio and Asterisk, as well as more than 15 years of experience in teaching, working as a consultant building large scale platforms and doing development in both projects.
We have a limited amount of seats in this new class – make sure you register today!
The hidden secret of TCP/IP – and how it affects your PBX
Olle E. Johansson, founder of Edvina, delivered a talk called “The Hidden Secret of TCP/IP – and How it affects your PBX” at the Voip2day + Elastixworld 2012 conference in Madrid, Sept 25th. The talk describes the background for the SIP2012 project and what needs to change in order to get back on track with realtime communications.
The presentation was uploaded to Slideshare and quickly got over 38.000 views, taking it to the top of the week list on Slideshare.net and got many comments, referrals and tweets. A video of the talk will be uploaded to the Voip2day.net site in a few weeks time. Voip2day + Elastixworld was a European Voip event that got over 1.000 visitors during three days, with a very interesting exhibition. Avanzada7, an Edvina partner in the Open Unified Alliance, has organized Voip2day in Madrid for many years.
– “I’m very proud of the reaction by the community – it shows me that Interoperability is still a hot and important issue. Hopefully the SIP2012 project will be able to raise the bar and take SIP away from the old PBX model and move it into the world of Open Unified Communication” says Olle.
Many people asked “What can we do?“. The answer is partly in the presentation – base your realtime multimedia architecture on a modern view of communication, instead of a 50 year old PBX model. Chat, presence, video and other applications, like desktop-sharing, needs to be part of the architecture. Require vendors to implement more than the 10 year old SIP 2.0 specification – follow the SIP2012 project and write better requirements when purchasing SIP products and services. Finally, make sure that your vendors participate in the SIPit interoperability test events with all their SIP products. It’s at SIPit vendors meet to test their products and improve the interoperability as well as the standards.
SIP 2012 :: ICE – finding the best media path between two phones
Edvina continues the work to define a new SIP reference profile, updated to fit current networks and implementations. SIP 2.0 is over 10 years old and the RFC does no longer cover enough to fit the needs of the enterprise. Networking has changed and much has been added to SIP by the IETF in cooperation with the SIP Forum, the 3GPP and the PacketCable consortium. Unfortunately, when purchasing SIP solutions, most customers still only refer to RFC 3261 as the reference specification. This is what SIP2012 is trying to change.
With ICE the best media path – and a working one – is always selected for the call
ICE – Interactive Connectivity Establishment – is now added to SIP2012. It’s a solution for always finding the best media path between two devices – regardless if it’s private or public addresses or if it’s an IPv4 or an IPv6 address. For a SIP call, the caller and callee needs to know that there is a working media path. If not, the call should fail. This is the promise of ICE.
ICE is also part of WebRTC
ICE is also part of the developing standard for adding interactive media – audio and video – to web browsers. With SIP in the web browser on your tablet, you want to connect over 3G or 4G, wifi enterprise networks as well as to the Wifi in a hotel or in a café. ICE handles finding the best media path in these environments, so it’s a natural part of the WebRTC standard. In that project, many additional requirements to ICE has been added, which will show up as new RFCs soon.
Read more about ICE on our SIP2012 project web and watch our presentation of ICE! Follow us on Twitter or Facebook to engage in the debate!
The new Edvina SIP Masterclass – Stockholm and Miami this fall!
Edvina today launches a new training class, based on years of experience of inhouse and public classes in SIP, Asterisk and Kamailio/OpenSER. The new class focuses on the SIP protocol and the Kamailio Open Source SIP server. By combining lessons on the core SIP standard with the Kamailio implementation and labs, the students will learn interactively and get real configurations to bring home. The new class will launch in October in Stockholm, Sweden and in December in Miami, Florida, USA.
– “We’ve updated the training materials during a series of in-house trainings during 2011 and 2012, which means that the material is almost totally new compared with the original class held seven years ago.” says Olle E. Johansson, Edvina’s founder and the teacher in the classes. “We’ve added more labs and more information about SIP presence and security. For students, this is an upgrade that fits everyone that has worked with Asterisk, FreeSwitch or Yate for many years and wants to learn how to scale and add new SIP services to their solutions.”
Experienced teacher guiding the way
Olle has many years of experience in trainings. He wrote the first Asterisk Bootcamp trainings in 2005 and was the author behind the Digium dCAP certification for Asterisk. Prior to that, he has been doing years of training in TCP/IP networks, from network basics to LDAP directories and SSL/TLS solutions. In addition he’s working as a consultant building large platforms with Kamailio and Asterisk as the primary tools. He has been involved in both Open Source projects since 2002, as a contributor, documentation writer, coder and project member.
From SIP to scalable realtime platforms with Kamailio
This class is built for persons that have used the PBX-class tools like Asterisk, Yate and FreeSwitch and wants to learn how to scale and add new applications like presense and instant messaging to their solution.
The class will spend a lot of time on the SIP standards, then move on to how to implement them using Kamailio – the Open Source SIP server – in combination with other tools. After the class, you will not only know how to operate Kamailio – you will also have a lot of knowledge about how the SIP standard works, what to expect from devices and how to troubleshoot your realtime network.
PS: A special thank you to Redfone Communications in Miami that hosts our training in December!
SIP2012 :: Redefining SIP
Edvina today launched SIP2012 – a project to try to redefine SIP in the eyes of all customers to our industry. We need to go beyond the 10-year old RFC and use all the solutions developed by the IETF and the SIP Forum during 10 years of SIP version 2.0. We’re not defining SIP 3.0 – but trying to combine a set all the specifications added during this time into a reference profile that customers can refer to and use when buying SIP devices – phones, servers, chat clients, PBXs and much more.
What’s the current state?
It’s very hard for customers to clearly specify functionality required. If they refer to RFC 3261 as a base RFC, that would include S/MIME which is almost never implemented. RFC is updated by 13 RFCs, added to by quite a lot of RFCs and there are 5 verified bug reports (ERRATA) against the text in the RFC.
An example: A large bid for a new telephony platform
Let’s start the discussion with an example. One large bid from last year specified the following standards as requirements:
- RFC 3261: SIP version 2.0
- RFC 3263: Locating SIP Servers
- RFC 3550: Real-time Transport Protocol (RTP)
- RFC 3551: RTP Profile for Audio and Video Conferences with minimal control
- RFC 3487: Priority mechanisms for SIP
- RFC 2833 or 4733: DTMF signalling
- RFC 4855/4856: Media Type Registration of RTP Payload Formats
- RFC 3611: RTP Control Protocol Extended Reports
- RFC 5391: RTP Payload format for G711.1
- RFC 3711: The secure Real-time Transport Protocol
- RFC 5246: The Transport Layer Security (TLS) Protocol
- RFC 2475: Diffserv
What is your opinion? Do you have a good specification that could be useful for other people? Please share!
Reference implementations are important!
We encourage Open Source projects to support the profile and take some extra steps to build reference implementations and test them at the SIPit test events, organized by the SIP forum. Edvina will participate and help tests by building self-test platforms where possible, as we’ve done for TLS, SIP Identity, early media and much more during earlier SIPit events.
Kamailio and OverSIP Open Source SIP Servers are working on implementing SIP Outbound, the first part of SIP2012. We need clients that support it so we can test it for interoperability. Please join this effort!
To find out more – click here for the SIP2012 web with a presentation of SIP Outbound – the first part of the SIP2012 profile.