At the recent Netnod Spring Meeting Olle E. Johansson from Edvina was invited to talk about his work with dual stacks in SIP. The talk was very personal and covered the progress, or lack of progress, in the IETF and the work done in the SIP Forum IPv6 working group.
The work started when Olle detected strange issues when testing the IPv6 implementation in Asterisk. It wasn’t the code, it was something else. After testing IPv6 at the SIPit events, work started in a working group of the SIP forum that later resulted in documents contributed to the IETF – one issue was adopted by the IETF SIPcore working group but is not seeing any interest so it doesn’t move forward. In the IETF, strange resistance was met and not much support for doing any changes or additions to the protocols in order to fix proven problems. In fact, quite a surprising opposition was met.
How do you fix issues in an old protocol?
Should it really be this hard to fix issues in a published protocol? Are SIP and HTTP the only protocols that have issues with dual stacks or is it just the two protocols that have been tested in detail?
Further tests at SIPit has not only confirmed the documented issues, but also revealed issues in many implementations that has been supporting IPv6 for a long time. Source address selection was missing. There’s a lot of work to do before we can get SIP implementations ready for dual stack use, both in development and in the standards.
The presentation is available on slideshare. We apologise that it is made for live presentation and doesn’t cover all the details on the actual slides, but it may still give some interesting facts.
Tags: ietf, ipv6, netnod, sipcore, sipv6
Want to learn Kamailio? Register today for the Edvina SIP Masterclass in Costa Dourada near Barcelona this summer! This year, we’re outside of Barcelona in the Costa Dourada.
This is the basic class that go through both SIP and Kamailio – in theory and with many labs. All you need is a laptop with a Linux virtual machine and energy for five days of training with Olle E. Johansson.
This is a unique class where you learn how to build a scalable SIP platforms and how to integrate media servers and PBXs like Asterisk or FreeSwitch into your service.
Olle has many years of experience with both Asterisk and Kamailio. He’s been a developer for over 10 years and have done trainings for a longer time in various topics. He founded the Asterisk trainings, wrote the dCAP and co-founded the Astricon conference. He’s a regular speaker in many conferences around the world, as well as in local geek meetings in Sweden. His SIP Masterclasses have been running for a long time and keep changing based on the development of the protocols and the products involved.
This class is hosted by Avanzada 7 in cooperation with Edvina. Contact us today to book your seat!
Tags: avanzada7, kamailio, masterclass, sip
Edvina yesterday evening launched the TLS-O-MATIC test site for applications. For many years, Olle E. Johansson has been operating a TLS-O-MATIC for SIP applications at the SIPit event organised by the SIP forum. The new TLS-O-MATIC is a public service launches with more than 15 tests for the HTTPS protocol. Tests for SIP will be added later.
TLS-O-MATIC has tests for all kinds of problems with certificate and host validation in TLS.
-“When you develop an application it’s easy to test success cases. It’s even more important to test failure cases – especially when it comes to TLS. There are a lot of API’s depending upon TLS for confidentiality, integrity and authentication. TLS-O-MATIC.com provides a test bed for these.” says Olle.
TLS-O-MATIC was launched at the #MoreCrypto Meetup in Stockholm yesterday as a joint presentation by Olle and Daniel Stenberg, Mozilla. Daniel is the developer of Curl and LibCurl. During the presentation, Daniel used Curl to run through the first ten tests.
All the scripts and configurations used to produce the tests are published on Github for review, forking and possible setup in internal labs. Edvina has a tradition of working with open standards and open source, so publishing the Tls-o-matic source on Github is a natural thing to do for us. We have already received the first bug report!
Tags: curl, test-o-matic, tls
Come to Stockholm, Sweden April 7-20, 2015 for the very first “Mastering Kamailio” class!
After more than 10 years of running the SIP Masterclass we had too much material and too many ideas to fit into five days of training. Students attending our SIP Masterclass got a chock when seeing the material – over 1000 powerpoint slides printed in a binder that was propably causing back pain problems when being carried home. We wanted to say so much, to share our knowledge.
This winter we decided to do something about it. We wanted a class that focus more on Kamailio. To get started, one needs to learn SIP so the SIP Masterclass still starts with a lot of SIP tutorials. Without this, students would get nowhere in Kamailio. The solution? We split the class in two. Part one is the same, we just made the binder a bit easier to carry and only print the material that we actually cover in the class. Part two is the exciting news.
Launching SIP Masterclass 2 – Mastering Kamailio
The SIP Masterclass 2 – Mastering Kamailio – is our new class. Four days of labs and tutorials. This is a perfect class for all the students that have been attending our SIP Masterclass during all these years – in the US, in Sweden, England or Spain.
This class assumes you have been running Kamailio, but never got around to learning the more advanced modules, how to do more than load balance a set of PBXs or handle SIP trunks.
In this class we cover dual stack use, security, SIP trunking, WebRTC and much more. We will build stress tests and functionality tests that you can continue to build on at home. We will run thousands of calls in various modes, make things break and learn to debug our platforms.
Kamailio has over 100 modules, so we can’t cover all of them, but we will use combinations of them to build new solutions and you will learn how to use the SIP protocol on a new level.
The adventure starts in Stockholm, Sweden in April
This will be fun. We run the first class in Stockholm after easter – April 7-10. Since this is the launch of this new training and we have so much we want to do, it will not be a perfect class. Labs may not work smoothly, there will be typos on the slides, slides may be out of order (things you notice when you run for the first time). One thing is for sure though – you will learn a lot during this class.
In order to get some
guineapigs first time SIP MASTERS for this first class, we have lowered the price. This class will normally cost 2.600 Euro ex VAT. If you register early for the very first SIP Masterclass 2 – Mastering Kamailio in Stockholm you can attend for 1.950 Euro ex VAT!
Register today by sending e-mail to firstname.lastname@example.org !
Looking forward to seeing you in Stockholm!
Tags: kamailio, masterclass, oej, sip, Trainings
The Edvina SIP Masterclass is the best introduction to Kamailio – the Open Source SIP server – and the SIP protocol that you can get! Hosted by Avanzada7 in Madrid this september we’re offering the class for a special summer price – 2.500 Euro for five days of labs and lessons. Book the flight and a seat on the class now!
Register today to spend one week with Olle E. Johansson, having fun with SIP, learning Kamailio in well-prepared labs and brainstorming about new realtime communication solutions like WebRTC and how it works with your SIP platform.
Come to Madrid – we start Monday September 15th!
The class is in Madrid, the capital of Spain, September 15-19, easy to reach from all over Europe! As usual, it is organized in cooperation with Edvina’s long-term partner Avanzada 7. Seats are limited in this popular class!
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 SIP scalability and functions 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. If you’ve used a SIP PBX – this is the next step!
Learn how to build scalable and resilient SIP networks with Kamailio!
This class is for users of Asterisk and FreeSwitch that wants to learn how to build larger, scalable and open SIP networks with Kamailio – the Open Source SIP server. The class interactively teaches you SIP and Kamailio, building a platform step by step. When you leave the class, you should know much more about how SIP works and how Kamailio can scale your existing solution or be the new platform for a Unified Communication network. Read more about the class and learn about the details in our product page! If you have any questions, please don’t hesitate to contact us at info (at) edvina.net or call our office at +46 8 96 40 20!
- Learn more about the Edvina SIP Masterclass – prices, location and agenda
- If you have questions, send email to info (at) edvina.net or call us on +46 8 96 40 20!