Edvina news

SIPit 28 was hosted by Digium in Huntsville, Alabama, USA the week of April 11-15, 2010. There were 54 attendees from 19 companies visiting from 10 countries, using 40 distinct implementations in the interoperability tests.

SIPit, organized by the SIP Forum, is one of the foundations that make SIP work across vendors and implementations. Twice each year, developers from all around the world meet and test, discuss, learn and fix issues both in implementations and standards. During SIPit events, many bugs in the RFCs – or just missing explanations – has been found. Under the leadership of Robert Sparks, SIPit has become the primary event for all SIP developers. Edvina proudly organized SIPit #26 in Stockholm in May 2010.

See the report for SIPit 28 at https://www.sipit.net/SIPit28_summary

I really miss attending SIPit. I do hope I can attend SIPit #29 in the fall. If you are developing SIP software – clients, servers, devices – make sure you attend the next SIPit with your team!  We have a lot to test, from SIP outbound to SRTP, MSRP, IPv6 and TLS.

Here’s a copy of the presentation delivered by Edvina’s Olle E. Johansson at the Telecom Management Association’s Cloud Computing Congres in Utrecht, the Netherlands in December 2010. (more…)

After each SIPit, Robert J Sparks writes a summary that includes the results of a survey done during SIPit and reports from the multiparty tests. These are all very interesting and show where the SIP developers are in relationship to the IETF work.

The SIPit26 summary shows  an uptake in SIP implementations that support TLS. It also reveals that we’re going to make the automated self-tests that has been created during SIPit available on line. We’ve created self-tests for TLS, Early media and IPv6. Hopefully there will be more tests added to this suite. Thanks to Nils Ohlmeier and Daniel Constatin Mierla for helping me with these!

The next SIPit is not determined yet – SIP forum is still looking for a host, primarily in Australia or New Zeeland. If you want  to know more about what it means to be a host, please don’t hesitate to contact me. SIPit test events are important for the whole business. Read the report and you’ll understand why.

At SIPit26 we began setting up a series of automated self-tests for IPv6, like we’ve done previously with SIP/TLS. We also integrated IPv6 in as many multiparty tests as possible, to see how IPv4 and IPv6 lived together.Some notes and experiences:

  • IPv4-only applications will receive IPv6 in messaging. Even if an application DO NOT support IPv6-native connections, the application will surely get IPv6 addresses in various places in the message. In SIP, a call may traverse an IPv6 proxy before reaching your IPv4 proxy or phone. Via headers will have IPv6 and maybe a record-route header too. All user agents needs to support this. We had at least one crash in a proxy that failed to parse an IPv6 address.
  • Placing an IPv4 call to a proxy that forwards the message to an IPv6 phone without handling RTP traversal leads to issues as well. The phone gets an IPv6 address in the Contact: header and failes to send the ACK properly. This happened with Asterisk. Because of parsing failure, the parser gave up and sent ACKs and BYEs to the wrong address.
  • We did successfully set up calls between IPv6 user agents using IPv6 proxys. The failures happened in the mixed scenarious.
  • When placing a call to a domain that was configured with both A and AAAA records for the SRV records, but only one of them responding, we noticed long timeouts before failover, if that even happened. Many discussions about this followed, which lead to the conclusion that this was a poorly configured domain. Some implementations have hard-coded a preference for IPv4 since IPv6 is mostly used over tunnels and add latency today. This should be user-configurable. An owner of a domain can use SRV record weights to indicate a preference to one or the other protocols, which is a better solution. If you use IPv6 over tunnels, make sure that you separate host records for A and AAAA and have a preference towards the A record hosts in your SRV records.

We do need to continue testing all kinds of migration scenarious to  be able to come up with a best current practise document. SIPit26 gave us many good experiences to build from. I hope that testing continues at SIPit27 with the new SIPit IPv6-o-matic(R)(C)(TM) and the prompts from Allison Smith!

SIPit 26 is over and everything was packed together quickly, thanks to help from sponsors and participants. It did surprise me a lot how quickly we could pack all the phones, cables, tables and equipment together. Three days to set everything up, three hours to take it down.


I think it’s been a great SIPit. Unfortunately, I did not have time to test much myself, but I will take the oppurtunity to test at the next SIPit. On the other hand, we’ve run many successful multiparty tests. Successful tests doesn’t mean that everything works as expected. It’s a good thing to locate new bugs in implementations, to find issues with the standard specifications.  That’s exactly why we run the SIPit interoperability tests.


SIPit26 added a lot of focus on IPv6. We integrated IPv6 in standard tests, we added IPv6 self-tests and we run IPv6 multiparty tests. I’ll write more about the results and experiences learned later.


A big thank you to:

  • The SIP Forum and Robert Sparks for organizing SIPits
  • My co-host, TANDBERG.
  • The sponsors: .se, Ingate, Intertex
  • Our partner, the IPv6 Forum
  • SNOM Technologies that provided all the phones
  • My family that helped me set things up. Lisa printed badges and welcome packs, Erik that helped me building the network
  • Lasse Andersson at Whizom who helped me all week, including setting up everything
  • Electrum Conference in Kista, that took care of us in a good way and handled everything very professionally.
  • All the SIPit partipant companies and attendees. Together we make the SIP world better, promoting interoperability between vendors and products.

Being a SIPit host is an experience I would happily repeat. Join the club and offer to host a coming SIPit. Next up is Asia, possibly Australia. After that it’s back to the USA again. Now, it’s time for me to get back to a normal life and get out in my garden.

With SIPit greetings from a sunny Sollentuna!

SIPit is the main interoperability event for all things SIP. It’s organized by the SIP Forum and creates good feedback to the IETF. Asterisk has been participating in SIPit during many years and in many variants  – videocaps, Marc Blanchet’s IPv6 branch and the standard Digium releases. All these tests has lead to a large amount of improvements for Asterisk and have helped us to build a network with other developers in the business, a network which helps when we have bugs that involve interoperability with these devices or servers. SIPit has proven important for the success of Asterisk, and thus it is also important for  everyone in the Asterisk community. 

Now, when we are working on the next  long-term release (1.8) we really need to test again and make sure that we interoperate properly. New stuff, like Terry’s SRTP branch, my RTCP work and the call completion and caller ID update work needs serious testing. We need feedback to be able to fix the issues with the TCP and TLS support. (more…)

I am working with two very different implementation projects this winter. One is about using Asterisk in large call center environments. Another has a goal of building a SIP network for 15.000 phones in a university.

Asterisk for large call centers – full control

The call center platform is focused on a lot of PBX functionality. Every agent stays connected to a conference session, where customers are connected and disconnected, recording is enabled and disabled, dtmf is used to control various services and both AGI and AMI (the manager interface) are both being used heavily to integrate with a controlling application. Asterisk is in full control of each and every call, all the time, but the application controls the flow of the calls. The dialplan is very small for a large-scale Asterisk installation.

Large scale SIP networks – Asterisk on the edge, providing services

For the university, scaling is important. It’s a plain SIP network, with two computer centers in different buildings. All accounts are managed by LDAP, there’s no account defined locally in the VoIP platform. SIP proxys (Kamailio) rule this network and DNS is used for failover. Many services like call transfer, three-party conference calls and call forwarding will be handled by the phones. Asterisk serves as gateways to the old world, Nortel systems, and feature servers for IVR, voicemail and switchboard. No single server is in control of any call.

The power of open standards and open source

Two completely different designs, both made possible by Open Source and Open Standards. Both of them needs to scale. Both of them needs failover, redundancy and stability. And in both cases, we’re replacing expensive legacy telecom equipment with new platforms that will cost less to operate, that has a higher degree of interoperability and much more functionality than the previous solutions. Open telephony wins.

In a surprising move, Digium in partnership with Edvina today released a new channel driver for Asterisk, chan_tweet. The driver connects seamlessly to several microblogging platforms, including Twitter, Facebook, Laconi.ca/Identi.ca and GSM text/SMS. The main feature of this new module is to add audio and video capabilities to microblogging, making the popular microblogging networks a new platform for VoIP and IP realtime communication.

– “I have seen that the microblogging solutions building on the social network infrastructure have had enourmous unexploited capabilities”, says Mill Biller at Digium, “I’ve used it for a long time both personally and for the company and we realized early that by adding IAX2 support, we could now take these platforms one giant leap forward by adding realtime multimedia. I can now spend evenings chit-chatting in audio and HD-resolution video with all my audience around the world instead of sending short text messages. It’s truly awsome!”

Digium contracted Edvina in Sweden, a well-known company in the Asterisk community and long-term Digium business partner, to build this solution. Edvina has many years of experience in building large-scale IAX2 networks, as well as doing development on the IAX2 support in Asterisk.

– “IAX2 recently was published in an IETF RFC and we’re pushing it heavily in all VoIP forums.” says Olle Johansson of Edvina, “We’re hoping that the IAX2FORUM will get a lot of new members that are willing to adopt this technology for their intranets, microblogging services and VoIP infrastructures. In the coming month, we will present more information about new partners with more than 100K users that are going to switch from old technologies, like Hype, SIP and H.323. All of these protocols failed, either because they where proprietary or simply became too complex. SIP currently has more than 5.000 pages of documents describing all the features of the protocol and there’s no single implementation of all of these to test with. Considering the protocol being over 10 years old, this is a sad story.”- “We’ve done our best to fix the Asterisk SIP channel support for customers, but the customer base has been shrinking as more and more converted their networks to IAX2 and now, there’s simply no one interested in us doing that work. We’ve stated over and over again that the SIP channel in Asterisk is broken and no one can prove us right or wrong, because the protocol is just too complex.”

The Microblogmedia platform

The Microblogmedia(TM) platform, developed by Digium and Edvina, let’s users use any microblogging network to set up multimedia sessions. By compressing an IAX2 call setup event in the microblog message, web browsers and clients will connect automatically peer-2-peer if possible, or through the MicroBlogMediaRelay network that supports seamless NAT and firewall traversal by using automatic IPv6 tunnels.Asterisk 1.6.3, released later this month, will support this feature in the IAX2, H.323 and maybe in the old SIP channel (that is now marked deprecated). There is work on adding this feature to ISDN calls, by using messages in the D-channel for tunneling the IAX2 call setup messages. Digium’s VoxSwitch will support this feature in the next release, planned for q3 2009.

Ending the Hype project

In the same press release, Sock Stevens, product manager at Digium finally acknowledged that the Hype channel driver that was launched at Astricon 2008 will not be released after all.

– “We found only one partner to test interoperability with, and that’s not enough to make sure the channel driver being compatible with the protocol. And the protocol wasn’t published in any RFC at all, or any other document. So we finally gave up. We’re now dedicating resources for the new chan_tweet project and enhancing presence support in our IAX2 solution. With the installed base of IAX2 and the new MicroBlogMedia platform, this will be an even more impressive solution, reaching millions of IAX2 users in the enterprise as well as public sector and homes.”

Technichal factoids

  • chan_tweet is the result of the project labelled “Codename orangepeel” amongst the development team and builds on the new “Pinemango” architecture. This is the first channel driver not connecting directly to the Asterisk core, but to the Pinemango API over Adversion, the Ruby framework developed by Phil Jaysip.
  • The MicroBlogMediaRelay IAX2 platform is an open distributed network that builds on IPv6 and a facebook application, thus using the enormous bandwidth provided for free by the Facebook(TM) platform
  • chan_tweet will be released with the core module in Open Source, but with a license exception for plugin developers to add proprietary modules, like the Wireless Village plugin provided by the 3GPP project and the Unistim Microblog Solution by Nertol Networks.

For more information, please do not contact Digium sales.

To be released: 2009-04-01

During 2008, I worked with my Portuguese business partner Wavecom to build a large installation of Asterisk and OpenSER in Portugal. The network is now running 300 servers, 250 of them running Asterisk, FreePBX and OpenSER. It’s handling 200 connections to legacy PBXs of all kinds and over 10.000 phone numbers. During 2009 we’ve developed a failover solution, to make sure that every FreePBX server always stays up, regardless of the state of the hardware.

We’ve migrated 34 Universities to use Open Realtime Communication with Open Source. They are now migrating their user base from old PBX phones to modern VoIP phones.

Want to learn more?
Come to Amoocom in Germany, May 4-5, 2009 and listen to my talk!   And if you need help with similar large-size projects, contact me on info@edvina.net!

BOB 2.0 - the conference.The world of realtime technologies on IP networks and the Internet is cooking with ideas, solutions and innovation. Combining it with social networks, and even more is happening. Protocols like SIP, XMPP/Jabber, SIMPLE, Jingle and Open Source solutions like Asterisk, OpenSER, Ekiga, Sofia-SIP, ReSIProcate and others are changing the landscape. It’s high time to get a helicopter view of what’s going on as well as detailed information about the building blocks for this new service delivery network.

Edvina.net is gathering leading experts in this area to the conference The IP Communications Technology Summit 2008 in Stockholm, April 2-4. Tutorials, hands-on labs, conference and user group meetings. Book this date in your almanac today! More information coming soon. If you have an idea or a talk proposal, send e-mail to me, Olle E. Johansson, at oej@edvina.net today!