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!

The document ”The use of the SIPS URI Scheme in the Session Initiation Protocol (SIP)“ is now approved by the IESG as a proposed standard. This is an update to RFC3261 that clears up a lot of the issues that has been open in regards to the SIPS: URL scheme.Now we need some clarifications on the “;transport=tls” use and implementation notes, so that all application developers can start working with fixing the applications.I did not realize how bad the situation was until last SIPit where I participated in a TLS interoperability test.  There where too many opinions on how to implement and support TLS in SIP, and too many non-interoperable implementations. A missing piece in many implementations, is DNS NAPTR support. NAPTR plays a very important role in secure SIP connection setups.   The biggest question still remains: What’s a secure call? For Asterisk, we have to do a lot of work here to implement security in the dialplan. But at least we have one document that more clearly explains SIPS: to work with now!

Garret Smith writes some thoughtful words about WiFi phones in his blog:

Infonetics Research released a report that showed WiFi ip phone sales increase 60% in 2007, with 682,000 units were sold worldwide. The report cited “increased vendor support” as the primary reason for the growth. 

[…]

While some people seem to like WiFi phones, they aren’t for the faint at heart, especially if your aren’t technically savvy. My advice, if you want to go wireless, is to pickup a DECT based solution. A little more expensive, but it works…for everyone. 

I still haven’t met a WiFi SIP phone I liked and used more than a couple of days…  

I’ve worked with Asterisk many years. I started in 2002 when I was working with a service provider here in Sweden, then co-founded Astricon, started the Asterisk Bootcamp trainings and the dCAP. Many years of working with Asterisk, but almost always in combination with a SIP proxy (mainly SER/OpenSER) and in carrier networks.During the last weeks I have assisted a Swedish company installing an office PBX. This was a new experience for me. Of course, I’ve installed Asterisk for my own use and in the trainings, but this time it was a customer with very well-specified requirements. And I enjoyed every moment.

 Asterisk works very well in these enviroments. It’s almost as if it was built as a PBX. Right. Of course. Sorry. Asterisk is built for this. Exactly this. It’s just that in my work, I’ve used Asterisk as a PSTN gateway, conference server, voicemail server, billing server, session border controller, queue server, IVR server and much more. In those cases, we send a lot of traffic through Asterisk and push it to it’s limits. In the Office PBX market, Asterisk has more than enough power and shines. The flexibility is enormous and the things we can do with just a few lines of code is marvellous.

Working with all kinds of issues in the large scale environments, it’s always important to remember what Asterisk is built for and how well it fits that market. I had a lot of fun configuring this PBX, discovering new parts of Asterisk and trying to solve the challenges from the customer. Asterisk really stood up to this challenge and came out as a shining new powerful sports car, replacing the old PBX.

Of course, I came up with a few ideas that would make this easier. I reported them on Asteriskideas.org - go there and check and report your ideas too!   

Asterisk 1.4 not only adds features to your PBX, it also adds enhanced voice quality for VoIP. The new and improved jitterbuffer implementation covers all RTP-based VoIP channels. Previoiusly, only the IAX2 channel driver had a jitter buffer implementation. (more…)

The Asterisk project has a very strict policy in regards to backwards compatibility. Unless we can’t find another solution, we’re not allowed to remove a function between releases. A configuration for Asterisk 2.4 should work in the next release. In order to be able to change functionality we warn users in one release and then remove the functionality in the coming release. So a configuration in 3.0 works in 3.2 but maybe not in 3.4.

This article tries to provide help with known problems with upgrading. Read on to learn how to avoid the traps! (more…)

Asterisk 1.4 delivers many new features. In regards to call state subscriptions, there are many news for you. Call state subscriptions are what makes the lamps blink on your phone when your collegue’s phone rings. In 1.4, you can make it blink based on activity in parking lots and meetme conferences as well. Read on! (more…)

Asterisk 1.4 introduces a new level of Jabber integration, developed by Matthew O’Gorman at Digium. The Asterisk Open Source PBX integrates with Jabber/XMPP in many ways. (more…)

I am proud to report that Alec Saunders report  that Infoworld reports that Yahoo is going to test OpenID. This is an important step for OpenID. I believe OpenID is a very good example of not trying to solve the whole puzzle with one solution, but build a small building block that moves us forward. I’ve blogged earlier about the importance of OpenID and how it relates to Enum and iname and… All the other solutions out there. I need an OpenID speaker for Bob 2.0 - anyone out there?Alec writes:

Infoworld reports this morning that Yahoo appears close to becoming an identity provider for OpenID. This, of course, is the next step in a full implementation. You can already use OpenID to log into Yahoo properties like Flickr, for instance.

• Yahoo to act as OpenID server — Alec Saunders .LOG