2004 February

February 2004


This week, a lot of new stuff is added to the Zap channel, indicating new products coming soon from Digium. There are rumors of an IAX device and new FXO modules to the TDM PCI cards. Functions for firmware download over the IAX protocol where added, as well as support for FXO in the kernel drivers for the TDM cards.

Pingtel recently announced that they are releasing their SIP based PBX as Open Source. This is an important move in the Open Source VoIP marketplace. Will this software be able to compete with Asterisk.org and SIP Express Router?

Announcing the emergence of an enterprise-class open source IP PBX

Pingtel’s open source business model will prove to be a fundamentally disruptive force in the $5 billion per year enterprise PBX market. By offering enterprise-class, all SIP-based, open source IP PBX software under a Linux-style subscription license, Pingtel is combining the best attributes of open source development – low cost, adaptability and flexibility – with the reliable solutions and support enterprises require for voice applications. Pingtel’s open source IP PBX software is the linchpin technology that will catalyze the movement of enterprise communications into the data center and away from purpose-built hardware. Like enterprise-grade Linux, this approach will drive commoditization of traditional telephony hardware and software and eliminate technology lock-ins that have plagued the industry for decades. This new business model will lead to pervasive adoption and shift significant market share away from vertically integrated, proprietary-solution competitors.

The SIP protocol for IP communication is elegant. Yet, it\’s partially based on a vision that we live in a end-to-end IP network, which is no longer the case. There\’s an oppurtunity to come back on this track with IP version 6, but the problem is no longer the technology, it\’s the user. We\’ve successfully taught the users that address translation is a natural thing, a broken network is what they should use.
That\’s sad.
…how does that visionary problem affect us? Well, this week I\’ve spent two days working with SIP and NAT problems, media gateways and ways to overcome this broken architecture. It\’s ugly, it\’s not very fun and it\’s resource intensive. The Internet architecture is end-to-end with a passive core network in the middle. The telecom network is the opposit, an intelligent network with very passive an un-intelligent endpoints. NAT forces us in a way to add intelligence to the core, which is not a good thing. SIP should be peer-to-peer.

What you also find out is that even though SIP has a few years of age, the implementations are not very mature. I spent a good part of yesterday trying to find out why some devices couldn\’t place a call, to end up finding out that if I used a specific codec, the phone did not acknowledge the SIP transaction at all, it just gave up. And this is a very common phone on the market. The call dropped to the floor and since I was in the middle of the NAT workaround mode, I blamed the NAT and not the phone for too long.

Hopefully new solutions like UPNP and STUN will help the situation, SIP-aware broadband routers will certainly help. But they\’re not out there today. And when they come, will they be interoperable, will they support the standards in the same way? Or do I have to spend hours testing them?
Regardless of what you do in the world of TCP/IP, interoperability is key. We need SIP interoperability testing and certification now, we need to join forces and change the market like what the WiFi branding did for wireless lans. Now, because one year from now, it\’ll be too late.

After moving and being off-line, the Asterisk bug tracker, the maling list and the main Digium web site are finally back on line. The mailbox is starting to fill up, and the bug tracker continues to cook.

If you\’re mailbox is empty and you can\’t reach the Asterisk CVS, don\’t worry. Digium, the company behind the Asterisk Open Source PBX, is moving to new offices this weekend and they\’re off line.

Take the time to track memory errors in chan_sip, fix the new script you wanted for advance agent rotation based on colour of the Agents baseball hat in queues or convert all your old records (with no copyright) to MP3 for music-on-hold. There\’s a lot to do. The Asterisk-users mailing list will explode your mail system soon, so use this valuable time! The #asterisk irc channel in freenode is still open, meet you there!

The new CVS release tree for the 1.0 release was created last week. From now on, all patches are patched either to the stable tree or the 1.0 tree or both.

The SIP community needs to develop guidelines to create a better SIP network out there. Learning from the SMTP relays, we need to stop \”open\” sip proxys. We need for the proxys to check what domains they are responsible for. And to set the right \”realm\”, not use generic ones like \”default\”.
We need for the UAs and Proxys to not answer SIP invites to IP addresses. But, wait, that adds some overhead for resolving… And how do you address a SIP device then?
I need to check the GRUU papers and learn. So do you :-) Will be back with more information.

Mark Spencer, Digium, today released Asterisk 0.7.2. The new release contains no major new functions, but a lot of bug fixes to bugs found in the 0.7-series of Asterisk. There\’s been some changes in the SIP channel, a few new CLI commands and fixes to the generation of WAV-files that Asterisk\’s voicemail system sends as attachments in mail messages.

The Asterisk developer team is now concentrating on stabilizing the code so the 0.7 tree can be split up in one stable branch and one development branch, heading for a 1.0 release.

If you\’re using a 0.7.x release of Asterisk, please upgrade to 0.7.2 now.