On the heels of Kevin Fleming’s announcement yesterday discussing the changes in the Asterisk versioning scheme, we’d like to formally announce that Asterisk 10, Beta 1 is now available for community testing. Asterisk 10, a Standard Support release, will be the next major release of Asterisk and follows the release of Asterisk 1.8 LTS, a Long Term Support release. For more information on the different releases of Asterisk, check out the Asterisk Versions page on the Wiki.
Let’s talk about some of its new capabilities.
A major focus of the Asterisk 10 development cycle was Asterisk’s support for media types. In versions of Asterisk 1.8 and prior, Asterisk supported a rather limited number of codecs due to some architectural limitations. Plumbing was ripped out, kitchens were remodeled, girders were swapped, and Asterisk 10 now has a media architecture that’s capable of handling both a nearly unlimited number of codecs as well as codecs with more complex parameters. What does this mean for users? First, it means that Asterisk now comes with some additional codecs, including the 32kHz variant of the Speex codec (previous versions of Asterisk only supported the 8kHz or 16kHz variants), Skype’s Superwideband SILK codec, and pass-through support for the 44.1kHz and 48kHz variants of the CELT format.
Astute readers will note that earlier versions of Asterisk were only capable of operating on 8kHz and 16kHz sampled audio, and that the aforementioned newly-supported codecs operate at rates other than these. You’re absolutely correct. In order to support these new codecs, Asterisk 10 has also been provided with support for a variety of super and ultra-wideband sampling rates, all of which are supported as file format types for file playback or recording.
One of the best ways to show off the enhanced media capability is with an all new conferencing application. Asterisk 10 features a fully-functional software-driven conferencing application called ConfBridge. Astute readers will again note that ConfBridge was first made available in Asterisk 1.6. Again you’re correct. But, the ConfBridge of yore was sparsly featured and was only capable of operating at 8kHz (or 16kHz with a compile-time option). The ConfBridge of today (Asterisk 10), supports a full range of sampling rates (8, 12, 16, 24, 32, 44.1, 48, 96, and 192kHz) for its participants – it automatically selects the best sampling rate based on the participants’ native codec. Besides that, ConfBridge provides much greater control over in-call menus (customizable on a per-DTMF, per-caller basis), per-user profiles, as well as dynamic conference creation.
Speaking of showing off, what better way to “show” off capability than with some video? ConfBridge in Asterisk 10 provides basic video conferencing support. That’s right, if you and your friends have video-capable SIP devices, that all speak the same video codec and profile, you can create multi-party video conferences. The video played out to each participant follows a marked user; a user chosen by DTMF; or, the most useful, the user who’s currently talking.
In addition to the improvements to voice and video capabilities, Asterisk 10 can also improve your faxing experience. Asterisk 1.4 is capable of T.38 pass-through, where one T.38 capable endpoint can send a fax directly to another T.38 capable endpoint – usually a couple of SIP peers. Asterisk 1.6.X and 1.8 are capable of T.38 termination, where Asterisk can read/write TIFF files from/to T.38 endpoints. Now, with Asterisk 10, transparency between non-T.38 and T.38 is possible, with its T.38 Gateway (aka T.38 Relay) support. So, if you have a Digium E1 card terminated to the PSTN, you can now carry a fax directly from the PSTN to a T.38-enabled SIP device, without the intermediary “termination” step of TIFF conversion required in earlier versions of Asterisk. Or, if you’ve got a T.38 capable SIP provider, and a second provider that isn’t capable of T.38, you can mediate between the two; again, without any intermediary “termination” step. That means much faster and more reliable faxing in Asterisk 10.
What do we need from the community?
Testing, Testing, Testing.
Besides these great features, Asterisk 10 features a number of other important changes, many of which are under-the-hood. All of these under-the-hood changes mean that there are probably (definitely) some broken bits that Digium hasn’t found. So, to find them, we need your help. Please report any issues you find to the issue tracker.
Again, please report any issues (bugs, etc) that you find to the Issue Tracker.
It’s also useful to see successful test reports. Please post these to the asterisk-dev mailing list. Or, if you IRC, then hop on over to the #asterisk-testing channel on the FreeNode network so we can work together in testing the many parts of Asterisk.
As always, we thank you for your participation.