While I have been basically offline for the past week, I took some time while awaiting one of my flights home to read some news. That little exercise revealed that the Freeswitch community call this past week featured Phil Zimmermann…
While I am not currently an Asterisk user I still try to stay in touch with what goes on in that realm. Earlier this week Rod Montgomery penned a post at the Digium blog entitled “Top 10 Tricks You Didn’t Know That Asterisk Could Do.”
The post is structured like a Top 10 list. Most of the items included are genuinely useful. However, right at the bottom in the #1 position, he highlights newfound support for very high-sample rate audio, aka Super-Wideband.
For me one of the great frustrations of conference bridges is that they don’t give you the kind of control of audio properties that is commonly found in even simple audio mixing and editing suites. Wideband conference services, like ZipDX, make the conference experience a lot better, but there’s a lot of room for improvement.
Of course, this comes from the perspective of someone who has spent their working life in audio/video production, and only came to voip & podcasting later in life. Yes, a veritable Michael-come-lately. I bring to projects like the VUC the expectation of control that simply isn’t commonly possible. However, that is changing and we can thank the Freeswitch dev team for taking a leadership role in crossing these worlds.
On the mailing list of the IETF’s CODEC working group Jean-Marc Valin made a significant announcement on Feb 4th. It reads as follows:
Hi everyone,
We’d like to announce that the Opus codec is now ready for testing. The bit-stream is now is a “pseudo-freeze”, which means that unless a problem is found during testing/review, there are no longer any changes planned. The only exception to this are the SILK-mode FEC and the stereo SILK mode, which should be landing in the next few days. Considering that these are not critical features, we felt like the testing phase could already begin.
<snip>
Cheers,
Jean-Marc
Please recall that OPUS is the new codec arising from the combination of CELT and Skype’s SILK. It’s multiple operating modes accommodate many different applications, from extremely low-latency high-quality links between production studios, to voice applications on very low bit-rate channels. OPUS brings us the current state-of-the-art in audio codec technology in a royalty-free, open source form.
Sometimes the simplest questions result in the most interesting path of investigation. So it has been with Soljon’s initial question;
I am looking for an IP phone that supports G.722 and has audio inputs / outputs so I can connect it to my mixer. We are trying to connect two studios together for an online radio station. I have yet to find anything other than high end Polycom gear that has something like RCA in/out jacks. Have you by any chance come across anything?
A little over a year ago I lamented the sad state of the telecom realm with respect to soft phones, and specifically wideband audio support in soft phones. In the passing year considerable progress has been made.
Counterpath, true to their word, released retail versions of Eyebeam & Bria that support G.722…at least on Windows. Similar wideband support on the Mac platform is now in beta, or so we’ve heard.