Phil is of course one of the leading lights in the world of encryption. The call features Phil speaking plainly and openly about the need for encryption and the manner of its implementation in ZRTP.
The call remains a community call, so it goes off in various directions at times, including a little Asterisk bashing. However, Phil makes a good effort to keep the call informative, making it a great listen for anyone interested in voice security.
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.
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.
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?