Observations of the vMix 17 Public Beta

Last week saw the release of the vMix Fun Time Live Show for March which was punctuated by the public release of a beta preview of vMix 17. The official release of vMix is being timed to coincide with the annual NAB Convention, which is April 16-21 in Las Vegas.

In the middle of 2015 vMix replaced Wirecast as my preferred desktop video production software. vMix is effectively a production switcher. It allows me to combine various audio and video sources in real-time, the results being sent to a Hangout-On-Air or recorded to disk. It handles webcams, graphics, animations, video capture cards, live desktop capture and even PowerPoint files with ease. Further, it does so while being less hardware intensive than its competition.

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Freeswitch En Route To Support For G.719 Codec

VVX_1500_D_Freeswitch.jpgThink back to the handful of new audio codecs that have been released over the past few years; CELT, SILK and Opus to name a few. Then there are the handful of proprietary codecs that have become available under more attractive licenses. Polycom’s Siren family come to mind on that front. In all of these cases I have observed that the Freeswitch development team are typically amongst the very first to implement any new codec.

In recent weeks they have added support for G.719, an ITU standard codec created by Polycom and Ericsson. With a  sample rate of 48 KHz, G.719 is a full-bandwidth codec, supporting a useful audio channel of 20 Hz- 20 KHz. It does so with end-to-end delay of only 40 ms and at bit rates from 32 kbps to 128 kbps. It also supports stereo audio.

Upon completion of the ITU standards process Polycom published a white paper on the codec; G.719: The First ITU-T Standard for Full-Band Audio (pdf).

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Ooma: HDVoice For The Thrifty Consumer

ooma-teloOoma has been around for quite some time. While the core of their service offering is free domestic long distance one you’ve bought the hardware, they have also made some effort to promote improved call quality…all the way to HDVoice.

The companies end-point device, a $199 device known as “Telo”, can be inserted inline with an existing landline, making your traditional home phone both voip and analog-capable. It can also be inserted inline with your internet access. Connected in this manner it provides managed quality of service (QoS) for voip traffic on your network. This is a sensible strategy, well established in many ATA type devices.

Telo is actually Linux-based and runs an instance of Freeswitch to handle its telephony functions. That open source project has consistently moved quickly to deploy new technologies…especially new HDVoice codecs. Ooma leverages this fact in offering what they call “PureVoice.”

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Freeswitch Community Call October 13th: Phil Zimmermann on VoIP Encryption

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 describing VoIP encryption and more specifically his ZRTP protocol. Happily, the recording of the call was put online Thursday.

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.

LADSPA Integrated Into FreeSWITCH

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.

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IETF CODEC News: OPUS Is Ready For Testing

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.

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