UC Strategies: If Not You, Then Who Will Use HDVoice?
It’s Q4-2014 and HDVoice is now largely passé. On that basis one might think that it’s use should become evident, especially amongst the telecom cognoscenti. So I was surprised to hear the most recent UC Strategies podcast entitled, “Connecting the Circuit.” This podcast, a discussion of a new UC service called “Circuit”, was derived from a conference call of leading telecom experts.
Sadly, with the exception of a little music at the beginning, the recording exemplifies the finest narrowband audio traditions of the last century. This is, to my mind, a disappointment. It boggles the mind to think that some of the leading thinkers about UC, are not themselves taking advantage of one of its core features…HDVoice.
The image below is a screenshot of the podcast as displayed in Adobe Audition 3.0, with the display set to show energy vs frequency. The application window indicates that the file is 16 bit, mono, sampled at 22,050 Hz.
The file is in fact stereo, likely the result of adding the music in post-production. I have made this image mono to make it easier to see the frequency scale on the right side of the image.
That frequency scale shows a maximum of 11 kHz, which corresponds to half the sample rate. The image documents the fact that none of the participants present any voice energy above 3.5 kHz.
There would seem to be a natural synergy between HDVoice and podcasting. However, many people don’t seem to make that connection. I’ve read podcasters sharing their experience with ways to edit multiple recordings, each made locally, so that they end up with a full fidelity recording…but only long after the original discussion occurred, with hours spent in editing.
I believe that the current crop of tools, hardware, software and services, readily facilitate production of high-quality podcasts. Leverage a wideband conference bridge like ZipDX, but in doing so make the effort to connect via SIP. Steal the audio stream from a Google+ Hangout, or any corporate UC or video conference system. Heck, get the gang together using a WebRTC-based tool like Talky.IO. As long as you avoid dialing a plain vanilla phone number the result will be vastly superior to the PSTN of old. Since the recording will live on forever, it’s well worth the effort.
I’ve long held that those who promote advanced communications solutions should be themselves making use of those tools. As Marshall McCluhan so famously offered, “The medium is the message.”
The situation with this podcast simply illustrates the reality that convenience trumps all else. In this case, I’d characterize the result as something of a disconnect. Given the topic, perhaps it’s better described as a short circuit.
Comments are closed.
I connect to the ZPDX Bridge with a Polycom endpoint and SIP URL. It said I was connected in Wideband. Perhaps it is the recording technology?
Dave,
ZipDX records the media it receives. It’s a simple process in reality. I would expect that the ZipDX recording would be mono and sampled at 16 kHz reflecting a G.722 encoded source. The fact that the final recording is stereo and sampled at 22.050 kHz, which suggests some post production occurred to add the music. It seems likely that they downsampled the call audio to make all the voices equally narrowband.
Michael
Dave,
I have been able to verify that you were in fact connected in wideband. You were in fact the only participant connected via IP.
That fact suggests that the change happened in post-production as you suggest. It’s entirely reasonable to think that the entire conversation may have been converted to narrowband to make all participants sound alike.
Michael
Although I do tend to use HD for these audio conferences – I must say that HD is not simple. I happen to have a dedicated “bat phone” set up with a URI to ZIPDX – but I would never dial a SIP URI manually on most phones. It’s easier with some soft clients, but these conferences often have 10 people on them, and many are not at their desk. Some cell carriers support HD, but unsure how that helps with a conference bridge. Most UC and VoIP services don’t support SIP URI dealing either.
It’s a shame, it’s a fee technology that should be easier – but it isn’t.
IMHO, the issue is the production quality of the podcast. It remains online a long time. It’s worth the effort to make it polished. It’s not THAT difficult. It just takes some care and concern.
If it’s true that smart phones and tablets are now ubiquitous then just have everyone use a SIP soft phone on their favorite portable device. Bria works great. Suggest a decent wired headset or a known suitable BT headset. Makes the whole process easy peasy.