Graves On SOHO Technology

This past week Junction Networks phone lab posted a review of the Polycom VVX-1500 Business Media Phone. They make note of the devices’ many fine qualities. The VVX is truly a joy to use. It’s build quality is outstanding, and you simply won’t find a better sounding phone anywhere.

That said, I remain curious about the use of one-on-one video calling. It’s remains unclear to what extent companies are making use of desktop video calling. It’s not the kind of this that springs up organically since one must first seed the organisation with a number of suitably capable phones.

I wonder if this tends to happen within companies that are already making use of traditional video conference installations? Do the Business Media Phones merely extend the reach of such facilities to to the desktop or SOHO users? Or are desktop video phones something completely different?

If you make use of such devices please leave a comment about your experience.

Here’s the headline:

Zoiper is the First 3rd Party Softphone to Support Skype’s SILK, Polycom’s SIREN and the Broadvoice Codecs

How does that sound? Pretty major, right? Wanna know more? Here are the details.

Given that ZoIPer Communicator is available in web, Windows, Mac & Linux flavors this could jump them to the head of the class for HDVoice support in a soft phone. And of course, ZoIPer Communicator has long supported both SIP and IAX protocols. Yes, you have options. Very nice.

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Having read & listened this far into this series you should now have some grasp of how narrowband (G.711) compares to wideband (G.722/G.722.1) and even super-wideband (G.722.1C) audio for telephony applications. The differences in many cases are quite pronounced, even startling. What you hear in the examples are just the most obvious properties of the encoding, sampling rate and by implication, the available audio bandwidth. It’s worth understanding a bit more about the evolution of the role of the codec over time. This will help you frame up how the Siren codecs fit into the Asterisk realm.

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In this third installment I’ll try to broaden your experience with wideband and super-wideband telephony by exposing you to a selection of recorded audio samples using various encoding techniques.

Until now the examples used were strictly in English. This next set of six samples recordings are in six different languages; Norwegian, Chinese, French, German, Russian & Spanish. Each is presented in a comparative form, with three codecs intercut into one example recording. Then again in each of the following; uncompressed, super-wideband (G.722.1C), wideband (G.722/G.722.1) and finally narrowband (G.711) a la PSTN.

I could have assembled all of this into a tight little library using Flash or some Javascript, but I decided that it would be better presented laid out across a number of pages in a very plain and simple form. This way you know for certain exactly what you’re hearing and seeing in each case. There’s no active logic in the background except for a little MP3 player embedded in the page.

In order to truly appreciate the difference between the various recordings you will need to be making use of high-quality audio playback hardware. Good quality computer speakers or, better yet, a high-quality headset will be the most revealing. But then, as someone who’s genuinely concerned about the quality of audio over IP telephony…you knew that, right? I thought so.

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In part 1 I gave you an introduction to Polycom’s Siren7 & 14 codecs, as well as a brief overview of their implementation in Asterisk v1.6. Now  it makes some sense to try and understand their advantages in use. This is really a more generalized exploration of narrowband (G.711 ala PSTN) vs wideband (G.722/G.722.1) vs Super-Wideband (G.722.1C)

I set about creating a series of audio recordings to illustrate the difference between the three codecs. If Asterisk had been capable of handling all three codecs then recording samples encoded in each fashion would have been relatively simple. The trouble is that in the period leading up to Astricon I didn’t yet have a version of Asterisk capable of handling Siren streams beyond pass through.

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Preface: This post is a rework of the HDVoice session I presented in cooperation with Polycom at Astricon 2009. The Powerpoint slides in support of that session as well as a videotape recording of the session are anticipated in a few weeks on the Astricon web site.

In considering this subject I developed more demo material than was possible to use in the 40 minute session at Astricon. This post begins a series that is a kind of superset of the Astricon session, intended to go into more depth with a larger variety of HDVoice examples.

The introduction to the session was given by Tim Yankee, Director of Product Marketing, Voice Communications at Polycom. Tim’s intro gave an overview of the state of HDVoice in the industry.  Hopefully his slide set will be included in the presentation materials to be put online at Astricon.net

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