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.
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.
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.
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.
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
This coming week I'll be doing a short presentation at Astricon 2009 in Phoenix. For some reason the description of the session has never made it onto the event web site so I thought I'd describe it here in case…
Michael Stanford of Wirevolution has an article called Better Sounding Calls in the March issue of Internet Telephony that was today published on TMCs HDVoice Community site. While very general it's nevertheless a nice article. He cites Speex developer Jean-Marc…