[post on Digium Blogs] If you’ve been following the development of Asterisk carefully, you’ve seen that some changes have been made with respect to Asterisk’s codec support.

Beginning with the 1.6.2 release of Asterisk, pass-through support for Polycom’s Siren 7 (G.722.1) and Siren 14 (G.722.1C) was added. Both of these codecs offer improvements upon the more commonly known and used G.722 codec.

G.722 is a 7kHz codec, sampled at 16kHz, that, for Asterisk, operates at 64kbit/s. Siren 7 (G.722.1) offers the same frequency response (7kHz) and sampling rate (16kHz) of regular G.722, but it’s done in half the bandwidth – 32kbit/s, as opposed to G.722’s 64kbit/s. Siren 14 (G.722.1C) is a 14kHz codec, sampled at 32kHz, that, for Asterisk, operates at 48kbit/s. Siren14 thus offers twice the frequency response and sampling rate of G.722 or G.722.1, and does so at 3/4 the bandwidth of G.722 and only 50% more bandwidth than G.722.1.

We’ve recently upgraded Asterisk’s pass-through capability for these two codecs to full codec translation support. Now available on our downloads server are Asterisk 1.6.2 codec translation modules for Polycom’s Siren 7 (G.722.1) and Siren 14 (G.722.1C). Digium has elected to provide these codec modules in binary format, as opposed to source code format, so that users do not have to individually execute licensing agreements with Polycom in order to use them.

Currently we’re aware of a number of endpoints that support Siren7 and Siren14 codecs including the Polycom Soundstation IP 6000 and 7000 model conference phones, the VVX 1500 media phone as well as the latest release of the Zoiper Communicator soft client.

Why does any of this matter? Well, if you’re a PSTN junkie, it doesn’t. But, if you find that you’re making most of your calls in the VoIP world, then it’s going to make those long days on the phone a bit more tolerable. At the very least, you’ll be able to more clearly understand the hold music.