As a part of the Media Overhaul project for Asterisk 10, changes have been made to Asterisk to increase the number of codecs it's capable of supporting, to handle codecs with custom formats, and to support audio sampling rates greater than 16kHz. This has resulted in several practical changes to Asterisk that will benefit its users.
Note that the additional codecs discussed here are available for use in Asterisk's SIP channel driver, only. Asterisk 10 does not make them available for IAX2, MGCP, SSCP, H.323, UniSTIM, etc.
Expanded Signed Linear Support
For comparison, here are some Speex samples, saved as WAV files in .mov containers, for easyease-of-playback.
CELT Pass-through Support
[silk8] type=silk samprate=8000 fec=true packetloss_percentage=10 maxbitrate=20000 dtx=false [silk12] type=silk samprate=12000 fec=true packetloss_percentage=10 maxbitrate=25000 dtx=false [silk16] type=silk samprate=16000 fec=true packetloss_percentage=10 maxbitrate=30000 dtx=false [silk24] type=silk samprate=24000 fec=true packetloss_percentage=10 maxbitrate=40000 dtx=false
In this case, we have defined 4 peers, each with a different SILK codec.
To our knowledge, there are no generally available SILK softphones (except for CSIPSimple) or hardphones - the Skype client itself doesn't count as it can only be connected to the Skype networkThe generally available SIP softphones that support SILK are, to our knowledge, CSIPSimple and nightly builds of Jitsi beginning with build 3648 (so that, and anything newer than that).
The SILK licensing, like the licensing for Polycom's Siren 7 G.722.1 and Siren 14 G.722.1C codecs, requires that we distribute the distribution of binary codec modules that can be used by Asterisk. To download the SILK codec module for Asterisk, browse to http://downloads.digium.com/pub/telephony/codec_silk/unsupported/asterisk-10.0/ and drop the untar'd .so file into /usr/lib/asterisk/modules and issue an Asterisk restart, or simply load the codec module from the Asterisk CLI