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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.

Info
titleSIP Only

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

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For comparison, here are some Speex samples, saved as WAV files in .mov containers, for easyease-of-playback.

8kHz

Multimedia
height20
width100
name8kHz-Speex.mov
width100

16kHz

Multimedia
height20
width100
name16kHz-Speex.mov
width100

32kHz

Multimedia
height20
width100
name32kHz-Speex.mov
width100

CELT Pass-through Support

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No Format
[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

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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).

Note

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