Asterisk 10 Codecs and Audio Formats

Overview¶

As a part of the 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.

SIP 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¶

Versions of Asterisk prior to 10 supported 16-bit Signed Linear sampled at 8kHz and at 16kHz (versions 1.6.0 - 1.8). New to Asterisk 10 is support for a much wider range of sampling rates. Asterisk can resample between any of these sampling rates and can read/write raw 16-bit signed linear audio files from/to disk. The complete list of supported sampling rates and file format extensions is:

Asterisk 10 removes the format_sln16 file format in favor of expanded support in the main format_sln file format for all sampling rates. So, users who notice the absence of format_sln16 from their Asterisk 10 builds should not panic.

Users can create 16-bit Signed Linear files of varying sampling rates from WAV files using the sox command-line audio utility.

• SOX example

sox input.wav -t raw -b 16 -r 32000 output.sln mv output.sln output.sln32

In this example, an input WAV file has been converted to Signed Linear at a depth of 16-bits and at a rate of 32kHz. The resulting output.sln file is then renamed output.sln32 so that it can be processed correctly by Asterisk.

32kHz Speex Support¶

Asterisk versions prior to 1.8 supported 8kHz Speex. Asterisk 1.8 supports 8 and 16kHz Speex. Asterisk 10 now supports 8, 16, and 32kHz Speex. Use of the 32kHz Speex mode is, like the other modes, controlled in the respective channel driver's configuration file, e.g. chan_sip's sip.conf.

• Speex Example

sip.conf

[mypeer] type=peer secret=mysupersecret!!! host=dynamic context=fancycalls disallow=all allow=speex

[mypeer2] type=peer secret=myothers3cr3t host=dynamic context=fancycalls disallow=all allow=speex16

[mypeer3] type=peer secret=passwordisaterriblepassword host=dynamic context=fancycalls disallow=all allow=speex32

In this example, we have created three SIP peers for 3 different devices. The first, mypeer, supports only the 8kHz sampling of Speex; the second, mypeer2, supports only the 16kHz sampling of Speex; and the third, mypeer3, supports the new 32kHz sampling of Speex.

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

CELT Pass-through Support¶

Asterisk 10 adds pass-through support for the CELT codec. CELT provides low-delay transmission of high-quality audio. Unlike many other codecs that are focused on the transmission of human speech only, CELT is suitable for the transmission of both speech and audio, e.g. music.

Because the CELT codec is being folded, along with SILK, into a future codec called OPUS, and because the CELT bitstream isn't finalized, we have chosen not to add transcoding support for CELT as this time. CELT is configured in codecs.conf with the following parameters.

CELT codecs.conf example¶

[celt32] type=celt samprate=32000

[celt44] type=celt samprate=44100

[celt48] type=celt samprate=48000

In this example, three different CELT codecs are created: one for 32kHz mode, one for 44kHz mode, and another for 48kHz mode.

These codecs cannot be dynamically changed while Asterisk is running. In order to make changes, an Asterisk restart is required.

To make sure a codec or format is setup correctly, you can execute:

core show codecs

from the Asterisk CLI

CELT sip.conf example¶

Corresponding SIP peer entries to use the CELT codec would look like:

[myceltpeer1] type=peer secret=passwordisstillaterriblepassword host=dynamic context=fancycalls disallow=all allow=celt32

[myceltpeer2] type=peer secret=momnowaitdontmakemomyourpassword host=dynamic context=fancycalls disallow=all allow=celt44

[myceltpeer3] type=peer secret=daddontmakedadyourpasswordeither host=dynamic context=fancycalls disallow=all allow=celt48

In this case, we have defined 3 peers, each with a different CELT sampling rate. Thus, you'd probably want to set at least two of them to the same CELT rate, so they could call each other.

For CELT-calling, there are not a host of options on the client side. One could try Ekiga or SFLphone as softclients to make CELT calls.

SILK Support¶

Asterisk 10 provides full support for Skype's SILK codec. SILK is an extremely flexible codec for the transmission of speech. It operates in low bitrate narrow-band modes as well as higher (but still very low, otherwise) bitrate super wide-band modes. With respect to CPU complexity, its consumption is roughly three times that of G.729a at comparable bitrates.

SILK is configured in codecs.conf with the following parameters

SILK codecs.conf example¶

[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 example, four different SILK codecs are created: one each for 8 (silk8), 12 (silk12), 16 (silk16), and 24kHz (silk24).

These codecs cannot be dynamically changed while Asterisk is running. In order to make changes, an Asterisk restart is required.

To make sure a codec or format is setup correctly, you can execute:

core show codecs

from the Asterisk CLI

SILK sip.conf example¶

Corresponding SIP peer entries to use the SILK codec would look like:

[mysilkpeer1] type=peer secret=thanksdigium host=dynamic context=fancycalls disallow=all allow=silk8

[mysilkpeer2] type=peer secret=forgivingme host=dynamic context=fancycalls disallow=all allow=silk12

[mysilkpeer3] type=peer secret=suchexcellentsoftware host=dynamic context=fancycalls disallow=all allow=silk16

[mysilkpeer4] type=peer secret=touse host=dynamic context=fancycalls disallow=all allow=silk24

In this case, we have defined 4 peers, each with a different SILK codec.

The 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 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/ 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