To add a codec for SIP/SDP (m=, rtmap, and ftmp), you create a format module in Asterisk: codec_silk.patch (for m= and rtmap) and res/res_format_attr_silk.c (for fmtp). However, this requires both call legs to support SILK (pass-through only). If one leg does not support SILK, the call has no audio. Or, if you use the pre-recorded voice and music files of Asterisk, these files cannot be heard, because they are not in SILK but in signed linear (SLN). Therefore, this repository adds not just a format module but a transcoding module as well: codecs/codec_silk.c.
The ZIP file for SILK contains the latest RTP payload-format description in the folder doc. This repository here is an wrapper implementation of that document. SILK is deprecated since September 2012 because its work continues within the Opus Codec in the mode Linear Prediction (LP). Nevertheless, you are here because you have a VoIP client which does not support Opus yet but still supports SILK. Or you want to learn more about SILK because it is one foundation of Opus. Research papers comparing SILK with other audio codecs: InterSpeech 2010 and 2011. Further examples…
Since Asterisk 13.12, SILK is not only supported for pass-through but can be transcoded as well. This allows you to translate to/from other audio codecs like those for landline telephones (ISDN: G.711; DECT: G.726-32; and HD: G.722) or mobile phones (GSM, AMR, AMR-WB, 3GPP EVS). This can be achieved by
A. enabling codec_silk via make menuselect, or
B. downloading the module from Digium Downloads » Add-on Voice Codecs, or
C. downloading the module directly.
That way, you get a binary module, which is closed software. Here, this repository offers Open Source Software. In contrast to a binary module, Open Source Software allows you to double-check the existing code, contribute and/or add your own features.
This repository is for Asterisk 13 and newer. If you still use Asterisk 11, please, continue there… however, Asterisk 11 does not offer negotiation of SDP parameters (fmtp). If you need that, please, upgrade to Asterisk 13.7 or newer.
At least Asterisk 13.7 is required. These changes were last tested with Asterisk 13.18 (and Asterisk 15.1). If you use a newer version and transcoding fails, please, report!
cd /usr/src/
wget downloads.asterisk.org/pub/telephony/asterisk/asterisk-13-current.tar.gz
tar zxf ./asterisk*
sudo apt-get --assume-yes install build-essential autoconf libssl-dev libncurses-dev libnewt-dev libxml2-dev libsqlite3-dev uuid-dev libjansson-dev libblocksruntime-dev
Install libraries:
If you do not want transcoding but pass-through only (because of license issues) please, skip this step. To support transcoding, you’ll need to compile SILK, for example the floating point variant (FLP) in Debian/Ubuntu:
cd /usr/src/
wget web.archive.org/web/20130205164531/http://developer.skype.com/silk/SILK_SDK_SRC_v1.0.9.zip
unzip -qq SILK*
cd ./SILK_SDK_SRC_FLP*
CFLAGS='-fPIC' make lib
sudo cp ./interface/*.h /usr/include/
sudo cp ./libSKP_SILK_SDK.a /usr/lib/
Preparations (required since Asterisk 13.18; undo the video-codec VP9)
cd /usr/src/asterisk*
wget www.traud.de/voip/sip/patches/24bb5a89089caca8e16989bab7458617b91e4ef4.patch
patch -p1 --reverse <./24bb5a89089caca8e16989bab7458617b91e4ef4.patch
Preparations (required since Asterisk 13.12; undo the changes for the Digium module)
cd /usr/src/asterisk*
wget github.com/asterisk/asterisk/commit/f6821fbaec3fed7bbc1c814de3a4824cc926a90d.patch
patch -p1 --reverse <./f6821fbaec3fed7bbc1c814de3a4824cc926a90d.patch
wget github.com/asterisk/asterisk/commit/28501051b47e6bb8968bb016abf0b3493c05fa21.patch
patch -p1 --reverse <./28501051b47e6bb8968bb016abf0b3493c05fa21.patch
Preparations (required since Asterisk 13.18; redo the video-codec VP9)
patch -p1 <./24bb5a89089caca8e16989bab7458617b91e4ef4.patch
Apply all patches:
cd /usr/src/asterisk*
wget github.com/traud/asterisk-silk/archive/master.zip
unzip -qq master.zip
rm master.zip
cp --verbose --recursive ./asterisk-silk*/* ./
patch -p0 <./codec_silk.patch
patch -p0 <./build_silk.patch
Run the bootstrap script to re-generate configure:
./bootstrap.sh
Configure your patched Asterisk:
./configure
Enable SLN16 in menuselect for transcoding, for example via:
make menuselect.makeopts
./menuselect/menuselect --enable-category MENUSELECT_CORE_SOUNDS
Compile and install:
make
sudo make install
You can test SILK out of the box using:
A. (Google Android) CSipSimple
B. (Google Android) CounterPath Bria
C. (Apple iOS) CounterPath Bria
D. (Windows Phone 8) Linphone
However, these VoIP/SIP clients offer G.722 and Opus, which should be used for wide-band audio instead. G.722 transcoding is build into Asterisk already. Opus can be added as transcoding module…
codecs.conf is not used, because the client should tell the server (Asterisk) which configuration is desired. However, tests revealed that none of the above SILK implementations send the format attribute maxaveragebitrate, although each implementation is using a rather low bit-rate actually. As consequence, the client is sending with a rather moderate quality and Asterisk is responding with the highest quality possible. This wastes bandwidth. If you are not able to convince the author of that VoIP client to send its maxaveragebitrate via SIP/SDP, you have to edit the source of res/res_format_attr_silk.c and codecs/codec_silk.c. If you need not a general but a configuration for each peer, please, add or report support for codecs.conf.
Currently, the bit-rate is fixed and does not change while in a call, because there is no connection between RTCP and SILK. Therefore, it is not possible to adjust the bit-rate for congestion control. For example, when a mobile network is used, the user might travel from very fast 4G (LTE) to a fast 3.5G (HSPA), dropping down to a moderate 3G (UMTS) or even near to unusable 2G (GSM) connection. Same applies for the Opus module, FreeSWITCH, and PJSIP.
Several frames per payload (compound payload) are another issue, for example when the packetization time (ptime) is longer than 20 ms. For sending, codecs/codec_silk.c:lintosilk_frameout would need an change and main/codec_builtin.c:silk_samples while receiving.
- Skype for creating SILK and its library.
- Asterisk team: Thanks to their efforts and architecture this module was adopted in one working day.
- Todd Mortimer provided the starting point with his module for Asterisk 11.