This document shows you how to
install and use Opus codec in Asterisk server:
1. Go to link,
http://downloads.digium.com/pub/telephony/codec_opus/
Here select
appropriate Asterisk version of your instalation and then server
architecture, in my case it was Asterisk 13 with 64 bit
2. Now download file
as below,
- cd /usr/src
- wget
http://downloads.digium.com/pub/telephony/codec_opus/asterisk-13.0/x86-64/codec_opus-13.0_current-x86_64.tar.gz
3. Extract the
downloaded file,
- tar -xvzf
codec_opus-13.0_current-x86_64.tar.gz
- cd
codec_opus-13.0_1.1.0-x86_64
4. Copy the
codec_opus.so file into the Asterisk module directory
- cp codec_opus.so
/usr/lib/asterisk/modules/
5. Copy the
codec_opus_config-en_US.xml file into the Asterisk external
documentation directory
- cp
codec_opus_config-en_US.xml
/var/lib/asterisk/documentation/thirdparty
6. Goto Asterisk CLI
and restart it
- asterisk -rvvvv
- core restart now
- asterisk -rvvvv
- core reload
7. Now check opus
codec running or not with below command
ubuntu*CLI>
module show like opus
Module
Description Use Count Status
Support Level
codec_opus.so
OPUS Coder/Decoder 0 Running
extended
res_format_attr_opus.so
Opus Format Attribute Module 1 Running
core
If module not
loaded, then try to load it manually from Asterisk cli,
- module load
codec_opus.so
- module load
res_format_attr_opus.so
8. Open file, nano
/etc/asterisk/sip.conf and allow opus codec in it as shown below, so
SIP soft phones can use that codec.
[general]
callcounter=yes ;
enable device states for SIP devices
rtcachefriends=yes
udpbindaddr=0.0.0.0:5060
disallow=all
allow=opus
allow=ulaw
allow=alaw
allow=gsm
After adding, do sip
reload from Asterisk cli to take changes into effect
ankit-desktop*CLI>
sip reload
Reloading SIP
== Parsing
'/etc/asterisk/sip.conf': Found
== Parsing
'/etc/asterisk/users.conf': Found
== Using SIP CoS
mark 4
== Parsing
'/etc/asterisk/sip_notify.conf': Found
ankit-desktop*CLI>
Hello ,
ReplyDeleteI am facing issue with the recorded voice using MixMonitor when the call is made using opus. our SIP Traffic and RTP are TLS and SRTP respectively. Is there any clue ?
Thanks
Abdul Rasheed