Asterisk Webrtc Setup (source : github/(jwoertink))

Setup Network config

1. vi /etc/sysconfig/network-scripts/ifcfg-eth0

DEVICE=eth0
BOOTPROTO=static
DHCP_HOSTNAME=pbx.dev
HOSTNAME="pbx.dev"
IPV6INIT=yes
MTU=1500
NM_CONTROLLED=yes
ONBOOT=yes
TYPE=Ethernet
HWADDR=00:00:00:00:00 #YOUR MAC ADDRESS of VirtualBox
DNS1=8.8.8.8
USERCTL=no
IPADDR=0.0.0.0 #IP YOU NEED i.e. 172.16.1.56
NETMASK=255.255.255.0
GATEWAY=0.0.0.0 #YOUR GATEWAY i.e. 172.16.1.1

1. service network restart If any fail, you may need to run reboot
2. ifconfig | grep "inet addr" Check for IP address
3. service iptables save
4. service iptables stop
5. chkconfig iptables off

Update Server Hostname

1. vi /etc/sysconfig/network
2. HOSTNAME=pbx.dev Set the HOSTNAME to pbx.dev or whatever you want your hostname to be
3. vi /etc/hosts

172.16.1.56 pbx.dev #use your IP address
127.0.0.1 localhost pbx.dev
::1 localhost pbx.dev

1. hostname pbx.dev
2. service network restart

Install Virtualbox Guest Additions

1. Devices > Insert Guest Additions CD Image
2. mkdir /media/VirtualBoxGuestAdditions
3. mount -r /dev/cdrom /media/VirtualBoxGuestAdditions
4. yum update -y
5. yum groupinstall -y "Development Tools"
6. rpm -Uvh http://dl.fedoraproject.org/pub/epel/6/x86_64/epel-release-6-8.noarch.rpm
7. yum install -y gcc kernel-devel kernel-headers dkms make bzip2 perl
8. yum install -y kernel-devel-$(uname -r) kernel-headers-$(uname -r)
9. KERN_DIR=/usr/src/kernels/$(uname -r)/
10. export KERN_DIR
11. cd /media/VirtualBoxGuestAdditions
12. ./VBoxLinuxAdditions.run
13. reboot

Configure Terminal

1. vi /etc/grub.conf
2. Find the kernel line and add vga=791 to the end of the line
3. vi /etc/bashrc
4. update PS1 to PS1='\[\033[02;32m\]\u@\h\[\033[02;34m\]\w\$\[\033[00m\] '
5. add alias currip="ifconfig eth0 | grep 'inet addr:' | cut -d: -f2 | awk '{print \$1}'"
6. save and exit
7. sed -i s/SELINUX=enforcing/SELINUX=disabled/g /etc/selinux/config
8. reboot

Install Asterisk Current

1. yum install -y wget gcc-c++ ncurses-devel libxml2-devel sqlite-devel libsrtp-devel libuuid-devel openssl-devel iksemel-devel jansson-devel
2. cd /usr/local/src/
3. wget downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linux-complete-current.tar.gz
4. wget downloads.asterisk.org/pub/telephony/dahdi-tools/dahdi-tools-current.tar.gz
5. wget http://www.pjsip.org/release/2.2.1/pjproject-2.2.1.tar.bz2
6. wget downloads.asterisk.org/pub/telephony/libpri/libpri-1.4-current.tar.gz
7. wget downloads.asterisk.org/pub/telephony/asterisk/asterisk-12-current.tar.gz
8. tar zxvf dahdi-linux*
9. cd dahdi-linux*
10. make && make install && make config
11. cd ..
12. tar zxvf dahdi-tools*
13. cd dahdi-tools*
14. make && make install && make config
15. cd ..
16. tar zxvf libpri*
17. cd libpri*
18. make && make install
19. cd ..
20. tar -xjvf pjproject-2.2.1.tar.bz2
21. cd pjproject*
22. ./configure --prefix=/usr/lib64/ --enable-shared
23. make dep
24. make && make install
25. pjproject installs files to /usr/lib64/lib cd /usr/lib64/lib mv lib* .. mv pkgconfig/libpjproject.pc ../pkgconfig/
26. cd /usr/local/src/pjproject*
27. ldconfig
28. ldconfig -p | grep pj should return a huge list of symlinked .so files
29. PKG_CONFIG_PATH=/usr/lib64/pkgconfig/
30. export PKG_CONFIG_PATH
31. cd ..
32. tar zxvf asterisk*
33. cd asterisk*
34. ./configure --libdir=/usr/lib64
35. make menuselect
36. Select Resource Modules then scroll down to ensure a * is next to res_srtp. Press x to save & quit
37. vi include/asterisk/autoconfig.h this is a super hack >_<
38. replace #undef HAVE_PJ_TRANSACTION_GRP_LOCK with #define HAVE_PJ_TRANSACTION_GRP_LOCK 1
39. make && make install
40. make samples
41. make config

Configure Asterisk for WebRTC

1. mkdir /etc/asterisk/keys
2. cd /usr/local/src/asterisk*/contrib/scripts
3. ./ast_tls_cert -C $(currip) -O "My Super Company" -d /etc/asterisk/keys
4. vi /etc/asterisk/http.conf

[general]
enabled=yes
bindaddr=127.0.0.1 ; Replace this with your IP address
bindport=8088 ; Replace this with the port you want to listen on

1. vi /etc/asterisk/sip.conf

[general]
context=default
allowguest=no
allowoverlap=no
accept_outofcall_message=yes
outofcall_message_context=default
realm=127.0.0.1 ; Replace this with your IP address
udpbindaddr=127.0.0.1 ; Replace this with your IP address
transport=ws,wss,udp
language=en
icesupport=yes
videosupport=yes
nat=auto_force_rport,auto_comedia
allow=!all,alaw,ulaw,gsm
;
[1060] ; This will be WebRTC client
type=friend
username=1060 ; The Auth user for SIP.js
host=dynamic ; Allows any host to register
secret=password ; The SIP Password for SIP.js
encryption=yes ; Tell Asterisk to use encryption for this peer
avpf=yes ; Tell Asterisk to use AVPF for this peer
icesupport=yes ; Tell Asterisk to use ICE for this peer
context=default ; Tell Asterisk which context to use when this peer is dialing
directmedia=no ; Asterisk will relay media for this peer
transport=ws,wss,udp ; Asterisk will allow this peer to register on UDP or WebSockets
force_avp=yes ; Force Asterisk to use avp. Introduced in Asterisk 11.11
dtlsenable=yes ; Tell Asterisk to enable DTLS for this peer
dtlsverify=no ; Tell Asterisk to not verify your DTLS certs
dtlscertfile=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS cert file is
dtlsprivatekey=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS private key is
dtlssetup=actpass ; Tell Asterisk to use actpass SDP parameter when setting up DTLS
videosupport=yes
nat=no
disallow=all
allow=ulaw,vp8,h264
;
[1061] ; This will be the legacy SIP client
type=friend
username=1061
host=dynamic
secret=password
context=default
directmedia=no
transport=udp
force_avp=yes
dtlsenable=no
videosupport=yes
nat=no
disallow=all
allow=ulaw,vp8

1. vi /etc/asterisk/extensions.conf

[default]
exten => 1060,1,Dial(SIP/1060) ; Dialing 1060 will call the SIP client registered to 1060
exten => 1061,1,Dial(SIP/1061) ; Dialing 1061 will call the SIP client registered to 1061

1. vi /etc/asterisk/manager.conf

[general]
enabled=yes
port=5038
bindaddr=0.0.0.0
;
[admin]
secret=password
read=all
write=all
writetimeout=5000

1. service asterisk restart
2. Ensure Linphone is installed, and all Video and Audio codecs are turned on
3. SIP account should be sip:1061@CENT_OS_IP_ADDRESS with udp transport

Install & Configure ejabberd

1. yum install -y ejabberd
2. vi /etc/ejabberd/ejabberd.cfg

% Find this line and add the pbx.dev
{hosts, ["localhost", "pbx.dev"]}.

1. service ejabberd start
2. ejabberdctl register admin pbx.dev password to create an account called admin on pbx.dev with the password "password"
3. vi /etc/ejabberd/ejabberd.cfg

% This goes under the ACCESS CONTROL section
{acl, admin, {user, "admin", "pbx.dev"}}.

1. service ejabberd restart
2. Browse to http://pbx.dev:5280/admin and use admin/password to log in
3. Virtual Hosts > pbx.dev > Users - Make a user called "asterisk". Also make yourself a user
4. vi /etc/asterisk/xmpp.conf

[general]
autoregister=yes
autoprune=no
;
[ejabberd]
type=client
serverhost=pbx.dev
username=asterisk@pbx.dev
secret=password ;your password you chose for asterisk
priority=1
port=5222
usetls=no
usesasl=yes
status=available
statusmessage="It's Asterisk!"
timeout=5

1. asterisk -r
2. module reload res_xmpp - should show reloading
3. xmpp show connections - should show 1 client connected. If so, then exit
4. vi /etc/asterisk/extensions.conf

; replace the exten 1061 dialplan with this
exten => 1061,1,Set(JSTATUS=${JABBER_STATUS(ejabberd,youruser@pbx.dev/Desktop)}) ; /Desktop is the resource you will set in your xmpp client like Adium/Pidgin
same => n,GotoIf($[0${JSTATUS}=1]?available:unavailable)
same => n(available), JabberSend(ejabberd,youruser@pbx.dev,"Incoming call from ${CALLERID(num)}")
same => n,Dial(SIP/1061)
same => n,Hangup()
same => n(unavailable),JabberSend(ejabberd,youruser@pbx.dev,"Missed call from ${CALLERID(num)}")
; do other dialplan stuff when you're not available

1. service asterisk restart