webrtc implementation asterisk : Webphone
webrtc implementation on asterisk with Webphone
What is WebRTC
WebRTC is a free, open project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs.
The WebRTC components have been optimized to best serve this purpose.The mission of WebRTC is to enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.
Asterisk and WebRTC
WebRTC/rtcweb is a defined API by JavaScript developers that allows them to venture into the world of real time communications.
This may be a click-to-call system or a “softphone” with both delivered as a webpage. No plug-ins are required and as this is a defined specification it can be used across different browsers where supported.
Asterisk has had support for WebRTC since version 11 but more stable in asterisk 13. A res_http_websocket module has been created which allows the JavaScript developers to interact and communicate with Asterisk. Support for WebSocket as a transport has been added to chan_sip to allow SIP to be used as the signaling protocol. ICE, STUN, and TURN support has been added to res_rtp_asterisk to allow clients behind NAT to better communicate with Asterisk. SRTP support was added in a previous version but it is also a requirement of WebRTC.
SRTP – What is SRTP and Why this is required
Secure media is a requirement of WebRTC and as a result SRTP must be available. In order for Asterisk to build SRTP support the libsrtp library and development headers must be available. This can be installed using the distribution’s package management system or from source. Failure to do this will result in the media offers being rejected.
pjproject – What is pjproject and Why this is required
Asterisk 11 comes with an embedded pjproject. When building Asterisk 11, to get ICE support you’ll need the UUID development library (uuid-dev for Debian, libuuid-devel for CentOS) library. If you don’t have ICE support, then you’ll likely run into audio issues in several scenarios, specifically when attempting to traverse NAT, as WebRTC uses ICE,STUN,TURN to do this.
Starting with Asterisk 12 you need to have pjproject libraries installed, otherwise you most likely won’t have audio in your WebRTC calls and no warning whatsoever!
Additionally on CentOS you may need to do “export LD_LIBRARY_PATH=/usr/lib”. The alternative is passing the correct flags (usually –with-libdir) to all dependencies to install the libraries in /usr/lib64 instead of /usr/lib. You could possibily also play with the /etc/ld.so.conf.d configuration to achieve the same effect.
Procedure to implement WebRTC with Asterisk
Step # 1
First install some of the dependencies of the Asterisk and WebRTC:
Installing dependencies
# yum update -y
# yum groupinstall “Development tools” -y
# yum install wget gcc gcc-c++ ncurses-devel libxml2-devel sqlite-devel libuuid-devel openssl-devel
# yum install gcc-c++ make gnutls-devel kernel-devel subversion doxygen texinfo curl-devel net-snmp-devel neon-devel
# yum install uuid-devel libuuid-devel sqlite-devel sqlite git speex-devel gsm-devel
Install Secure RTP library from source:
# cd /usr/src/ # sudo wget http://srtp.sourceforge.net/historical/srtp-1.4.2.tgz # sudo tar zxvf srtp-1.4.2.tgz # cd srtp* # sudo autoconf # sudo make && make install # cp /usr/local/lib/libsrtp.a /lib
If you have any issue regarding this installation, Please Join the discussion regarding this on https://discuss.eduguru.in
Install jansson library from source:
# cd /usr/src # sudo wget http://www.digip.org/jansson/releases/jansson-2.5.tar.gz # sudo tar zxvf jansson-2.5.tar.gz # cd jansson-2.5 # sudo ./configure --prefix=/ # sudo make && make install
If you have any issue regarding this installation, Please Join the discussion regarding this on https://discuss.eduguru.in
Install latest PjSIP libraries from source:
# cd /usr/src # sudo git clone https://github.com/asterisk/pjproject # cd pjproject # sudo ./configure CFLAGS="-DNDEBUG -DPJ_HAS_IPV6=1" --prefix=/usr --libdir=/usr/lib64 --enable-shared --disable-video --disable-sound --disable-opencore-amr # sudo make dep # sudo make # sudo make install # sudo ldconfig
verify pjsip library
# ldconfig -p | grep pj
If you have any issue regarding this installation, Please Join the discussion regarding this on https://discuss.eduguru.in
Now we will install Asterisk 13 from source:
# sudo wget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-13-current.tar.gz # sudo tar -zxvf asterisk-13-current.tar.gz # cd asterisk-13* # sudo make clean # sudo ./configure --with-crypto --with-ssl --with-srtp=/usr/local/lib --with-jansson=/ # sudo contrib/scripts/get_mp3_source.sh Make sure that all required modules and resources i.e. res_pjsip are enabled. Also if any Error in pjsip installation try to configure asterisk with pjsip from the below command ---------- #./configure --with-crypto --with-ssl --with-srtp=/usr/local/lib --with-jansson=/--with-pjproject-bundled ------------ # sudo make menuselect # sudo make && make install # sudo make samples # sudo make config # sudo chkconfig asterisk on
Make sure that all required modules and resources i.e. res_pjsip are enabled while make menuselect.
# make menuselect
# make && make install
# make samples
# make config
# chkconfig asterisk on
If you have any issue regarding this installation, Please Join the discussion regarding this on https://discuss.eduguru.in
Step # 3
Now we will generate TLS certificates for secure socket connections in asterisk.
# mkdir /etc/asterisk/keys
# cd /usr/src/asterisk-13*
# cd contrib/scripts
# ./ast_tls_cert -C X.X.X.X -O “My Test Company” -d /etc/asterisk/keys
Add IP Address of Server at X.X.X.X
- ast_tls_cert is a shell script located at contrib/script.
- -C is for host name or IP address of the server.
- -O is for the name of company. You can Enter Test or anything else
- -d is for destination where TLS certificate will be placed.
Point to be noted
Please check if host name of server is set correctly by hostname -f . If this showing as unknow host. Then set the proper host name like test.com, call.test.com etc
If you have any issue regarding this installation, Please Join the discussion regarding this on https://discuss.eduguru.in
Step # 4
Now we have to configure Asterisk to run with WebRTC support:
Edit rtp.conf file with following code:
;rtp.conf
[general]
rtpstart=10000
rtpend=20000
icesupport=yes
stunaddr=stun.l.google.com:19302
;res_stun_monitor.conf
[general]
stunaddr=stun.l.google.com:19302
stunrefresh = 30
Edit http.conf file with following code:
;http.conf
[general]
enabled=yes
bindaddr=0.0.0.0
bindport=8088
tlsenable=yes
tlsbindaddr=0.0.0.0:7443
tlscertfile=/etc/asterisk/keys/asterisk.pem
tlsprivatekey=/etc/asterisk/keys/asterisk.pem
Edit sip.conf file with following code:
;sip.conf
[general]
udpbindaddr = 0.0.0.0:5060
realm = X.X.X.X ; replace with your Server’s Public IP Address
transport = udp,ws,wss
externaddr = X.X.X.X ; replace with your Server’s Public IP Address
websocket_enabled = true
;SIP User Creation by adding sip account at the end of sip.conf file
[1000]
host=dynamic
secret=12345
context=nicedial
type=friend
encryption=yes
avpf=yes
;force_avp=yes
icesupport=yes
directmedia=no
disallow=all
dial = SIP/1000
disallow=all
allow=ulaw
allow=alaw
allow=speex
allow=gsm
dtlsenable=yes
dtlsverify=fingerprint
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass
nat=force_rport,comedia
[2000]
host=dynamic
secret=12345
context=nicedial
type=friend
encryption=yes
avpf=yes
;force_avp=yes
icesupport=yes
directmedia=no
disallow=all
dial = SIP/2000
disallow=all
allow=ulaw
allow=alaw
allow=speex
allow=gsm
dtlsenable=yes
dtlsverify=fingerprint
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass
nat=force_rport,comedia
Edit extensions.conf file with following code:
;extensions.conf
[nicedial]
; For Testing Audio
exten => 1111,1,Answer()
same => n,Playback(demo-thanks)
same => n,Hangup()
; For testing SIP to SIP calling
exten => _X.,1,Dial(SIP/${EXTEN})
exten => _X.,n,Hangup()
Step # 5
So far, we have added all important configurations in Asterisk.
Now we are going to start Asterisk:
# service asterisk start
Check the Port being used by Asterisk
#netstat -pln | grep asterisk
tcp 0 0 0.0.0.0:2000 0.0.0.0:* LISTEN 21012/asterisk
tcp 0 0 0.0.0.0:7443 0.0.0.0:* LISTEN 21012/asterisk
tcp 0 0 0.0.0.0:8088 0.0.0.0:* LISTEN 21012/asterisk
udp 0 0 0.0.0.0:4569 0.0.0.0:* 21012/asterisk
udp 0 0 0.0.0.0:5000 0.0.0.0:* 21012/asterisk
udp 0 0 0.0.0.0:2727 0.0.0.0:* 21012/asterisk
udp 0 0 0.0.0.0:4520 0.0.0.0:* 21012/asterisk
udp 0 0 0.0.0.0:57769 0.0.0.0:* 21012/asterisk
udp 0 0 0.0.0.0:5060 0.0.0.0:* 21012/asterisk
# asterisk -vvvvr
Asterisk 13.10.0, Copyright (C) 1999 – 2014, Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.
=========================================================================
Connected to Asterisk 13.10.0 currently running on Asterisk-13 (pid = 21012)
Asterisk-13*CLI> http show status
HTTP Server Status:
Prefix:
Server: Asterisk/13.10.0
Server Enabled and Bound to 0.0.0.0:8088
HTTPS Server Enabled and Bound to 0.0.0.0:7443
Enabled URI’s:
/httpstatus => Asterisk HTTP General Status
/phoneprov/… => Asterisk HTTP Phone Provisioning Tool
/static/… => Asterisk HTTP Static Delivery
/ari/… => Asterisk RESTful API
/ws => Asterisk HTTP WebSocket
Enabled Redirects:
None.
Asterisk-13*CLI> sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
1000 (Unspecified) D Yes Yes 0 Unmonitored
2000 (Unspecified) D Yes Yes 0 Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline]
It shows that Asterisk is running smoothly.
Step # 6
Now we are going to configure Web SIP Phone for calling:
First of all open below link in your browser and add exception for your Server’s IP (replace X.X.X.X with your Server’s IP) in your browser:
https://X.X.X.X:7443/httpstatus
After successfully adding exception for your Server’s IP Address, you’ll see below page:
We are using sipml5 based web phone for testing and You can download this from the below link
https://blog.eduguru.in/download/webrtc-sip.zip
or Configure on any webrtc client.
