Thai Hoa Group

First let's define a few concepts to start, as a starting point we WebRTC which is a protocol originally created by Google which using standards like HTML5 and JavaScript can provide the web capabilities of VoIP communication among many other things , Moreover SIP is a signaling protocol that enables communication between users using a server and / or SIP client.
So that both communication systems while it is true are different, they have the same goal which is "communication" because it has built software that can integrate both, so that can be programmed pages that use WebRTC and communicate with SIP servers or SIP platforms supporting such as Asterisk

In the case of Elastix 4 features support for WebRTC because it uses Asterisk version 11, which implements the res_http_websocket module that has been created by Digium to allow developers to interact and communicate with WebRTC, also in this version have been added protocols signaling as ICE, STUN, TURN, SRTP because they are requirement WebRTC

While setting up a WebPhone would be just one of many applications that we could build, we will set up a basic using SIPml5 which is a SIP based client written in HTML5 and javascript recently released.

Considerations before starting
Before starting the configuration you need to run the update of the platform since upgrading can enable support for the transport protocol WSS necessary for WebRTC..

yum -y update

1.- Set the name of the machine

Set the name of the machine as "elx4" with the domain "mycompany.com" as follows..

vim /etc/hosts

Enter the end of file

192.168.10.48 elx4.mycompany.com elx4

Where is the IP address of the virtual machine and "elx4" is the name of the machine installed then test the configuration

hostname -f

The result MUST be: elx4.mycompany.com

2.- Create certificates for SRTP

Create Folder certificates

mkdir /etc/asterisk/keys

Create server certificates. During the process you prompted to enter a password..

/usr/share/doc/asterisk-11.21.0/contrib/scripts/ast_tls_cert -C pbx.mycompany.com -O “My Super Company” -d /etc/asterisk/keys

Create client certificates, you need to enter the same key used in the previous step.

/usr/share/doc/asterisk-11.21.0/contrib/scripts/ast_tls_cert -m client -c /etc/asterisk/keys/ca.crt -k /etc/asterisk/keys/ca.key -C phone1.mycompany.com -O “My Super Company” -d /etc/asterisk/keys -o malcolm

Configure permissions

chown -R asterisk.asterisk /etc/asterisk/keys

Configure SRTP support level Asterisk

vim /etc/asterisk/sip_general_custom.conf

Add at the end of the file.

tlsenable=yes
tlsbindaddr=0.0.0.0
tlscertfile=/etc/asterisk/keys/asterisk.pem
tlscafile=/etc/asterisk/keys/ca.crt
tlscipher=ALL
tlsclientmethod=tlsv1

3.- Configure Asterisk embedded HTTP server

Create certificates

openssl req -new -x509 -days 365 -nodes -out /etc/asterisk/keys/foo.pem -keyout /etc/asterisk/keys/foo.pem

During the process requests the following sample data.

Country Name (2 letter code) [XX]:PE
State or Province Name (full name) []:LIMA
Locality Name (eg, city) [Default City]:LIMA
Organization Name (eg, company) [Default Company Ltd]:SILCOM
Organizational Unit Name (eg, section) []:TI
Common Name (eg, your name or your server’s hostname) []:elx4
Email Address []:

Then to configure the service is necessary to enter from the web interface Security / Advanced Settings and enable the use of Freepbx as follows:

Then enter: https://192.168.10.48/admin/ and enter the "Settings / Advanced Settings" and change the settings in the "Asterisk Builtin mini-HTTP server" section as shown below:

Parameter	Value
Certificate file	/etc/asterisk/keys/foo.pem
Enabled HTTPS support for the mini-HTTP Server	True
HTTP Blind Address	0.0.0.0
HTTPS Bind Address/Port	0.0.0.0:8089
Private key file	/etc/asterisk/keys/foo.pem

The "http_additional.conf" file must reflect the following settings:

vim /etc/asterisk/http_additional.conf

[general]
enabled=yes
enablestatic=no
bindaddr=0.0.0.0
bindport=8088
prefix=asterisk
tlsenable=yes
tlsbindaddr=0.0.0.0:8089
tlscertfile=/etc/asterisk/keys/foo.pem
tlsprivatekey=/etc/asterisk/keys/foo.pem

4.- Create extensions with support for WebRTC

Enter PBX -> PBX Configuration -> Extensions then create 02 extensions as follows:

Go to the bottom of the form and click the "submit" button then edit the extension created and set the following parameters.

Parameter	Value
Transport	All – WS Primary
avpf	Yes
icesupport	Yes
dtlsenable	Yes
dtlsverify	No
dtlssetup	Incoming and Outgoing
dtlscertfile	/etc/asterisk/keys/foo.pem
dtlsprivatekey	/etc/asterisk/keys/foo.pem
encryption	Yes (SRTP only)

5.- Install Webphone for WebRTC

Download from the Web Phone:

https://github.com/DoubangoTelecom/sipml5

Then upload it to the / var / www / html / and set as follows:

cd /var/www/html
yum -y install unzip
unzip sipml5-master.zip
mv sipml5-master phone

6.- Configure the Webphone for WebRTC

As a first step it is necessary to enter:

https://192.168.10.48:8089/asterisk/ws

Then accept the certificate as follows:

Then enter the customer WebRTC: http://192.168.10.48/phone/call.htm and show us the WebPhone
Next you need to enter the button "Expert Mode" as shown below:

Then set the preferences as shown below:

The WebSocket server must be: wss: //192.168.10.48: 8889 / asterisk / ws
Now registration set as follows Wephone:

We will see that the WebPhone be recorded and from the Asterisk CLI will see the record as follows:

6.- Test calls from the Web Phone

We will try to validate call to voicemail (* 98), when dialing voicemail browser request permission to activate the micro and establish the call, as shown below.

Finally create a "normal" extension / annex and test calls to and from the WebPhone as shown below..

7.- Reference
Probando WebRTC en Elastix 4 (paso a paso)

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