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$ sudo service asterisk restart
Verify that Asterisk is now running not just an HTTP server, but also HTTPS:
*CLI> http show status
HTTP Server Status:
Server Enabled and Bound to 0.0.0.0:8088
HTTPS Server Enabled and Bound to 0.0.0.0:8089
Enabled URI's:
/ws => Asterisk HTTP WebSocket
You’re looking in the output for HTTPS to verify that the certificates are working, and
you also want to see /ws as that indicates the WebSockets components have loaded.
Hint: if it’s not working, always check /var/log/messages for any
SELinux messages.
$ sudo grep sealert /var/log/messages
The firewall isn’t currently configured for those ports, so we’ll need to add a few rules
to handle that:
$ sudo firewall-cmd --zone=public --add-port=8088/tcp
$ sudo firewall-cmd --zone=public --add-port=8088/tcp --permanent
$ sudo firewall-cmd --zone=public --add-port=8089/tcp
$ sudo firewall-cmd --zone=public --add-port=8089/tcp --permanent
$ sudo firewall-cmd --zone=public --add-port=5061/udp
$ sudo firewall-cmd --zone=public --add-port=5061/udp --permanent
At this point you need to fire up your web browser and make a connection. Your
browser will complain about the connection if you are using a self-signed certificate,
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Chapter 20: WebRTC
but it will allow you to make the connection. This is a critical step, as you need to tell
your browser to store the certificate permanently, so that WebRTC can use the Web‐
Socket connection. The following URL will connect you:
https://ip-of-asterisk-server:8089/ws
If you get to an Upgrade Required message, that’s a good thing. It means that the con‐
nection is good, and that’s just the protocol telling you there’s not enough technology
being served up for this to be an actual WebSocket connection. We’re where we need
to be.
Of course the next thing is to actually experience a WebRTC session through this
environment we’ve configured, and in order to test all this out, we’re going to need to
fire up our browser and load some sort of WebRTC client into it. The next section
will do just that.
Cyber Mega Phone
In order to see WebRTC in action on your Asterisk system, you’ll need something
running on your browser. The easiest way to see this in action is to take Digium’s
Cyber Mega Phone for a spin. This will allow you to quickly set up a working
WebRTC session using Asterisk.
First up, since WebRTC requires the use of TLS (it’s not optional, as it is with SIP),
we’re going to nag you one more time to verify that your certificates are installed. If
you haven’t yet done so, now is the time to work through Chapter 4, or there is also a
script provided as part of the Asterisk source code that will generate the keys and cer‐
tificates (you’ll find it in the Asterisk source code under the /home/astmin/src/
asterisk-16.<TAB>/contrib/scripts/ folder. The script is named ast_tls_cert, and it is
documented on the Asterisk wiki.
OK, now we need a tiny bit of dialplan for our WebRTC calls to arrive at:
$ vim /etc/asterisk/extensions.conf
exten => 246,1,Noop()
same => n,Answer()
same => n,Wait(0.5)
same => n,StreamEcho(4)
same => n,Hangup()
The Cyber Mega Phone itself is found at GitHub, under the Asterisk account.
You can download the code and run it from your local PC, or you can load it into a
web server and serve it from there.
Let’s serve it up from our Asterisk server:
$ cd /var/lib/asterisk/static-http
$ sudo git clone https://github.com/asterisk/cyber_mega_phone_2k.git
Cyber Mega Phone
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$ sudo chown -R asterisk:asterisk cyber_mega_phone_2k ; sudo chmod 755 cyber_mega_phone_2k