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We’ll need a small change to the configuration of Asterisk’s HTTP server to allow it to
serve static content.
$ sudo vim /etc/asterisk/http.conf
[general]
enabled=yes
bindaddr=0.0.0.0
bindport=8088
tlsenable=yes
tlsbindaddr=0.0.0.0:8089
tlscertfile=/home/asterisk/certs/asterisk.crt
tlsprivatekey=/home/asterisk/certs/asterisk.key
enablestatic=yes
redirect=/cmp2k /static/cyber_mega_phone_2k/index.html
Save and reload the http module from your Asterisk console:
*CLI> module reload http
Now, using your browser, you can navigate over to your new WebRTC client app:
https://your asterisk server:8089/cmp2k
If all went as planned, you should see something like Figure 20-1.
Figure 20-1. Cyber Mega Phone 2K
Press the Account button, and input the credentials for your WebRTC user (see
Figure 20-2).
Figure 20-2. WebRTC account credentials
Once you’ve input the details relevant to your system, press X to save and close.
Now, you can press the Connect button, and if all went well your WebRTC client
should register to Asterisk (this would be a good time to monitor the Asterisk console
to see what’s happening and whether there are any errors).
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Chapter 20: WebRTC
If you press the Call button now, you should end up connected via WebRTC, and
you’ll see two windows (Figure 20-3). One of them is your local video, and the other
is reflected back from the far end (i.e., it is simulating another user by echoing back
what you sent). If all your audio is working too, you might even get some feedback
noise!
Figure 20-3. Echo application with video
You see that there is a Remote Video window alongside the Local Video window. We
haven’t achieved much to brag about, perhaps, but your Asterisk system is handling
WebRTC, so smile and take a break. You’ve earned it.
More About WebRTC
The WebRTC ecosystem is rapidly evolving, and what is true as of this writing may
not be true in the near future. We have found the following resources to be very
helpful:
• Tsahi Levant-Levi is involved in many different WebRTC initiatives, and he gen‐
erously shares knowledge relating to how to learn WebRTC. Check out his blog‐
geek.me website. Follow him.
• A group of folks under the Kranky Geek handle produced some WebRTC confer‐
ences, and shared many useful videos on YouTube. The YouTube channel Kranky
Geek is where you’ll find them.
• Get familiar with the various signaling protocols that are popular with WebRTC:
SIP, VIRTO (from the FreeSwitch project), XMPP, and even JSON.
• Look up various WebRTC signaling libraries. Currently, the popular ones
include: sipML5 (arguably the very first WebRTC library) and JsSIP (plus a fork
of JsSIP named SIP.js).
More About WebRTC
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• webrtc.org is the official home of WebRTC, and certainly deserves some of your
time. Check out the Getting Started page.
• O’Reilly’s online learning platform has a few videos that are worth a watch. For
any books and videos, keep an eye on the publish date, since anything older than
a year or two is likely to be out of date—WebRTC is still under rapid develop‐
ment.
There is so much more to learn, but we’re out of pages here.
Conclusion
WebRTC is exciting and important, and it’s very likely that VoIP developers and inte‐
grators are going to need to be familiar with this technology if they are to keep their
skill sets relevant. As of this writing, WebRTC is still very much a work in progress.
As with any exploration of new frontiers, those who blaze a trail must be creative,