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<h1 id="InstallingandConfiguringCyberMegaPhone-Introduction" style="margin: 10px 0 0 0; margin-top: 0; font-size: 24px; font-weight: normal; line-height: 30px; margin: 40px 0 0 0; margin-top: 0">Introduction</h1>
<p style="margin: 10px 0 0 0">At <a class="confluence-link unresolved" href="#" style="color: #3b73af; text-decoration: none">AstriDevCon 2017</a>, Digium introduced a sample WebRTC Video Conference Web Application called CyberMegaPhone (CMP2K). This document will walk you through installing the application and configuring it and Asterisk as a simple video conference server.</p>
<p style="margin: 10px 0 0 0"> </p>
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<h1 id="InstallingandConfiguringCyberMegaPhone-Prerequisites" style="margin: 10px 0 0 0; margin-top: 0; font-size: 24px; font-weight: normal; line-height: 30px; margin: 40px 0 0 0; margin-top: 0">Prerequisites</h1>
<p style="margin: 10px 0 0 0">Before proceeding, follow the instructions for <a class="confluence-link unresolved" href="#" style="color: #3b73af; text-decoration: none">Configuring Asterisk for WebRTC Clients</a> and then use SIPML5 to test your connectivity by following the instructions at <a class="confluence-link unresolved" href="#" style="color: #3b73af; text-decoration: none">WebRTC tutorial using SIPML5</a>. The instructions below assume you've completed those steps.<span class="diff-html-added" id="added-diff-0" style="font-size: 100%; background-color: #ddfade;"> Don't forget, Asterisk 15.5 or better is required.</span> </p>
<p style="margin: 10px 0 0 0">You'll also need a working webcam and microphone on your client computer. CMP2K will not connect unless both are available.</p>
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<h1 id="InstallingandConfiguringCyberMegaPhone-GetTheCode" style="margin: 10px 0 0 0; margin-top: 0; font-size: 24px; font-weight: normal; line-height: 30px; margin: 40px 0 0 0; margin-top: 0">Get The Code</h1>
<p style="margin: 10px 0 0 0">The CyberMegaPhone (CMP2K) code is located in Asterisk's public Github repository at <a href="https://github.com/asterisk/cyber_mega_phone_2k.git" class="external-link" rel="nofollow" style="color: #3b73af; text-decoration: none">https://github.com/asterisk/cyber_mega_phone_2k</a>. You can either download the code as a zip file or clone the repository using git. Which ever way you choose, download it now to the directory of your choice. We'll use <code style="font-family: monospace">/usr/src/asterisk/cyber_mega_phone_2k</code> in the instructions below.</p>
<p style="margin: 10px 0 0 0">From an installation perspective, that's all there is to it. It's just configuration from now on.</p>
<p style="margin: 10px 0 0 0">As a reminder, we'll be using <code style="font-family: monospace">pbx.example.com</code> as our hostname so substitute it with your own hostname or ip address.</p>
<h1 id="InstallingandConfiguringCyberMegaPhone-ConfigureAsterisk" style="margin: 10px 0 0 0; font-size: 24px; font-weight: normal; line-height: 30px; margin: 40px 0 0 0">Configure Asterisk</h1>
<h2 id="InstallingandConfiguringCyberMegaPhone-(Re)ConfiguretheAsteriskHTTPServer" style="margin: 10px 0 0 0; font-size: 20px; font-weight: normal; line-height: 30px; margin: 40px 0 0 0; margin-top: 10px">(Re)Configure the Asterisk HTTP Server</h2>
<p style="margin: 10px 0 0 0">The CMP2K software needs to be served by a TLS capable web server. The easiest way to do this by far is to simply use Asterisk's built-in HTTP server. Here's what we need to add...</p>
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<td class="diff-macro-body" style="background-color: #fff;border: 1px solid #dddddd;padding: 10px;; padding: 0px; border-collapse: collapse"> <pre style="margin: 10px 0 0 0; margin-top: 0">; Existing definition
[general]
enabled=yes
bindaddr=0.0.0.0
bindport=8088
tlsenable=yes
tlsbindaddr=0.0.0.0:8089
tlscertfile=<your_cert_file>
tlsprivatekey=<your_key_file>
tlscafile=<your_ca_cert_file>
; Add the following if not already present
; Allow the HTTP server to serve static content from /var/lib/asterisk/static-http
enablestatic = yes
; Create an alias that will allow us to easily load the client in a web browser.
redirect = /cmp2k /static/cyber_mega_phone_2k/index.html
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<p style="margin: 10px 0 0 0">Restart Asterisk or issue the CLI command "<code style="font-family: monospace">config reload /etc/asterisk/http.conf</code>"</p>
<p style="margin: 10px 0 0 0">Now check that the configuration was applied. From the Asterisk CLI...</p>
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<td class="diff-macro-body" style="background-color: #fff;border: 1px solid #dddddd;padding: 10px;; padding: 0px; border-collapse: collapse"> <pre style="margin: 10px 0 0 0; margin-top: 0">
*CLI> http show status
HTTP Server Status:
Prefix:
Server: Asterisk/GIT-16-a84c257cd6
Server Enabled and Bound to [::]:8088
HTTPS Server Enabled and Bound to [::]:8089
Enabled URI's:
/test_media_cache/... => HTTP Media Cache Test URI
/guimohdir_rh => HTTP POST mapping
/httpstatus => Asterisk HTTP General Status
/phoneprov/... => Asterisk HTTP Phone Provisioning Tool
/amanager => HTML Manager Event Interface w/Digest authentication
/backups => HTTP POST mapping
/arawman => Raw HTTP Manager Event Interface w/Digest authentication
/manager => HTML Manager Event Interface
/rawman => Raw HTTP Manager Event Interface
/static/... => Asterisk HTTP Static Delivery
/amxml => XML Manager Event Interface w/Digest authentication
/mxml => XML Manager Event Interface
/moh => HTTP POST mapping
/ari/... => Asterisk RESTful API
/ws => Asterisk HTTP WebSocket
Enabled Redirects:
/cmp2k => /static/cyber_mega_phone_2k/index.html
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<p style="margin: 10px 0 0 0"> </p>
<p style="margin: 10px 0 0 0">Notice that there's a new Redirect entry. </p>
<p style="margin: 10px 0 0 0">For security reasons, the HTTP server will not serve arbitrary paths so the <code style="font-family: monospace">/static/cyber_mega_phone_2k/index.html</code> path will actually resolve to is <code style="font-family: monospace">/var/lib/asterisk/static-http/cyber_mega_phone_2k/index.html</code>. You can either move the CMP2K directory that you downloaded to <code style="font-family: monospace">/var/lib/asterisk/static-http</code> or you can simply create a symlink to it as follows:</p>
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<td class="diff-macro-body" style="background-color: #fff;border: 1px solid #dddddd;padding: 10px;; padding: 0px; border-collapse: collapse"> <pre style="margin: 10px 0 0 0; margin-top: 0"># cd /var/lib/asterisk/static-http
# ln -s /usr/src/asterisk/cyber_mega_phone_2k</pre> </td>
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<p style="margin: 10px 0 0 0">OK, let's test. From your web browser, visit <code style="font-family: monospace">https://pbx.example.com:8089/cmp2k</code> remembering to substitute your hostname or ip address as appropriate.</p>
<p style="margin: 10px 0 0 0">Did you get?...</p>
<p style="margin: 10px 0 0 0"> <img class="confluence-embedded-image" width="500" src="/wiki/download/attachments/40818102/Screenshot_2018-09-07_12-03-18.png?version=1&modificationDate=1536343468621&api=v2" data-image-src="/wiki/download/attachments/40818102/Screenshot_2018-09-07_12-03-18.png?version=1&modificationDate=1536343468621&api=v2" /></p>
<p style="margin: 10px 0 0 0">Great. </p>
<h2 id="InstallingandConfiguringCyberMegaPhone-(Re)ConfigurePJSIP" style="margin: 10px 0 0 0; font-size: 20px; font-weight: normal; line-height: 30px; margin: 40px 0 0 0">(Re)Configure PJSIP</h2>
<p style="margin: 10px 0 0 0">In the <a class="confluence-link unresolved" href="#" style="color: #3b73af; text-decoration: none">Configuring Asterisk for WebRTC Clients</a> tutorial, you created a PJSIP Endpoint named "webrtc_client". We need to modify that definition for our purposes.</p>
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<td style="background-color: #fafafa; padding: 0 0 0 5px; font-size: 12px; text-align: left;; padding: 0px; border-collapse: collapse">/etc/asterisk/pjsip.conf</td>
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<td class="diff-macro-body" style="background-color: #fff;border: 1px solid #dddddd;padding: 10px;; padding: 0px; border-collapse: collapse"> <pre style="margin: 10px 0 0 0; margin-top: 0">[webrtc_client]
type=endpoint
aors=webrtc_client
auth=webrtc_client
dtls_auto_generate_cert=yes
webrtc=yes
context=default
disallow=all
; We need to allow more codecs.
; vp8, vp9 and h264 are video pass-through codecs.
; No special Asterisk modules are required to support them.
allow=opus,g722,ulaw,vp9,vp8,h264
; Since video conferencing makes use of the Streams functionality added in Asterisk 15
; we need to indicate the maximum number of streams allowed for audio and video.
max_audio_streams = 1
max_video_streams = 15
</pre> </td>
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<p style="margin: 10px 0 0 0">There are two more Asterisk changes we need to make so no need to restart Asterisk just yet.</p>
<h2 id="InstallingandConfiguringCyberMegaPhone-Configureapp_confbridge" style="margin: 10px 0 0 0; font-size: 20px; font-weight: normal; line-height: 30px; margin: 40px 0 0 0">Configure app_confbridge</h2>
<p style="margin: 10px 0 0 0">The sample <code style="font-family: monospace">confbridge.conf</code> file is enough to get you going with one exception. In the <code style="font-family: monospace">default_bridge</code> section, we need to set <code style="font-family: monospace">video_mode=sfu</code>.</p>
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<td class="diff-macro-body" style="background-color: #fff;border: 1px solid #dddddd;padding: 10px;; padding: 0px; border-collapse: collapse"> <pre style="margin: 10px 0 0 0; margin-top: 0">[default_bridge]
type=bridge
; other stuff
; SFU is Selective Forwarding Mode
; Basically all participant's video streams are relayed to all other participants.
video_mode = sfu
</pre> </td>
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<p style="margin: 10px 0 0 0">One more change...</p>
<h2 id="InstallingandConfiguringCyberMegaPhone-Configureextensions.conf" style="margin: 10px 0 0 0; font-size: 20px; font-weight: normal; line-height: 30px; margin: 40px 0 0 0">Configure extensions.conf</h2>
<p style="margin: 10px 0 0 0">Now we need to configure an extension that, when dialed, will put us into the video conference bridge, so add the following to extensions.conf</p>
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<td class="diff-macro-body" style="background-color: #fff;border: 1px solid #dddddd;padding: 10px;; padding: 0px; border-collapse: collapse"> <pre style="margin: 10px 0 0 0; margin-top: 0">[default]
exten = my_video_conference,1,Confbridge(MYCONF,default_bridge,default_user,sample_user_menu)
</pre> </td>
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<p style="margin: 10px 0 0 0">NOW, restart Asterisk!</p>
<h1 id="InstallingandConfiguringCyberMegaPhone-JointheConferenceBridge!" style="margin: 10px 0 0 0; font-size: 24px; font-weight: normal; line-height: 30px; margin: 40px 0 0 0">Join the Conference Bridge!</h1>
<p style="margin: 10px 0 0 0">Open a browser window and visit <code style="font-family: monospace"><a href="https://pbx.example.com:8089/cmp2k" class="external-link" rel="nofollow" style="color: #3b73af; text-decoration: none">https://pbx.example.com:8089/cmp2k</a></code> remembering to substitute your hostname or ip address as appropriate.</p>
<p style="margin: 10px 0 0 0">You should be back at the page you got when you tested earlier...</p>
<p style="margin: 10px 0 0 0"> <img class="confluence-embedded-image" width="500" src="/wiki/download/attachments/40818102/Screenshot_2018-09-07_12-03-18.png?version=1&modificationDate=1536343468621&api=v2" data-image-src="/wiki/download/attachments/40818102/Screenshot_2018-09-07_12-03-18.png?version=1&modificationDate=1536343468621&api=v2" /></p>
<p style="margin: 10px 0 0 0">Click the <code style="font-family: monospace">Account</code> button and fill in the details as follows...</p>
<p style="margin: 10px 0 0 0"> <img class="confluence-embedded-image" src="/wiki/download/attachments/40818102/Screenshot_2018-09-07_12-30-43.png?version=1&modificationDate=1536345105356&api=v2" data-image-src="/wiki/download/attachments/40818102/Screenshot_2018-09-07_12-30-43.png?version=1&modificationDate=1536345105356&api=v2" /></p>
<p style="margin: 10px 0 0 0">Now click anywhere outside the edit box or click the <span style="font-family: monospace;">X</span> in the upper right corner to save the information and you'll be be back to the previous page.</p>
<p style="margin: 10px 0 0 0">Now click <code style="font-family: monospace">Connect</code> and you should see the button change to <code style="font-family: monospace">Disconnect</code> </p>
<p style="margin: 10px 0 0 0"> <img class="confluence-embedded-image" width="500" src="/wiki/download/attachments/40818102/Screenshot_2018-09-07_12-34-07.png?version=1&modificationDate=1536345321576&api=v2" data-image-src="/wiki/download/attachments/40818102/Screenshot_2018-09-07_12-34-07.png?version=1&modificationDate=1536345321576&api=v2" /></p>
<p style="margin: 10px 0 0 0">You should have also seen an <code style="font-family: monospace">== Endpoint webrtc_client is now Reachable</code> message on the Asterisk console (if you were looking).</p>
<p style="margin: 10px 0 0 0">Finally, click <code style="font-family: monospace">Call</code>...</p>
<p style="margin: 10px 0 0 0">You may be prompted to allow access to your microphone and camera. If so, allow them both.</p>
<p style="margin: 10px 0 0 0">Now, did you hear the <code style="font-family: monospace">You are the only person in this conference</code> prompt? Do you see yourself in the video preview window?</p>
<p style="margin: 10px 0 0 0"> <img class="confluence-embedded-image" width="500" src="/wiki/download/attachments/40818102/Screenshot_2018-09-07_12-40-16.png?version=1&modificationDate=1536345688115&api=v2" data-image-src="/wiki/download/attachments/40818102/Screenshot_2018-09-07_12-40-16.png?version=1&modificationDate=1536345688115&api=v2" /></p>
<p style="margin: 10px 0 0 0">That's it!</p>
<h1 id="InstallingandConfiguringCyberMegaPhone-BonusPoints" style="margin: 10px 0 0 0; font-size: 24px; font-weight: normal; line-height: 30px; margin: 40px 0 0 0">Bonus Points</h1>
<p style="margin: 10px 0 0 0">Have a friend or co-worker join the bridge. They can use the same <code style="font-family: monospace">webrtc_client</code> credentials.</p>
<p style="margin: 10px 0 0 0"> </p>
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