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<td id="header-avatar-image-container" valign="top" style="padding: 0px; border-collapse: collapse; vertical-align: top; width: 32px; padding-right: 9px"><a href="https://wiki.asterisk.org/wiki/display/~bford?src=email" style="color: #3b73af; text-decoration: none"><img id="header-avatar-image" class="image_fix" src="cid:avatar_69f4768fe914a834abed04bedb6511a8" height="32" width="32" border="0" style="border-radius: 3px; vertical-align: top" /></a></td>
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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. Don't forget, Asterisk 15.5 or better is required.</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">You may already have some of the config from previous webrtc endpoints for certificates, keys, encryption, ice support etc and think you don't need to add the magical <code style="font-family: monospace">webrtc=yes</code> but you do! The <code style="font-family: monospace">webrtc=yes</code> flag does more than just shortcut already existing flags which are needed for proper SFU support.</p>
<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>
<h1 id="InstallingandConfiguringCyberMegaPhone-Recommendations" style="margin: 10px 0 0 0; font-size: 24px; font-weight: normal; line-height: 30px; margin: 40px 0 0 0"> <span class="diff-html-added" id="added-diff-0" style="font-size: 100%; background-color: #ddfade;">Recommendations</span> </h1>
<p style="margin: 10px 0 0 0"> <span class="diff-html-added" style="font-size: 100%; background-color: #ddfade;">If you experience audio issues, it may be a good idea to turn on the jitterbuffer. This can cause the audio to be slightly delayed, but will also eliminate problems such as bursty audio packets causing disruptions. You can enable this option in confbridge.conf for a user, or you can do it through the dialplan before placing the user in the conference by using the JITTERBUFFER dialplan function for a more fine tuned experience.</span> </p>
<p style="margin: 10px 0 0 0"> </p>
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