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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/~mattf?src=email" style="color: #3b73af; text-decoration: none"><img id="header-avatar-image" class="image_fix" src="cid:avatar_9eca7bba63959d79ad6da1a09812b37a" height="32" width="32" border="0" style="border-radius: 3px; vertical-align: top" /></a></td>
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<td id="page-title-pattern-icon-image-container-cell" style="padding: 0px; border-collapse: collapse; width: 16px; padding: 9px 8px 0px 0px; mso-text-raise: 5px; mso-line-height-rule: exactly"><a href="https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5?src=email" title="page icon" style="vertical-align: top;; color: #3b73af; text-decoration: none"><img style="vertical-align: top; display: block;" src="cid:page-icon" alt="page icon" title="page icon" height="16" width="16" border="0" /></a></td>
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<h1 id="WebRTCtutorialusingSIPML5-TutorialOverview" 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">Tutorial Overview</h1>
<p style="margin: 10px 0 0 0">This tutorial demonstrates basic WebRTC support and functionality within Asterisk. Asterisk will be configured to support a remote WebRTC client, the <a href="https://www.doubango.org/sipml5/" class="external-link" rel="nofollow" style="color: #3b73af; text-decoration: none">sipml5</a> client, for the purposes of making calls to/from Asterisk within a web browser. You must be running a recent (as of September 2018) version of a Mozilla or Chromium based web browser.</p>
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<p style="margin: 10px 0 0 0; margin-top: 0"></p>
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<th class="diff-macro-title" style="background-color: transparent; text-align: left; font-weight: normal;padding: 5px;"><span class="icon macro-placeholder-icon" style="background-color: ;line-height: 20px;"><img src="https://wiki.asterisk.org/wiki/s/en_GB/5639/a252d7f5e75d7a8bf7047b4b2c92f71a56a8f048.48/_/images/icons/macrobrowser/dropdown/toc.png" style="padding-right: 5px; vertical-align: text-bottom;" /> </span>Table of Contents</th>
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<h1 id="WebRTCtutorialusingSIPML5-SetupAsterisk" 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">Setup Asterisk</h1>
<p style="margin: 10px 0 0 0">Follow the instructions at <a class="confluence-link unresolved" href="#" style="color: #3b73af; text-decoration: none">Configuring Asterisk for WebRTC Clients</a> before proceeding, The rest of this tutorial assumes that your PBX is reachable at <code style="font-family: monospace">pbx.example.com</code> and that the client is known as <code style="font-family: monospace">webrtc_client</code>.</p>
<p style="margin: 10px 0 0 0"> <span style="color: rgb(133,120,102);font-family: MuseoSans500 , sans-serif;font-size: 20.0px;">Configure Asterisk Dialplan</span> </p>
<p style="margin: 10px 0 0 0">We'll make a simple dialplan for receiving a test call from the sipml5 client.</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">title</td>
<td style="background-color: #fafafa; padding: 0 0 0 5px; font-size: 12px; text-align: left;; padding: 0px; border-collapse: collapse">/etc/asterisk/extensions.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">[default]
exten => 200,1,Answer()
same => n,Playback(demo-congrats)
same => n,Hangup()</pre> </td>
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<p style="margin: 10px 0 0 0"> </p>
<p style="margin: 10px 0 0 0">This instructs Asterisk to Answer a call to "200," to play a file named "demo-<span class="diff-html-removed" id="removed-diff-0" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">contracts</span><span class="diff-html-added" id="added-diff-0" style="font-size: 100%; background-color: #ddfade;">congrats</span>" (included in Asterisk's core sound file packages), and to hang up. To make the extension active, either restart Asterisk or issue a "dialplan reload" command from the Asterisk CLI.</p>
<h1 id="WebRTCtutorialusingSIPML5-BrowsersandWSS" style="margin: 10px 0 0 0; font-size: 24px; font-weight: normal; line-height: 30px; margin: 40px 0 0 0">Browsers and WSS</h1>
<p style="margin: 10px 0 0 0"> <span>When using WSS as a transport, Chrome and Firefox will not allow you, by default, to connect using WSS to a server with a self-signed certificate. Rather, you'll have to install a publicly-signed certificate into Asterisk. Or, you'll have to import the the self-signed certificate we made earlier into your browser's keychain, which is outside the scope of this Wiki. </span> </p>
<p style="margin: 10px 0 0 0"> <span>Or, for Firefox and Chrome, you can open a separate browser tab and point it to Asterisk's HTTPs server's TLS port and WS path, e.g. <code style="font-family: monospace">https://pbx.example.com:8089/ws</code>, and you can manually confirm the security exception.</span> </p>
<p style="margin: 10px 0 0 0"> <span> <br /> </span> </p>
<h1 id="WebRTCtutorialusingSIPML5-ConfigureSIPML5" style="margin: 10px 0 0 0; font-size: 24px; font-weight: normal; line-height: 30px; margin: 40px 0 0 0">Configure SIPML5</h1>
<p style="margin: 10px 0 0 0"> </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"> <p style="margin: 10px 0 0 0; margin-top: 0">SIPML5 is a useful client for testing Asterisk. Many real-world users explore other options that may include rolling your own client.</p> </td>
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<p style="margin: 10px 0 0 0">Next, visit <a href="https://sipml5.org" class="external-link" rel="nofollow" style="color: #3b73af; text-decoration: none">https://sipml5.org</a> - you'll be redirected to <a href="https://www.doubango.org/sipml5/" class="external-link" rel="nofollow" style="color: #3b73af; text-decoration: none">https://www.doubango.org/sipml5/</a> </p>
<p style="margin: 10px 0 0 0">Once there, click the "Enjoy our live demo" link to be directed to the sipml5 client.</p>
<p style="margin: 10px 0 0 0">In the Registration box, use configuration similar to the following:</p>
<p style="margin: 10px 0 0 0"> <img class="confluence-embedded-image" src="/wiki/download/attachments/27200129/Screenshot_2018-09-07_06-51-14.png?version=1&modificationDate=1536324753479&api=v2" data-image-src="/wiki/download/attachments/27200129/Screenshot_2018-09-07_06-51-14.png?version=1&modificationDate=1536324753479&api=v2" /></p>
<p style="margin: 10px 0 0 0">Here, we have input the following:</p>
<ul style="margin: 10px 0 0 0">
<li>Display Name is a free-form string</li>
<li>Private Identity is our username from our PJSIP auth object</li>
<li>Public Identity is in the format:
<ul style="margin: 10px 0 0 0; margin-top: 0">
<li>sip : (name of our PJSIP aor object) @ (IP Address of the Asterisk system)</li>
</ul> </li>
<li>Password is our password from our PJSIP auth object</li>
<li>Realm is "asterisk.org"</li>
</ul>
<p style="margin: 10px 0 0 0">Next, click the "Expert mode?" form button. It will open a new browser tab. In the Expert settings box, use a configuration similar to the following:</p>
<p style="margin: 10px 0 0 0"> <img class="confluence-embedded-image" width="500" src="/wiki/download/attachments/27200129/Screenshot_2018-09-07_06-57-13.png?version=1&modificationDate=1536325094792&api=v2" data-image-src="/wiki/download/attachments/27200129/Screenshot_2018-09-07_06-57-13.png?version=1&modificationDate=1536325094792&api=v2" /></p>
<p style="margin: 10px 0 0 0"> </p>
<p style="margin: 10px 0 0 0">Here, we have made the following changes:</p>
<ul style="margin: 10px 0 0 0">
<li>Checked the "Disable Video" box</li>
<li>Filled in the WebSocket Server URL using the format:<br />
<ul style="margin: 10px 0 0 0">
<li>wss : // (ip address of asterisk) : 8089 / ws</li>
</ul> </li>
<li>Checked the "Disable 3GPP Early IMS" box</li>
</ul>
<p style="margin: 10px 0 0 0">Click "Save" and return to the other demo tab with the Registration box.</p>
<p style="margin: 10px 0 0 0">Next, click "Login" and you should see <em>Connected</em> as such:</p>
<p style="margin: 10px 0 0 0"> <img class="confluence-embedded-image" src="/wiki/download/attachments/27200129/Screenshot_2018-09-07_06-58-43.png?version=1&modificationDate=1536325184562&api=v2" data-image-src="/wiki/download/attachments/27200129/Screenshot_2018-09-07_06-58-43.png?version=1&modificationDate=1536325184562&api=v2" /></p>
<p style="margin: 10px 0 0 0"> </p>
<p style="margin: 10px 0 0 0">You should see a corresponding connection happen on the Asterisk CLI. You can log into the Asterisk CLI by performing:</p>
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<th class="diff-macro-title" style="background-color: transparent; text-align: left; font-weight: normal;padding: 5px;"><span class="icon macro-placeholder-icon" style="background-color: ;line-height: 20px;"><img src="https://wiki.asterisk.org/wiki/s/en_GB/5639/a252d7f5e75d7a8bf7047b4b2c92f71a56a8f048.48/_/images/icons/macrobrowser/dropdown/noformat.png" style="padding-right: 5px; vertical-align: text-bottom;" /> </span>No Format</th>
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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"># asterisk -vvvr</pre> </td>
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<p style="margin: 10px 0 0 0"> </p>
<p style="margin: 10px 0 0 0">Then, you can LogOut and Login and see something like:</p>
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<th class="diff-macro-title" style="background-color: transparent; text-align: left; font-weight: normal;padding: 5px;"><span class="icon macro-placeholder-icon" style="background-color: ;line-height: 20px;"><img src="https://wiki.asterisk.org/wiki/s/en_GB/5639/a252d7f5e75d7a8bf7047b4b2c92f71a56a8f048.48/_/images/icons/macrobrowser/dropdown/noformat.png" style="padding-right: 5px; vertical-align: text-bottom;" /> </span>No Format</th>
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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">== WebSocket connection from '192.168.147.245:49976' for protocol 'sip' accepted using version '13'
-- Added contact 'sips:webrtc_client@192.168.147.245:49976;transport=ws;rtcweb-breaker=no' to AOR 'webrtc_client' with expiration of 200 seconds
== Endpoint webrtc_client is now Reachable
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<h2 id="WebRTCtutorialusingSIPML5-Makeatestcall" style="margin: 10px 0 0 0; font-size: 20px; font-weight: normal; line-height: 30px; margin: 40px 0 0 0">Make a test call</h2>
<p style="margin: 10px 0 0 0">In the sipml5 Call control box input <strong>200</strong>. Then press the Call button. You'll see a drop-down:</p>
<p style="margin: 10px 0 0 0"> <img class="confluence-embedded-image" width="500" src="/wiki/download/attachments/27200129/Screen%20Shot%202017-06-28%20at%202.25.55%20PM.png?version=1&modificationDate=1498681650233&api=v2" data-image-src="/wiki/download/attachments/27200129/Screen%20Shot%202017-06-28%20at%202.25.55%20PM.png?version=1&modificationDate=1498681650233&api=v2" /></p>
<p style="margin: 10px 0 0 0">Select "Audio" to continue. Once you do this, Firefox will display a popup asking permission to use your microphone:</p>
<p style="margin: 10px 0 0 0"> <img class="confluence-embedded-image" src="/wiki/download/attachments/27200129/Screen%20Shot%202017-06-28%20at%202.26.48%20PM.png?version=1&modificationDate=1498681650399&api=v2" data-image-src="/wiki/download/attachments/27200129/Screen%20Shot%202017-06-28%20at%202.26.48%20PM.png?version=1&modificationDate=1498681650399&api=v2" /></p>
<p style="margin: 10px 0 0 0">Click "Allow."</p>
<p style="margin: 10px 0 0 0">Next, the Call control box will indicate that the call is proceeding:</p>
<p style="margin: 10px 0 0 0"> <img class="confluence-embedded-image" width="500" src="/wiki/download/attachments/27200129/Screen%20Shot%202017-06-28%20at%202.27.40%20PM.png?version=1&modificationDate=1498681650614&api=v2" data-image-src="/wiki/download/attachments/27200129/Screen%20Shot%202017-06-28%20at%202.27.40%20PM.png?version=1&modificationDate=1498681650614&api=v2" /></p>
<p style="margin: 10px 0 0 0">Finally, when the call is connected, you will see <em>In Call</em>:</p>
<p style="margin: 10px 0 0 0"> <img class="confluence-embedded-image" width="500" src="/wiki/download/attachments/27200129/Screen%20Shot%202017-06-28%20at%202.28.37%20PM.png?version=1&modificationDate=1498681650772&api=v2" data-image-src="/wiki/download/attachments/27200129/Screen%20Shot%202017-06-28%20at%202.28.37%20PM.png?version=1&modificationDate=1498681650772&api=v2" /></p>
<p style="margin: 10px 0 0 0">and you will hear "Congratulations, you have successfully installed and executed the Asterisk open source PBX..."</p>
<p style="margin: 10px 0 0 0">You've just made your first call via WebRTC using Asterisk!</p>
<p style="margin: 10px 0 0 0"> </p>
<p style="margin: 10px 0 0 0"></p>
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