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<!DOCTYPE html>
<html lang="en">
<head>
<title>RTP Streaming</title>
<meta charset="utf-8">
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prettyPrint();
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<header>
<a href="/"><img src="images/logo.svg" alt=""></a>
<h2><<span>proxy</span>></h2><h1><span>USECALLMANAGER</span>.nz</h1><h2></<span>proxy</span>></h2>
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<article>
<h1>RTP Streaming</h1>
Audio can be directly streamed one-way to a Cisco phone using the new <code class="function">SIPCiscoPage</code> application either via a unicast or multicast RTP stream.<br>
<br>
<h2 id="Options">Options <a href="#Options" title="Link">link</a></h2>
The following application options are available.<br>
<br>
<table>
<tbody>
<tr>
<td><b>m(address)</b></td>
<td>Stream audio to the specified multicast IP address.</td>
</tr>
<tr>
<td><b>p(port)</b></td>
<td>Port on the phone to receive the audio, an even number between <code class="literal">20480</code> (default) and <code class="literal">32768</code>.</td>
</tr>
<tr>
<td><b>v(volume)</b></td>
<td>Set the receive volume percentage (<code class="literal">1</code>-<code class="literal">100</code>) on the phone's speaker.</td>
</tr>
<tr>
<td><b>d(message)</b></td>
<td>Display a message in the status line of the phone.</td>
</tr>
<tr>
<td><b>t(seconds)</b></td>
<td>Number of seconds to display status line message, default is <code class="literal">10</code>.</td>
</tr>
<tr>
<td><b>a</b></td>
<td>Play an alert beep on the phone.</td>
</tr>
<tr>
<td><b>o</b></td>
<td>Indicate that peers are off-hook when paging.</td>
</tr>
<tr>
<td><b>b</b></td>
<td>Page the phone even if it is busy (Off-Hook, Ringing, Busy or DND).</td>
</tr>
</tbody>
</table>
<br>
<h2 id="Unicast-Audio">Unicast Audio <a href="#Unicast-Audio" title="Link">link</a></h2>
Unicast uses a separate RTP audio stream for each phone.<br>
<br>
<code class="prettify lang-astexten">; Unicast page 301 and 302 and set speaker volume to 75%
exten => 380,1,SIPCiscoPage(301&302,ov(75)ad(From ${CALLERID(number)}))</code>
<br>
<br>
<h2 id="Multicast-Audio">Multicast Audio <a href="#Multicast-Audio" title="Link">link</a></h2>
Multicast uses only a single RTP audio stream for all phones. You must specify a multicast IP address.<br>
<br>
<code class="prettify lang-astexten">; Multicast page 301, 302 and 303
exten => 380,1,SIPCiscoPage(301&302&303,oam(224.0.1.1))</code>
<br>
<br>
<h2 id="External-Streaming">External Streaming <a href="#External-Streaming" title="Link">link</a></h2>
The <code class="literal">mediastream</code> script from the commands archive below can be used to stream a <code class="literal">.wav</code> file to multiple phones. See <a href="cgi-execute.html">CGI Execute</a> for more information.<br>
<br>
<a href="https://github.com/usecallmanagernz/commands/archive/v2.9.tar.gz" download><span class="icon">file_download</span> commands-2.9.tar.gz</a> (18K) <span class="icon">event</span> 16/12/2025 <span class="icon">security</span> SHA256:68056aa3b752872079d00680662449771583a57ccf7395097bd21dd27e0273ab.<br>
</article>
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