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<html xmlns="http://www.w3.org/1999/xhtml"><head><meta http-equiv="Content-Type" content="text/html; charset=UTF-8" /><title>rtp.conf</title><link rel="stylesheet" href="styles.css" type="text/css" /><meta name="generator" content="DocBook XSL Stylesheets V1.69.1" /><link rel="start" href="index.html" title="Asterisk™: The Future of Telephony" /><link rel="up" href="asterisk-APP-D.html" title="Appendix D. Configuration Files" /><link rel="prev" href="asterisk-APP-D-SECT-36.html" title="rpt.conf" /><link rel="next" href="I_sect1_tt2669.html" title="say.conf" /></head><body><div class="navheader"><table width="100%" summary="Navigation header"><tr><th colspan="3" align="center">rtp.conf</th></tr><tr><td width="20%" align="left"><a accesskey="p" href="asterisk-APP-D-SECT-36.html">Prev</a> </td><th width="60%" align="center">Appendix D. Configuration Files</th><td width="20%" align="right"> <a accesskey="n" href="I_sect1_tt2669.html">Next</a></td></tr></table><hr /></div><div class="sect1" lang="en" xml:lang="en"><div class="titlepage"><div><div><h2 class="title" style="clear: both"><a id="asterisk-APP-D-SECT-37"></a>rtp.conf</h2></div></div></div><p><a id="I_indexterm_tt2667" class="indexterm"></a>The <span class="emphasis"><em>rtp.conf</em></span> file controls the
    Real-time Transport Protocol (RTP) ports that Asterisk uses to generate
    and receive RTP traffic. The RTP protocol is used by SIP, H.323, MGCP, and
    possibly other protocols to carry media between endpoints.</p><p>The default <span class="emphasis"><em>rtp.conf</em></span> file uses the RTP port
    range of 10,000 through 20,000. However, this is far more ports than
    you’re likely to need, and many network administrators may not be
    comfortable opening up such a large range in their firewalls. You can
    limit the RTP port range by changing the upper and lower bound limits
    within the <span class="emphasis"><em>rtp.conf</em></span> file.</p><p>For every bidirectional SIP call between two endpoints, five ports
    are generally used: port 5060 for SIP signaling, one port for the data
    stream and one port for the Real-Time Control Protocol (RTCP) in one
    direction, and an additional two ports for the data stream and RTCP in the
    opposite direction.</p><p>UDP datagrams contain a 16-bit field for a Cyclic Redundancy Check
    (CRC), which is used to verify the integrity of the datagram header and
    its data. It uses polynomial division to create the 16-bit checksum from
    the 64-bit header. This value is then placed into the 16-bit CRC field of
    the datagram, which the remote end can then use to verify the integrity of
    the received datagram.</p><p>Setting <code class="literal">rtpchecksums=no</code> requests
    that the OS not do UDP checksum creating/checking for the sockets used by
    RTP. If you add this option to the sample <span class="emphasis"><em>rtp.conf</em></span>
    file, it will look like this:</p><a id="I_programlisting_tt2668"></a><pre class="programlisting">[general]
rtpstart=10000
rtpend=20000
rtpchecksums=no</pre></div><div class="navfooter"><hr /><table width="100%" summary="Navigation footer"><tr><td width="40%" align="left"><a accesskey="p" href="asterisk-APP-D-SECT-36.html">Prev</a> </td><td width="20%" align="center"><a accesskey="u" href="asterisk-APP-D.html">Up</a></td><td width="40%" align="right"> <a accesskey="n" href="I_sect1_tt2669.html">Next</a></td></tr><tr><td width="40%" align="left" valign="top">rpt.conf </td><td width="20%" align="center"><a accesskey="h" href="index.html">Home</a></td><td width="40%" align="right" valign="top"> say.conf</td></tr></table></div><div xmlns="" id="svn-footer"><hr /><p>You are reading <em>Asterisk: The Future of Telephony</em> (2nd Edition for Asterisk 1.4), by Jim van Meggelen, Jared Smith, and Leif Madsen.<br />
       This work is licensed under the <a href="http://creativecommons.org/licenses/by-nc-nd/3.0/">Creative Commons Attribution-Noncommercial-No Derivative Works License v3.0</a>.<br />
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