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📄 manpage.xml

📁 RTP Proxy,用于Voip网络数据的代理传递
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<?xml version='1.0' encoding='ISO-8859-1'?><!-- Use the following command to generate rtpproxy.8: --><!-- xsltproc /usr/local/share/xsl/docbook/manpages/docbook.xsl manpage.xml --><!DOCTYPE refentry PUBLIC "-//OASIS//DTD DocBook XML V4.2//EN""http://www.oasis-open.org/docbook/xml/4.2/docbookx.dtd" [  <!ENTITY dhfirstname "<firstname>Maxim</firstname>">  <!ENTITY dhsurname   "<surname>Sobolev</surname>">  <!-- Please adjust the date whenever revising the manpage. -->  <!ENTITY dhdate      "<date>Jun 16, 2008</date>">  <!-- SECTION should be 1-8, maybe w/ subsection other parameters are       allowed: see man(7), man(1). -->  <!ENTITY dhsection   "<manvolnum>8</manvolnum>">  <!ENTITY dhemail     "<email>sobomax@sippysoft.com</email>">  <!ENTITY dhusername  "janakj">  <!ENTITY dhucpackage "<refentrytitle>rtpproxy</refentrytitle>">  <!ENTITY dhpackage   "rtpproxy">  <!ENTITY debian      "<productname>Debian</productname>">  <!ENTITY gnu         "<acronym>GNU</acronym>">  <!ENTITY gpl         "&gnu; <acronym>GPL</acronym>">]><refentry>    <refentryinfo>	<address>	    &dhemail;	</address>	<author>	    &dhfirstname;	    &dhsurname;	</author>	<copyright>	    <year>2006</year>	    <holder>&dhusername;</holder>	</copyright>	&dhdate;    </refentryinfo>    <refmeta>	&dhucpackage;		&dhsection;    </refmeta>    <refnamediv>	<refname>&dhpackage;</refname>	<refpurpose>RTP (Real-time Transport Protocol) Proxy Server</refpurpose>    </refnamediv>    <refsynopsisdiv>	<cmdsynopsis>	    <command>&dhpackage;</command>	    	    <arg choice="opt"><option>-?</option></arg>	    <arg choice="opt"><option>-2</option></arg>	    <arg choice="opt"><option>-f</option></arg>	    <arg choice="opt"><option>-v</option></arg>	    <arg choice="opt"><option>-R</option></arg>	    <arg choice="opt"><option>-l</option> <replaceable>addr1<optional>/addr2</optional></replaceable></arg>	    <arg choice="opt"><option>-6</option> <replaceable>addr1<optional>/addr2</optional></replaceable></arg>	    <arg choice="opt"><option>-s</option> <replaceable>ctrl_socket</replaceable></arg>	    <arg choice="opt"><option>-t</option> <replaceable>tos</replaceable></arg>	    <arg choice="opt"><option>-p</option> <replaceable>pidfile</replaceable></arg>	    <arg choice="opt"><option>-T</option> <replaceable>max_ttl</replaceable></arg>	    <arg choice="opt">		<option>-r</option> <replaceable>rdir</replaceable>		<arg choice="opt"><option>-S</option> <replaceable>sdir</replaceable></arg>	    </arg>	    <arg choice="opt"><option>-m</option> <replaceable>min_port</replaceable></arg>	    <arg choice="opt"><option>-M</option> <replaceable>max_port</replaceable></arg>	    <arg choice="opt"><option>-u</option> <replaceable>uname<optional>:gname</optional></replaceable></arg>	    <arg choice="opt"><option>-F</option></arg>	</cmdsynopsis>    </refsynopsisdiv>    <refsect1>	<title>DESCRIPTION</title>	<para>	    <command>&dhpackage;</command> is a symmetric RTP proxy designed to	    be used in conjunction with the SIP Express Router (SER) or any	    other SIP proxy or SIP B2BUA capable of rewriting SDP bodies in SIP messages	    that it processes.	</para>	<para>	    The main purpose of rtpproxy is to make the communication between	    SIP user agents behind NAT(s) (Network Address Translator)	    possible. Several cases exists when direct end-to-end communication	    is not possible and RTP streams have to be relayed through another	    host. Rtpproxy can be used to setup such a relaying host.	</para>	<para>	    When two listen interfaces have been specified using the command	    line parameters described below then rtpproxy will enter so called	    bridging mode. In briding mode rtpproxy forwards RTP packets	    received on one interface to the other interface and vice	    versa. This mode can be used to forward RTP packets between	    networks without direct network level connectivy (provided that the	    host running rtpproxy has one interface in both of them). One	    particular application of bridging mode is IPv4/IPv6 traversal of	    RTP packets.	</para>	<para>	    When instructured by SER rtpproxy can also record the entire RTP	    session in a file on a local harddisk or play a pre-recorded file	    to the user agent (so called Music-on-Hold).	</para>    </refsect1>    <refsect1>	<title>OPTIONS</title>	<variablelist>	    <varlistentry>		<term><option>-?</option></term>		<listitem>		    <para>Show summary of options.</para>		</listitem>	    </varlistentry>	    <varlistentry>		<term><option>-2</option></term>		<listitem>		    <para>			Send every RTP packet twice in sessions that use			low-bitrate codecs. Only packets that are smaller than			128 bytes will be sent twice. This option can improve			audio quality on lossy links.		    </para>		</listitem>	    </varlistentry>	    <varlistentry>		<term><option>-f</option></term>		<listitem>		    <para>			Rtpproxy will stay in foreground mode if this option is			set.		    </para>		</listitem>	    </varlistentry>	    <varlistentry>		<term><option>-v</option></term>		<listitem>		    <para>Show version of program.</para>		</listitem>	    </varlistentry>	    <varlistentry>		<term><option>-l</option> <replaceable>addr1<optional>/addr2</optional></replaceable></term>		<listitem>		    <para>			IPv4 listen IP address(es). You can specify either			one or two addresses. If two addresses have been			specified then rtpproxy will work in bridging mode.		    </para>		</listitem>	    </varlistentry>	    <varlistentry>		<term><option>-6</option> <replaceable>addr1<optional>/addr2</optional></replaceable></term>		<listitem>		    <para>			IPv6 listen IP address(es). You can specify either one			or two addresses. If two addresses have been specified			then rtpproxy will work in bridging mode.		    </para>		</listitem>	    </varlistentry>	    <varlistentry>		<term><option>-s</option> <replaceable>ctrl_socket</replaceable></term>		<listitem>		    <para>			This parameter configures rtpproxy control socket. The			control socket is used by nathelper module of SER to 			create/modify/delete RTP sessions to be relayed. 			Format of <replaceable>ctrl_socket</replaceable> is 			&lt;type&gt;:&lt;socket&gt;. Following types are			supported:			<itemizedlist>			    <listitem>				<para>				    <emphasis>udp:</emphasis> Create UDP				    control socket. In this mode RTPProxy will				    listen on UDP for control messages from				    SER/nathelper.				</para>				<para>				    Example: -s udp:127.0.0.1:9000				</para>				<para>				    IP address can be '*' in which case				    rtpproxy will listen on all local				    interfaces. If omitted port 22222 is used.				</para>				<note>				    <para>					RTPProxy control protocol has no					built-in security mechanisms. Make sure					that you protect the listening IP and					port properly when using RTPProxy with					UDP control socket.				    </para>				</note>			    </listitem>			    <listitem>				<para>				    <emphasis>udp6:</emphasis> Create IPv6 UDP				    control socket. In this mode RTPProxy will				    listen on UDP/IPv6 for control messages				    from SER/nathelper.				</para>				<para>				    Example: -s udp6:::1:9000				</para>			    </listitem>			    <listitem>				<para>				    <emphasis>unix:</emphasis> Create UNIX				    domain socket for control interface. In				    this mode SER/nathelper and RTPProxy must				    be running on the same host. This is the				    default setting for both SER/nathelper and				    rtpproxy.				</para>				<para>Example: -s unix:/var/run/rtpproxy.sock</para>				<para>Default value is <filename>/var/run/rtpproxy.sock</filename>.</para>			    </listitem>			</itemizedlist>		    </para>		</listitem>	    </varlistentry>	    <varlistentry>		<term><option>-t</option> <replaceable>tos</replaceable></term>		<listitem>		    <para>			Set this ToS (Type of Service) in outgoing			packets. Default value is 0xB8.		    </para>		</listitem>	    </varlistentry>	    <varlistentry>		<term><option>-r</option> <replaceable>rec_dir</replaceable></term>		<listitem>		    <para>Directory where recorded RTP sessions will be stored.</para>		</listitem>	    </varlistentry>	    <varlistentry>		<term><option>-S</option> <replaceable>spool_dir</replaceable></term>		<listitem>		    <para>			Spool directory for RTP sessions being recorded. The			file will be moved to directory configured in			<option>-r</option> option after the session finishes.		    </para>		</listitem>	    </varlistentry>	    <varlistentry>		<term><option>-R</option></term>		<listitem>		    <para>			Do not record RTCP when recording an RTP session. This			option is disabled (rtpproxy will record RTCP) by default.		    </para>		</listitem>	    </varlistentry>	    <varlistentry>		<term><option>-p</option> <replaceable>pid_file</replaceable></term>		<listitem>		    <para>			This parameter configures the name of the file where			PID of running rtpproxy will be stored. Default is			<filename>/var/run/rtpproxy.pid</filename>.		    </para>		</listitem>	    </varlistentry>	    <varlistentry>		<term><option>-T</option> <replaceable>max_ttl</replaceable></term>		<listitem>		    <para>			Limit the maximum TTL (Time To Live) of outgoing IP packets to			the value of <replaceable>max_ttl</replaceable>.		    </para>		</listitem>	    </varlistentry>	    <varlistentry>	        <term><option>-m</option> <replaceable>min_port</replaceable></term>	        <listitem>	            <para>	                Set lower limit on UDP ports range that the RTPproxy uses for	                RTP/RTCP sessions to <replaceable>min_port</replaceable>.	                Default is 35000.	            </para>	        </listitem>	    </varlistentry>	    <varlistentry>	        <term><option>-M</option> <replaceable>max_port</replaceable></term>	        <listitem>	            <para>	                Set upper limit on UDP ports range that the RTPproxy uses for	                RTP/RTCP sessions to <replaceable>max_port</replaceable>.	                Default is 65000.	            </para>	        </listitem>	    </varlistentry>	    <varlistentry>	        <term><option>-u</option> <replaceable>uname<optional>:gname</optional></replaceable></term>	        <listitem>	            <para>	                Switch RTPproxy to UID identified by the <replaceable>uname</replaceable>	                and optional GID identified by <replaceable>gname</replaceable> when	                proxy is up and running.	            </para>	        </listitem>	    </varlistentry>	    <varlistentry>	        <term><option>-F</option></term>	        <listitem>	            <para>	                By default the RTPproxy will warn user if running as superuser (UID 0) in	                local control mode and refuse to run in remote control mode at all.	                This switch removes the check.	            </para>	        </listitem>	    </varlistentry>	</variablelist>    </refsect1>    <refsect1>	<title>HowItWorks</title>	<para>	    When SER receives an INVITE request, it extracts Call-ID from it	    and communicates it to rtpproxy via Unix domain socket or	    UDP. Rtproxy looks for an existing session with such Call-ID. If	    the session exists it returns UDP port for that session, if not,	    then it creates a new session, binds to a first empty UDP port from	    the range specified at the compile time and returns number of that	    port to a SER. After receiving reply from the proxy, SER replaces	    media ip:port in the SDP to point to the proxy and forwards request	    as usually.	</para>	<para>	    When SER receives a non-negative SIP reply with SDP it again extracts	    Call-ID from it and communicates it to the proxy. In this case the proxy	    does not allocate a new session if it doesn't exist, but simply performs a	    lookup among existing sessions and returns either a port number if the	    session is found, or error code indicating that there is no session with	    such id. After receiving positive reply from the proxy, SER replaces media	    ip:port in the SIP reply to point to the proxy and forwards reply as	    usually.	</para>	<para>	    After the session has been created, the proxy listens on the port	    it has allocated for that session and waits for receiving at least	    one UDP packet from each of two parties participating in the	    call. Once such packet is received, the proxy fills one of two	    ip:port structures associated with each call with source ip:port of	    that packet. When both structures are filled in, the proxy starts	    relaying UDP packets between parties.	</para>	<para>	    The proxy tracks idle time for each of existing sessions (i.e. the time	    within which there were no packets relayed), and automatically cleans	    up a sessions whose idle times exceed the value specified at compile	    time (60 seconds by default).	</para>    </refsect1>    <refsect1>	<title>FILES</title>	<para>	    <filename>/usr/sbin/rtpproxy</filename>	</para>    </refsect1>    <refsect1>	<title>LICENSE</title>	<para>	    This program is licensed under the BSD license. See	    <filename>COPYING</filename> file in the rtpproxy sources for details.	</para>    </refsect1>    <refsect1>	<title>AVAILABILITY</title>	<para>	    The latest version of this program can be found at 	    <ulink url="http://www.rtpproxy.org/">http://www.rtpproxy.org/</ulink>.	</para>    </refsect1>    <refsect1>	<title>SEEALSO</title>	<para>ser(8).</para>    </refsect1></refentry>

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