rtp_media_server: adding module

kamailio · Nov 13, 2018 · 47e35db · 47e35db
1 parent 96aa799
commit 47e35db
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diff --git a/src/Makefile.groups b/src/Makefile.groups
@@ -207,6 +207,9 @@ mod_list_rabbitmq=rabbitmq
 # - modules depending on libphonenumber library
 mod_list_phonenum=phonenum
 
+# - modules depending on oRTP and mediastreamer2 libraries
+mod_list_rtp_media=rtp_media_server
+
 # - all modules
 mod_list_all=$(sort $(mod_list_basic) $(mod_list_extra) \
 			   $(mod_list_db) $(mod_list_dbuid) \
@@ -239,7 +242,8 @@ mod_list_all=$(sort $(mod_list_basic) $(mod_list_extra) \
 			   $(mod_list_erlang) $(mod_list_systemd) \
 			   $(mod_list_http_async) $(mod_list_nsq) \
 			   $(mod_list_rabbitmq) $(mod_list_jsdt)) \
-			   $(mod_list_phonenum)
+			   $(mod_list_phonenum) \
+			   $(mod_list_rtp_media)
 
 
 

diff --git a/src/modules/rtp_media_server/Makefile b/src/modules/rtp_media_server/Makefile
@@ -0,0 +1,13 @@
+include ../../Makefile.defs
+auto_gen=
+NAME=rtp_media_server.so
+
+DEFS+=-I$(LOCALBASE)/lib
+
+ORTPLIBS=-lortp
+BCUNITLIBS=-lbcunit
+MS2LIBS=-lmediastreamer_voip -lmediastreamer_base
+
+LIBS=$(ORTPLIBS) $(BCUNITLIBS) $(MS2LIBS)
+DEFS+=-DKAMAILIO_MOD_INTERFACE
+include ../../Makefile.modules
diff --git a/src/modules/rtp_media_server/config_example/kamailio.cfg b/src/modules/rtp_media_server/config_example/kamailio.cfg
@@ -0,0 +1,56 @@
+
+debug=3
+log_stderror=yes
+memdbg=5
+memlog=5
+
+# number of SIP routing processes
+children=5
+
+loadmodule "ctl"
+loadmodule "tm"
+loadmodule "tmx"
+loadmodule "sl"
+loadmodule "rr"
+loadmodule "pv"
+loadmodule "textops"
+loadmodule "siputils"
+loadmodule "xlog"
+loadmodule "nathelper"
+loadmodule "rtp_media_server"
+modparam("rtp_media_server", "log_file_name", "/tmp/rms_transfer.log")
+modparam("tm", "wt_timer", 1000)
+listen=udp:147.75.39.121:5090
+
+event_route[rms:start] {
+	xnotice("[rms:start] play ...\n");
+	rms_play("./voice_file/Bach_10s_8000.wav", "rms:after_play");
+};
+
+event_route[rms:after_play] {
+	xnotice("[rms:after_play] play done...\n");
+	rms_hangup();
+};
+
+route {
+	if (t_precheck_trans()) {
+		t_check_trans();
+		exit;
+	}
+	t_check_trans();
+
+	xnotice("[$rm][$ci]\n");
+	if (is_method("INVITE") && !has_totag()) {
+		fix_nated_contact();
+		if (!rms_answer()) {
+			t_reply("503", "server error");
+			xerr("rtp_media_server error!\n");
+			exit;
+		}
+	}
+	if (is_method("BYE")){
+		rms_media_stop();
+	}
+	exit;
+}
+
diff --git a/src/modules/rtp_media_server/doc/Makefile b/src/modules/rtp_media_server/doc/Makefile
@@ -0,0 +1,4 @@
+docs = rtp_media_server.xml
+
+docbook_dir = ../../../../doc/docbook
+include $(docbook_dir)/Makefile.module
diff --git a/src/modules/rtp_media_server/doc/rtp_media_server.xml b/src/modules/rtp_media_server/doc/rtp_media_server.xml
@@ -0,0 +1,42 @@
+<?xml version="1.0" encoding='ISO-8859-1'?>
+<!DOCTYPE book PUBLIC "-//OASIS//DTD DocBook XML V4.4//EN"
+"http://www.oasis-open.org/docbook/xml/4.4/docbookx.dtd" [
+
+<!-- Include general documentation entities -->
+<!ENTITY % docentities SYSTEM "../../../../doc/docbook/entities.xml">
+%docentities;
+
+]>
+
+<book xmlns:xi="http://www.w3.org/2001/XInclude">
+	<bookinfo>
+	<title>rtp_media_server Module</title>
+	<productname class="trade">&kamailio;</productname>
+	<authorgroup>
+        <author>
+        <firstname>Julien</firstname>
+        <surname>Chavanton</surname>
+        <email>jchavanton@gmail.com</email>
+        </author>
+		<author>
+		<firstname>Julien</firstname>
+		<surname>Chavanton</surname>
+		<affiliation><orgname>flowroute.com</orgname></affiliation>
+		<email>jchavanton@gmail.com</email>
+		</author>
+		<editor>
+		<firstname>Julien</firstname>
+		<surname>Chavanton</surname>
+		<affiliation><orgname>flowroute.com</orgname></affiliation>
+		<email>jchavanton@gmail.com</email>
+		</editor>
+	</authorgroup>
+	<copyright>
+		<year>2017-2018</year>
+		<holder>Flowroute.com</holder>
+	</copyright>
+	</bookinfo>
+	<toc></toc>
+
+	<xi:include href="rtp_media_server_admin.xml"/>
+</book>
diff --git a/src/modules/rtp_media_server/doc/rtp_media_server_admin.xml b/src/modules/rtp_media_server/doc/rtp_media_server_admin.xml
@@ -0,0 +1,193 @@
+<?xml version="1.0" encoding='ISO-8859-1'?>
+<!DOCTYPE book PUBLIC "-//OASIS//DTD DocBook XML V4.4//EN"
+"http://www.oasis-open.org/docbook/xml/4.4/docbookx.dtd" [
+
+<!-- Include general documentation entities -->
+<!ENTITY % docentities SYSTEM "../../../../doc/docbook/entities.xml">
+%docentities;
+
+]>
+
+<chapter>
+	<title>&adminguide;</title>
+
+<section>
+	<title>Overview</title>
+	<para>
+		rtp_media_server module is adding RTP and media processing functionalities to Kamailio
+	</para>
+	<para>
+		Kamailio is providing SIP signaling including and enpoint with Dialog state, SDP parsing and scripting language
+	</para><para>
+		oRTP: is providing Real-time Transport Protocol (RFC 3550)
+	</para><para>
+		mediastreamer2: is providing mediaprocessing functionnalities using graphs and filters, many modules are available
+		to support various features, it should be relatively simple to integrated them.
+	</para><para>
+		mediastreamer2 is also providing a framework to create custom mediaprocessing modules.
+	</para>
+</section>
+
+<section>
+	<title>Dependencies</title>
+	<section>
+		<title>&kamailio; Modules</title>
+		<para>
+		The module depends on the following modules (in the other words
+		the listed modules must be loaded before this module):
+		<itemizedlist>
+			<listitem>
+			<para><emphasis>tm</emphasis> - accounting module</para>
+			</listitem>
+		</itemizedlist>
+		</para>
+	</section>
+	<section>
+		<title>External Libraries or Applications</title>
+		<para>
+		The following libraries or applications must be installed
+		before running &kamailio; with this module loaded:
+		</para><para>
+		If you want to build oRTP and mediastreamer from source, you can use the provided script for Debian "install_bc.sh".
+		</para>
+		<itemizedlist>
+			<listitem>
+			<para>
+				<emphasis>oRTP</emphasis> git://git.linphone.org/ortp.git
+			</para>
+			<para>
+				oRTP is a library implemeting Real-time Transport Protocol (RFC 3550), distributed under GNU GPLv2 or proprietary license.
+			</para>
+			</listitem>
+			<listitem>
+			<para>
+				<emphasis>mediastreamer2</emphasis> git clone git://git.linphone.org/mediastreamer2.git
+			</para>
+			<para>
+				Mediastreamer2 is a powerful and lightweight streaming engine specialized for voice/video telephony applications.
+			</para>
+			</listitem>
+		</itemizedlist>
+	</section>
+</section>
+
+<section>
+	<title>Parameters</title>
+	<section>
+	<title><varname>log_file_name</varname> (string)</title>
+		<para>
+		oRTP and MediaStreamer2 log file settings
+		the log mask is not configurable :
+		MESSAGE | WARNING | ERROR | FATAL
+		levels are activated.
+		</para>
+		<para>
+		Default value is not-set (no logging to file).
+		</para>
+		<example>
+		<title>log_file_name example</title>
+		<programlisting format="linespecific">
+...
+modparam("rtp_media_server", "log_file_name", "/var/log/rms/rms_ortp.log")
+...
+		</programlisting></example>
+	</section>
+</section>
+
+<section>
+	<title>Functions</title>
+
+	<section id="rtp_media_server.f.rms_answer">
+		<title><varname>rms_answer</varname> ()</title>
+		<para>
+		Create a session and a call leg and call the event_route[rms:start]
+		config example
+		</para>
+		<example>
+		<title>usage example</title>
+		<programlisting format="linespecific">
+...
+event_route[rms:start] {
+	xnotice("[rms:start] play ...\n");
+	rms_play("/tmp/reference_8000.wav", "rms:after_play");
+};
+
+event_route[rms:after_play] {
+	xnotice("[rms:after_play] play done...\n");
+	rms_hangup();
+};
+
+route {
+	if (t_precheck_trans()) {
+		t_check_trans();
+		exit;
+	}
+	t_check_trans();
+	if (is_method("INVITE") &amp;&amp; !has_totag()) {
+		if (!rms_answer()) {
+			t_reply("503", "server error");
+		}
+	}
+
+	if (is_method("BYE")){
+		xnotice("BYE RECEIVED [$ci]\n");
+		rms_media_stop();
+	}
+...
+		</programlisting></example>
+	</section>
+
+	<section id="rtp_media_server.f.rms_hangup">
+		<title><varname>rms_hangup</varname> ()</title>
+		<para>
+		Send a BYE, delete the RTP session and the media ressources.
+		</para>
+		<example>
+		<title>usage example</title>
+		<programlisting format="linespecific">
+...
+	rms_hangup();
+...
+		</programlisting></example>
+	</section>
+
+	<section id="rtp_media_server.f.rms_media_stop">
+		<title><varname>rms_media_stop</varname> ()</title>
+		<para>
+		This should be called on reception of a BYE, this will
+		delete the RTP session and the media ressources.
+		and reply "200 OK".
+		</para>
+		<para>
+		If the SIP session is not found "481 Call/Transaction Does Not Exist"
+		is returned.
+		</para>
+		<example>
+		<title>usage example</title>
+		<programlisting format="linespecific">
+...
+	if (is_method("BYE")){
+		rms_media_stop();
+	}
+...
+		</programlisting></example>
+	</section>
+
+	<section id="rtp_media_server.f.rms_play">
+		<title><varname>rms_play</varname> ()</title>
+		<para>
+		Play a wav file, a resampler is automaticaly configured to resample
+		and convert stereo to mono if needed.
+		</para><para>
+		The second parameter is the event route that will be called when the file was played.
+		</para>
+		<example>
+		<title>usage example</title>
+		<programlisting format="linespecific">
+...
+	rms_play("file.wav", "event_route_name");
+...
+		</programlisting></example>
+	</section>
+</section>
+</chapter>
diff --git a/src/modules/rtp_media_server/install_bc.sh b/src/modules/rtp_media_server/install_bc.sh
@@ -0,0 +1,45 @@
+# https://github.com/jchavanton/rtp_media_server.git
+apt-get install automake autogen autoconf libtool pkg-config
+# bcunit
+git clone https://github.com/BelledonneCommunications/bcunit.git
+cd bcunit
+git checkout 29c556fa8ac1ab21fba1291231ffa8dea43cf32a
+./autogen.sh
+./configure
+make
+make install
+cd ..
+
+# bctoolbox
+apt-get install libmbedtls-dev
+git clone https://github.com/BelledonneCommunications/bctoolbox.git
+cd bctoolbox
+git checkout 971953a9fa4058e9c8a40ca4a3fa12d832445255
+./autogen.sh
+./configure
+make
+make install
+cd ..
+
+# oRTP
+git clone https://github.com/BelledonneCommunications/ortp.git
+git checkout 6e13ef49a55cdd19dae395c38cfff7ffa518a089
+cd ortp
+./autogen.sh
+./configure
+make
+make install
+cd ..
+
+# mediastreamer2
+apt-get install intltool libspeex-dev libspeexdsp-dev
+git clone https://github.com/BelledonneCommunications/mediastreamer2.git
+cd mediastreamer2
+git checkout d935123fc497d19a24019c6e7ae4fe0c5f19d55a
+./autogen.sh
+./configure --disable-sound --disable-video --enable-tools=no --disable-tests
+make
+make install
+cd ..
+
+ldconfig