@@ -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)

new file mode 100644

@@ -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

new file mode 100755

@@ -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;

+}

+

new file mode 100644

@@ -0,0 +1,4 @@

+docs = rtp_media_server.xml

+

+docbook_dir = ../../../../doc/docbook

+include $(docbook_dir)/Makefile.module

new file mode 100644

@@ -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>

new file mode 100644

@@ -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") && !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>

new file mode 100755

@@ -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

new file mode 100644

@@ -0,0 +1,319 @@

+/*

+ * Copyright (C) 2017-2018 Julien Chavanton jchavanton@gmail.com

+ *

+ * This file is part of Kamailio, a free SIP server.

+ *

+ * Kamailio is free software; you can redistribute it and/or modify

+ * it under the terms of the GNU General Public License as published by

+ * the Free Software Foundation; either version 2 of the License, or

+ * (at your option) any later version

+ *

+ * Kamailio is distributed in the hope that it will be useful,

+ * but WITHOUT ANY WARRANTY; without even the implied warranty of

+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the

+ * GNU General Public License for more details.

+ *

+ * You should have received a copy of the GNU General Public License

+ * along with this program; if not, write to the Free Software

+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301 USA

+ */

+

+#include "../../core/mem/shm.h"

+#include "../../core/sr_module.h"

+#include "rtp_media_server.h"

+

+inline static void *ptr_shm_malloc(size_t size)

+{

+	return shm_malloc(size);

+}

+inline static void *ptr_shm_realloc(void *ptr, size_t size)

+{

+	return shm_realloc(ptr, size);

+}

+inline static void ptr_shm_free(void *ptr)

+{

+	shm_free(ptr);

+}

+

+typedef struct shared_global_vars

+{

+	MSFactory *ms_factory;

+	gen_lock_t lock;

+} shared_global_vars_t;

+

+static shared_global_vars_t *vars;

+

+MSFilterDesc *rms_ms_filter_descs[] = {&ms_alaw_dec_desc, &ms_alaw_enc_desc,

+		&ms_ulaw_dec_desc, &ms_ulaw_enc_desc, &ms_rtp_send_desc,

+		&ms_rtp_recv_desc, &ms_dtmf_gen_desc, &ms_volume_desc,

+		&ms_equalizer_desc, &ms_channel_adapter_desc, &ms_audio_mixer_desc,

+		&ms_tone_detector_desc, &ms_speex_dec_desc, &ms_speex_enc_desc,

+		&ms_speex_ec_desc, &ms_file_player_desc, &ms_file_rec_desc,

+		&ms_resample_desc,

+		//       &ms_opus_dec_desc,

+		//       &ms_opus_enc_desc,

+		NULL};

+

+static MSFactory *rms_create_factory()

+{

+	MSFactory *f = ms_factory_new();

+	int i;

+	for(i = 0; rms_ms_filter_descs[i] != NULL; i++) {

+		ms_factory_register_filter(f, rms_ms_filter_descs[i]);

+	}

+	ms_factory_init_plugins(f);

+	ms_factory_enable_statistics(f, TRUE);

+	ms_factory_reset_statistics(f);

+	return f;

+}

+

+int rms_media_init()

+{

+	OrtpMemoryFunctions ortp_memory_functions;

+	ortp_memory_functions.malloc_fun = ptr_shm_malloc;

+	ortp_memory_functions.realloc_fun = ptr_shm_realloc;

+	ortp_memory_functions.free_fun = ptr_shm_free;

+	ortp_set_memory_functions(&ortp_memory_functions);

+	ortp_init();

+	vars = shm_malloc(sizeof(shared_global_vars_t));

+	return 1;

+}

+

+static MSTicker *rms_create_ticker(char *name)

+{

+	MSTickerParams params;

+	params.name = name;

+	params.prio = MS_TICKER_PRIO_NORMAL;

+	return ms_ticker_new_with_params(&params);

+}

+

+void rms_media_destroy(call_leg_media_t *m)

+{

+	LM_INFO("rtp_session_destroy[%p]\n", m->rtps);

+	rtp_session_destroy(m->rtps);

+	m->rtps = NULL;

+	LM_INFO("ms_ticker[%p]\n", m->ms_ticker);

+	ms_ticker_destroy(m->ms_ticker);

+	m->ms_ticker = NULL;

+	LM_INFO("ms_factory_destroy[%p]\n", m->ms_factory);

+	ms_factory_destroy(m->ms_factory);

+	m->ms_factory = NULL;

+}

+

+int create_call_leg_media(call_leg_media_t *m)

+{

+	m->ms_factory = rms_create_factory();

+	// create caller RTP session

+	LM_INFO("RTP session [%s:%d]<>[%s:%d]\n", m->local_ip.s, m->local_port,

+			m->remote_ip.s, m->remote_port);

+	m->rtps = ms_create_duplex_rtp_session(m->local_ip.s, m->local_port,

+			m->local_port + 1, ms_factory_get_mtu(m->ms_factory));

+	rtp_session_set_remote_addr_full(m->rtps, m->remote_ip.s, m->remote_port,

+			m->remote_ip.s, m->remote_port + 1);

+	rtp_session_set_payload_type(m->rtps, m->pt->type);

+	rtp_session_enable_rtcp(m->rtps, FALSE);

+	// create caller filters : rtprecv1/rtpsend1/encoder1/decoder1

+	m->ms_rtprecv = ms_factory_create_filter(m->ms_factory, MS_RTP_RECV_ID);

+	m->ms_rtpsend = ms_factory_create_filter(m->ms_factory, MS_RTP_SEND_ID);

+

+	LM_INFO("codec[%s]\n", m->pt->mime_type);

+	m->ms_encoder = ms_factory_create_encoder(m->ms_factory, m->pt->mime_type);

+	if(!m->ms_encoder) {

+		LM_ERR("creating encoder failed.\n");

+		return 0;

+	}

+	m->ms_decoder = ms_factory_create_decoder(m->ms_factory, m->pt->mime_type);

+

+	/* set filter params */

+	ms_filter_call_method(m->ms_rtpsend, MS_RTP_SEND_SET_SESSION, m->rtps);

+	ms_filter_call_method(m->ms_rtprecv, MS_RTP_RECV_SET_SESSION, m->rtps);

+	return 1;

+}

+

+int rms_bridge(call_leg_media_t *m1, call_leg_media_t *m2)

+{

+	MSConnectionHelper h;

+	m1->ms_ticker = rms_create_ticker(NULL);

+

+	// direction 1

+	ms_connection_helper_start(&h);

+	ms_connection_helper_link(&h, m1->ms_rtprecv, -1, 0);

+	ms_connection_helper_link(&h, m2->ms_rtpsend, 0, -1);

+

+	// direction 2

+	ms_connection_helper_start(&h);

+	ms_connection_helper_link(&h, m2->ms_rtprecv, -1, 0);

+	ms_connection_helper_link(&h, m1->ms_rtpsend, 0, -1);

+

+	ms_ticker_attach_multiple(

+			m1->ms_ticker, m1->ms_rtprecv, m2->ms_rtprecv, NULL);

+

+	return 1;

+}

+

+static void rms_player_eof(

+		void *user_data, MSFilter *f, unsigned int event, void *event_data)

+{

+	if(event == MS_FILE_PLAYER_EOF) {

+		rms_session_info_t *si = (rms_session_info_t *)user_data;

+		si->action = RMS_DONE;

+	}

+	MS_UNUSED(f), MS_UNUSED(event_data);

+}

+

+int rms_stop_bridge(call_leg_media_t *m1, call_leg_media_t *m2)

+{

+	MSConnectionHelper h;

+	if(!m1->ms_ticker)

+		return -1;

+	if(m1->ms_rtpsend)

+		ms_ticker_detach(m1->ms_ticker, m1->ms_rtpsend);

+	if(m1->ms_rtprecv)

+		ms_ticker_detach(m1->ms_ticker, m1->ms_rtprecv);

+	if(m2->ms_rtpsend)

+		ms_ticker_detach(m1->ms_ticker, m2->ms_rtpsend);

+	if(m2->ms_rtprecv)

+		ms_ticker_detach(m1->ms_ticker, m2->ms_rtprecv);

+	rtp_stats_display(rtp_session_get_stats(m1->rtps),

+			" AUDIO BRIDGE offer RTP STATISTICS ");

+	rtp_stats_display(rtp_session_get_stats(m2->rtps),

+			" AUDIO BRIDGE answer RTP STATISTICS ");

+	ms_factory_log_statistics(m1->ms_factory);

+

+	ms_connection_helper_start(&h);

+	if(m1->ms_rtprecv)

+		ms_connection_helper_unlink(&h, m1->ms_rtprecv, -1, 0);

+	if(m2->ms_rtpsend)

+		ms_connection_helper_unlink(&h, m2->ms_rtpsend, 0, -1);

+

+	ms_connection_helper_start(&h);

+	if(m2->ms_rtprecv)

+		ms_connection_helper_unlink(&h, m2->ms_rtprecv, -1, 0);

+	if(m1->ms_rtpsend)

+		ms_connection_helper_unlink(&h, m1->ms_rtpsend, 0, -1);

+

+	if(m1->ms_rtpsend)

+		ms_filter_destroy(m1->ms_rtpsend);

+	if(m1->ms_rtprecv)

+		ms_filter_destroy(m1->ms_rtprecv);

+	if(m2->ms_rtpsend)

+		ms_filter_destroy(m2->ms_rtpsend);

+	if(m2->ms_rtprecv)

+		ms_filter_destroy(m2->ms_rtprecv);

+	return 1;

+}

+

+int rms_playfile(call_leg_media_t *m, char *file_name)

+{

+	int file_sample_rate = 8000;

+	if(!m->ms_player)

+		return 0;

+	ms_filter_add_notify_callback(m->ms_player, rms_player_eof, m->si, TRUE);

+	ms_filter_call_method(m->ms_player, MS_FILE_PLAYER_OPEN, (void *)file_name);

+	ms_filter_call_method(m->ms_player, MS_FILE_PLAYER_START, NULL);

+	ms_filter_call_method(m->ms_player, MS_FILTER_GET_SAMPLE_RATE, &file_sample_rate);

+	ms_filter_call_method(m->ms_resampler, MS_FILTER_SET_SAMPLE_RATE, &file_sample_rate);

+	ms_filter_call_method(m->ms_resampler, MS_FILTER_SET_OUTPUT_SAMPLE_RATE, &m->pt->clock_rate);

+	ms_filter_call_method(m->ms_resampler, MS_FILTER_SET_OUTPUT_NCHANNELS, &m->pt->channels);

+	LM_INFO("clock[%d][%d]\n", m->pt->clock_rate, file_sample_rate);

+	return 1;

+}

+

+int rms_start_media(call_leg_media_t *m, char *file_name)

+{

+	MSConnectionHelper h;

+	int channels = 1;

+	int file_sample_rate = 8000;

+	m->ms_ticker = rms_create_ticker(NULL);

+	if(!m->ms_ticker) goto error;

+	m->ms_player = ms_factory_create_filter(m->ms_factory, MS_FILE_PLAYER_ID);

+	if(!m->ms_player) goto error;

+	m->ms_resampler = ms_factory_create_filter(m->ms_factory, MS_RESAMPLE_ID);

+	if(!m->ms_resampler) goto error;

+	// m->ms_recorder = ms_factory_create_filter(m->ms_factory,

+	// MS_FILE_PLAYER_ID);

+	m->ms_voidsink = ms_factory_create_filter(m->ms_factory, MS_VOID_SINK_ID);

+	if(!m->ms_voidsink) goto error;

+	LM_INFO("m[%p]call-id[%p]\n", m, m->si->callid.s);

+

+	ms_filter_call_method(

+			m->ms_player, MS_FILTER_SET_OUTPUT_NCHANNELS, &channels);

+	ms_filter_call_method_noarg(m->ms_player, MS_FILE_PLAYER_START);

+	ms_filter_call_method(m->ms_player, MS_FILTER_GET_SAMPLE_RATE, &file_sample_rate);

+	ms_filter_call_method(m->ms_resampler, MS_FILTER_SET_SAMPLE_RATE, &file_sample_rate);

+	ms_filter_call_method(m->ms_resampler, MS_FILTER_SET_OUTPUT_SAMPLE_RATE, &m->pt->clock_rate);

+

+	// sending graph

+	ms_connection_helper_start(&h);

+	ms_connection_helper_link(&h, m->ms_player, -1, 0);

+

+	ms_connection_helper_link(&h, m->ms_resampler, 0, 0);

+	ms_connection_helper_link(&h, m->ms_encoder, 0, 0);

+	ms_connection_helper_link(&h, m->ms_rtpsend, 0, -1);

+

+	// receiving graph

+	ms_connection_helper_start(&h);

+	ms_connection_helper_link(&h, m->ms_rtprecv, -1, 0);

+	// ms_connection_helper_link(&h, m->ms_decoder, 0, 0);

+	ms_connection_helper_link(&h, m->ms_voidsink, 0, -1);

+

+	ms_ticker_attach_multiple(m->ms_ticker, m->ms_player, m->ms_rtprecv, NULL);

+	return 1;

+error:

+	LM_ERR(" can not start media!\n");

+	return 0;

+}

+

+int rms_stop_media(call_leg_media_t *m)

+{

+	MSConnectionHelper h;

+	if(!m->ms_ticker) {

+		LM_ERR("RMS STOP MEDIA\n");

+		return -1;

+	}

+	if(m->ms_player)

+		ms_ticker_detach(m->ms_ticker, m->ms_player);

+	if(m->ms_rtprecv)

+		ms_ticker_detach(m->ms_ticker, m->ms_rtprecv);

+

+	rtp_stats_display(

+			rtp_session_get_stats(m->rtps), " AUDIO SESSION'S RTP STATISTICS ");

+	ms_factory_log_statistics(m->ms_factory);

+

+	/*dismantle the sending graph*/

+	ms_connection_helper_start(&h);

+	if(m->ms_player)

+		ms_connection_helper_unlink(&h, m->ms_player, -1, 0);

+	if(m->ms_resampler)

+		ms_connection_helper_unlink(&h, m->ms_resampler, 0, 0);

+	if(m->ms_encoder)

+		ms_connection_helper_unlink(&h, m->ms_encoder, 0, 0);

+	if(m->ms_rtpsend)

+		ms_connection_helper_unlink(&h, m->ms_rtpsend, 0, -1);

+	/*dismantle the receiving graph*/

+	ms_connection_helper_start(&h);

+	if(m->ms_rtprecv)

+		ms_connection_helper_unlink(&h, m->ms_rtprecv, -1, 0);

+	if(m->ms_voidsink)

+		ms_connection_helper_unlink(&h, m->ms_voidsink, 0, -1);

+

+	if(m->ms_player)

+		ms_filter_destroy(m->ms_player);

+	if(m->ms_resampler)

+		ms_filter_destroy(m->ms_resampler);

+	if(m->ms_encoder)

+		ms_filter_destroy(m->ms_encoder);

+	if(m->ms_rtpsend) {

+		LM_ERR("detroy rtpsend\n");

+		ms_filter_destroy(m->ms_rtpsend);

+	} else {

+		LM_ERR("no rtpsend\n");

+	}

+	if(m->ms_rtprecv)

+		ms_filter_destroy(m->ms_rtprecv);

+	if(m->ms_voidsink)

+		ms_filter_destroy(m->ms_voidsink);

+

+	rms_media_destroy(m);

+	return 1;

+}

new file mode 100644

@@ -0,0 +1,101 @@

+/*

+ * Copyright (C) 2017-2018 Julien Chavanton jchavanton@gmail.com

+ *

+ * This file is part of Kamailio, a free SIP server.

+ *

+ * Kamailio is free software; you can redistribute it and/or modify

+ * it under the terms of the GNU General Public License as published by

+ * the Free Software Foundation; either version 2 of the License, or

+ * (at your option) any later version

+ *

+ * Kamailio is distributed in the hope that it will be useful,

+ * but WITHOUT ANY WARRANTY; without even the implied warranty of

+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the

+ * GNU General Public License for more details.

+ *

+ * You should have received a copy of the GNU General Public License

+ * along with this program; if not, write to the Free Software

+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301  USA

+ */

+

+#ifndef rms_media_h

+#define rms_media_h

+

+#include "../../core/mem/shm.h"

+#include <mediastreamer2/mediastream.h>

+#include <mediastreamer2/msrtp.h>

+#include <mediastreamer2/dtmfgen.h>

+#include <mediastreamer2/msfileplayer.h>

+#include <mediastreamer2/msfilerec.h>

+#include <mediastreamer2/msrtp.h>

+#include <mediastreamer2/mstonedetector.h>

+#include <mediastreamer2/msfilter.h>

+#include <mediastreamer2/mscommon.h>

+#include <ortp/ortp.h>

+#include <ortp/port.h>

+

+struct rms_session_info;

+

+typedef struct call_leg_media

+{

+	MSFactory *ms_factory;

+	RtpSession *rtps;

+	PayloadType *pt;

+	MSTicker *ms_ticker;

+	MSFilter *ms_encoder;

+	MSFilter *ms_decoder;

+	MSFilter *ms_rtprecv;

+	MSFilter *ms_rtpsend;

+	MSFilter *ms_resampler;

+	MSFilter *ms_player;

+	MSFilter *ms_recorder;

+	MSFilter *ms_dtmfgen;

+	MSFilter *ms_tonedet;

+	MSFilter *ms_voidsource;

+	MSFilter *ms_voidsink;

+	str local_ip;

+	int local_port;

+	str remote_ip;

+	int remote_port;

+	struct rms_session_info *si;

+} call_leg_media_t;

+

+int create_call_leg_media(call_leg_media_t *m);

+

+int rms_media_init();

+void rms_media_destroy();

+

+MSFactory *rms_get_factory();

+

+int rms_stop_media(call_leg_media_t *m);

+int rms_playfile(call_leg_media_t *m, char *file_name);

+int rms_start_media(call_leg_media_t *m, char *file_name);

+int rms_bridge(call_leg_media_t *m1, call_leg_media_t *m2);

+int rms_stop_bridge(call_leg_media_t *m1, call_leg_media_t *m2);

+

+extern MSFilterDesc ms_pcap_file_player_desc;

+extern MSFilterDesc ms_rtp_send_desc;

+extern MSFilterDesc ms_rtp_recv_desc;

+extern MSFilterDesc ms_udp_send_desc;

+extern MSFilterDesc ms_alaw_dec_desc;

+extern MSFilterDesc ms_alaw_enc_desc;

+extern MSFilterDesc ms_ulaw_dec_desc;

+extern MSFilterDesc ms_ulaw_enc_desc;

+extern MSFilterDesc ms_dtmf_gen_desc;

+extern MSFilterDesc ms_volume_desc;

+extern MSFilterDesc ms_equalizer_desc;

+extern MSFilterDesc ms_channel_adapter_desc;

+extern MSFilterDesc ms_audio_mixer_desc;

+extern MSFilterDesc ms_tone_detector_desc;

+extern MSFilterDesc ms_genericplc_desc;

+extern MSFilterDesc ms_file_player_desc;

+extern MSFilterDesc ms_file_rec_desc;

+extern MSFilterDesc ms_vad_dtx_desc;

+extern MSFilterDesc ms_speex_dec_desc;

+extern MSFilterDesc ms_speex_enc_desc;

+extern MSFilterDesc ms_speex_ec_desc;

+extern MSFilterDesc ms_opus_enc_desc;

+extern MSFilterDesc ms_opus_dec_desc;

+extern MSFilterDesc ms_resample_desc;

+

+#endif

new file mode 100644

@@ -0,0 +1,291 @@

+/*

+ * Copyright (C) 2017-2018 Julien Chavanton jchavanton@gmail.com

+ *

+ * This file is part of Kamailio, a free SIP server.

+ *

+ * Kamailio is free software; you can redistribute it and/or modify

+ * it under the terms of the GNU General Public License as published by

+ * the Free Software Foundation; either version 2 of the License, or

+ * (at your option) any later version

+ *

+ * Kamailio is distributed in the hope that it will be useful,

+ * but WITHOUT ANY WARRANTY; without even the implied warranty of

+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the

+ * GNU General Public License for more details.

+ *

+ * You should have received a copy of the GNU General Public License

+ * along with this program; if not, write to the Free Software

+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301 USA

+ */

+

+#include "rms_sdp.h"

+#include "rms_util.h"

+#include "../../core/data_lump.h"

+#include "../../core/parser/parse_content.h"

+

+// https://tools.ietf.org/html/rfc4566

+// (protocol version)

+const char *sdp_v = "v=0\r\n";

+// (session name)

+const char *sdp_s = "s=-\r\n";

+// (time the session is active)

+const char *sdp_t = "t=0 0\r\n";

+//"a=rtpmap:101 telephone-event/8000\r\n"

+//"a=fmtp:101 0-15\r\n";

+//"a=rtpmap:0 PCMU/8000\r\n"

+//"a=rtpmap:8 PCMA/8000\r\n"

+//"a=rtpmap:96 opus/48000/2\r\n"

+//"a=fmtp:96 useinbandfec=1\r\n";

+

+static char *rms_sdp_get_rtpmap(str body, int type_number)

+{

+	char *pos = body.s;

+	while((pos = strstr(pos, "a=rtpmap:"))) {

+		int id;

+		int sampling_rate;

+		char codec[64];

+		sscanf(pos, "a=rtpmap:%d %s/%d", &id, codec, &sampling_rate);

+		if(id == type_number) {

+			LM_INFO("[%d][%s/%d]\n", id, codec, sampling_rate);

+			return rms_char_dup(codec, 1);

+		}

+		pos++;

+	}

+	return NULL;