openmeetings-server/src/site/xdoc/voip-sip-integration.xml - openmeetings - Git at Google

 <?xml version="1.0" encoding="UTF-8"?>
 <!--
    Licensed under the Apache License, Version 2.0 (the "License");
    you may not use this file except in compliance with the License.
    You may obtain a copy of the License at

        http://www.apache.org/licenses/LICENSE-2.0

    Unless required by applicable law or agreed to in writing, software
    distributed under the License is distributed on an "AS IS" BASIS,
    WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
    See the License for the specific language governing permissions and
    limitations under the License.
  -->
 <document xmlns="http://maven.apache.org/XDOC/2.0"
 		xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance"
 		xsi:schemaLocation="http://maven.apache.org/XDOC/2.0 http://maven.apache.org/xsd/xdoc-2.0.xsd">
 	<properties>
 		<title>VoIP and SIP Integration</title>
 		<author email="dev@openmeetings.apache.org">Apache OpenMeetings Team</author>
 	</properties>
 	<body>
 		<section name="Not implemented">
 			<div class="bd-callout bd-callout-danger">
 				Please NOTE: this functionality is not yet implemented in 5.0.x
 			</div>
 		</section>

 		<section name="VoIP and SIP Integration">
 			<p>
 				There are multiple ways to integrate with VoIP and or SIP.
 				OpenMeetings does not provide out of the box a ready to run VoIP
 				integration / integration to cell phone or usual land lane.
 				The
 				nature of such integrations is that it depends heavily on the
 				infrastructure that you are using and where you would like to
 				integrate OpenMeetings into.
 				<br />
 				<br />
 				It also depends on a number of factors of which OpenMeetings is
 				impossible to set up for you, for example setting up your VoIP
 				server or provide you with a range of telephone numbers reserved for
 				conference calls in your national phone network.
 				Such an integration
 				project is likely to become a consulting job for a
 				telecommunications consultant.
 				<br />
 				<br />
 				To get help on the integration you can contact the
 				<a href="mailing-lists.html">mailing lists</a>
 				or for example somebody from the list of
 				<a href="commercial-support.html">commercial support</a>.
 				<br/><br/>
 			</p>
 		</section>
 		<section name="SIP-Transport Integration">
 			<p>You need Apache OpenMeetings <strong>version 4.0+</strong> to apply this guide!</p>
 			<p>You need Asterisk <strong>version 13+</strong> to apply this guide!</p>
 			<p>Here is instruction how-to set up red5sip transport integration with OpenMeetings on Ubuntu 16.04.</p>
 		</section>
 		<section name="Prerequisites">
 			<div>
 				Run the commands
 				<source>
 <![CDATA[
 sudo apt update && sudo apt upgrade
 ]]>
 				</source>
 			</div>
 		</section>
 		<section name="Building and setting up Asterisk">
 			<div>
 				Run the commands
 				<source>
 <![CDATA[
 sudo mkdir /usr/src/asterisk && cd /usr/src/asterisk
 sudo wget http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-13.17.0.tar.gz
 sudo tar -xvzf asterisk-13.17.0.tar.gz
 cd ./asterisk-13.17.0
 sudo make clean
 sudo contrib/scripts/install_prereq install
 sudo ./configure
 sudo make menuconfig
 ]]>
 				</source>
 				Make sure you have selected  <tt>Add-ons -> res_config_mysql</tt>, Press F12 to save
 				<source>
 <![CDATA[
 sudo make
 sudo make install
 sudo make samples
 sudo make config
 sudo service asterisk start
 ]]>
 				</source>
 			</div>
 		</section>
 		<section name="Configure Asterisk">
 			<div>
 				Enable asterisk MySQL module:<br /><br />
 				Modify "[modules]" section of <tt>/etc/asterisk/modules.conf</tt> as follows:<br />
 				<strong>Add/uncomment the following lines</strong>
 				<source>
 <![CDATA[
 preload => res_config_mysql.so
 ]]>
 				</source>
 			</div><br />
 			<div>
 				Configure MySQL module:<br /><br />
 				Set valid data for MySQL in <tt>/etc/asterisk/res_config_mysql.conf</tt> :<br />
 				<strong>Example</strong>
 				<source>
 <![CDATA[
 [general]
 dbhost = 127.0.0.1
 dbname = openmeetings
 dbuser = root
 dbpass =
 dbport = 3306
 dbsock = /var/lib/mysql/mysql.sock
 dbcharset = utf8
 requirements=warn
 ]]>
 				</source>
 			</div><br />
 			<div>
 				Modify <tt>/etc/asterisk/sip.conf</tt><br />
 				<strong>Add/uncomment the following line</strong>:<br />
 				<source>
 <![CDATA[
 videosupport=yes
 rtcachefriends=yes
 ]]>
 				</source>
 				<strong>Increase maxexpiry value to 43200</strong>:<br />
 				<source>
 <![CDATA[
 maxexpiry=43200
 ]]>
 				</source>
 				<strong>Add user for the "SIP Transport"</strong>:<br />
 				<source>
 <![CDATA[
 [red5sip_user]
 type=friend
 secret=12345
 disallow=all
 allow=ulaw
 allow=h263
 host=dynamic
 nat=force_rport,comedia
 context=rooms-red5sip
 ]]>
 				</source>
 			</div><br />
 			<div>
 				Add next lines into the <tt>/etc/asterisk/extconfig.conf</tt>:
 				<source>
 <![CDATA[
 [settings]
 sippeers => mysql,general,sipusers
 ]]>
 				</source>
 			</div><br />
 			<div>
 				Modify <tt>/etc/asterisk/extensions.conf</tt><br />
 				<strong>Add the following section</strong>:<br />
 				<source>
 <![CDATA[
 ; *****************************************************
 ; The below dial plan is used to dial into a Openmeetings Conference room
 ; The first line DB_EXISTS(openmeetings/room/ does not belong to the openmeetings application but is the name of astDB containing the astDB family/key pair and values
 ; To Check if your astDB has been created do the following in a terminal window type the following:
 ; asterisk –rx “database show”
 ; If you do not receive an output with that resembles openmeetings/rooms/400## where “##” will equal the extension assigned when you created your room
 ; If you do not receive the above output check your parameters in /opt/om/webapps/openmeetings/WEB-INF/classes/applicationContext.xml
 ; Go back into the Administrator Panel and remove the PIN number in each room save the record with no PIN number and then re-enter the pin again resave the record.
 ; *****************************************************

 [rooms]
 exten => _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavail)
 exten => _400X!,n(ok),SET(PIN=${DB(openmeetings/rooms/${EXTEN})})
 exten => _400X!,n,Set(CONFBRIDGE(user,template)=sip_user)
 exten => _400X!,n,Set(CONFBRIDGE(user,pin)=${PIN})
 exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,)
 exten => _400X!,n,Hangup
 exten => _400X!,n(notavail),Answer()
 exten => _400X!,n,Playback(invalid)
 exten => _400X!,n,Hangup

 [rooms-originate]
 exten => _400X!,1,Confbridge(${EXTEN},default_bridge,sip_user)
 exten => _400X!,n,Hangup

 [rooms-out]
 ; *****************************************************
 ; Extensions for outgoing calls from Openmeetings room.
 ; *****************************************************

 [rooms-red5sip]
 exten => _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavail)
 exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,red5sip_user)
 exten => _400X!,n(notavail),Hangup
 ]]>
 				</source>
 			</div><br />
 			<div>
 				Modify <tt>/etc/asterisk/confbridge.conf</tt><br />
 				<strong>Add/Modify the following secions</strong>:<br />
 				<source>
 <![CDATA[
 [general]

 [red5sip_user]
 type=user
 marked=yes
 dsp_drop_silence=yes
 denoise=true

 [sip_user]
 type=user
 end_marked=yes
 wait_marked=yes
 music_on_hold_when_empty=yes
 dsp_drop_silence=yes
 denoise=true

 [default_bridge]
 type=bridge
 video_mode=follow_talker
 ]]>
 				</source>
 			</div><br />
 			<div>
 				To enable Asterisk Manager API modify <tt>/etc/asterisk/manager.conf</tt><br />
 				<strong>Add/Modify the following sections</strong>:<br />
 				<source>
 <![CDATA[
 [general]
 enabled = yes
 webenabled = no
 port = 5038
 bindaddr = 127.0.0.1

 [openmeetings]
 secret = 12345
 deny=0.0.0.0/0.0.0.0
 permit=127.0.0.1/255.255.255.0
 read = all
 write = all
 ]]>
 				</source>
 			</div><br />
 			<div>
 				Update OpenMeetings with credentials for Asterisk manager.
 				Modify <tt>/opt/om/webapps/openmeetings/WEB-INF/classes/applicationContext.xml</tt><br />
 				find <strong>&lt;bean id="sipDao" class="org.apache.openmeetings.db.dao.room.SipDao"&gt;</strong>
 				uncomment its parameters and set it to your custom values.<br/>
 				set value for <tt>uid</tt> property to unique secret value (can be generated here <a href="https://www.uuidtools.com">https://www.uuidtools.com</a>)
 				and sync it with <tt>settings.properties</tt> of red5sip (see below)
 				<p style="font-size: larger; color: blue;">
 					IMPORTANT: this step should be done <strong>BEFORE</strong> system install/restore
 					otherwise all SIP related room information will be lost
 				</p>
 			</div><br />
 			<div>
 				Restart asterisk:
 				<source>
 <![CDATA[
 service asterisk restart
 ]]>
 				</source>
 			</div><br />
 		</section>

 		<section name="Setup red5sip transport">
 			<ul>
 				<li>Download red5sip from <tt>https://github.com/openmeetings/red5sip</tt>
 					<source>
 <![CDATA[
 git clone https://github.com/openmeetings/red5sip.git
 ]]>
 					</source>
 				</li>
 				<li>Build with Apache Maven
 					<source>
 <![CDATA[
 cd red5sip
 mvn clean package
 ]]>
 					</source>
 				</li>
 				<li>All necessary files will be available in <tt>target</tt> folder, copy/move it to /opt/red5sip/</li>
 				<li>Insert proper values to the <tt>/opt/red5sip/settings.properties</tt>
 					<source>
 <![CDATA[
 red5.host=127.0.0.1 # red5 server address
 om.context=openmeetings # Openmeetings context
 red5.codec=asao
 red5.codec.rate=22 # should correlate with mic setting in Admin->Config `flash.mic.rate`
 sip.obproxy=127.0.0.1 # asterisk adderss
 sip.phone=red5sip_user # sip phone number
 sip.authid=red5sip_user # sip auth id
 sip.secret=12345 # sip password
 sip.realm=asterisk # sip realm
 sip.proxy=127.0.0.1 # address of sip proxy
 rooms.forceStart=no # TBD
 uid=87dddad4-9ca5-475b-860f-2e0825d02b76 #can be generated here: https://www.uuidtools.com/
 rooms=1 # TBD (not in use)
 ]]>
 					</source>
 				</li>
 				<li>Set correct permissions on red5sip files:
 					<source>
 <![CDATA[
 sudo chown -R nobody:nogroup /opt/red5sip
 ]]>
 					</source>
 				</li>
 				<li>Add red5sip to autostart:
 					<source>
 <![CDATA[
 sudo cp /opt/red5sip/red5sip /etc/init.d/
 sudo chmod a+x /etc/init.d/red5sip
 sudo update-rc.d red5sip defaults
 ]]>
 					</source>
 				</li>
 				<li>Start openmeetings
 					<source>
 <![CDATA[
 service red5 start
 ]]>
 					</source>
 				</li>
 				<li>
 					Enable <tt>SIP</tt> in openmeetings: <br/>
 					<tt>Administration->Configuration->red5sip.enable == yes</tt>
 				</li>
 				<li>
 					Enable SIP for particular room(s): <br/>
 					<tt>Administration->Conference rooms->Room->Enable SIP transport in the room == checked</tt><br/>
 					(SIP number will be assigned to room if everything is OK)
 				</li>
 				<li>Start red5sip
 					<source>
 <![CDATA[
 service red5sip start
 ]]>
 					</source>
 				</li>
 			</ul>
 		</section>
 	</body>
 </document>
	<?xml version="1.0" encoding="UTF-8"?>
	<!--
	Licensed under the Apache License, Version 2.0 (the "License");
	you may not use this file except in compliance with the License.
	You may obtain a copy of the License at

	http://www.apache.org/licenses/LICENSE-2.0

	Unless required by applicable law or agreed to in writing, software
	distributed under the License is distributed on an "AS IS" BASIS,
	WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
	See the License for the specific language governing permissions and
	limitations under the License.
	-->
	<document xmlns="http://maven.apache.org/XDOC/2.0"
	xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance"
	xsi:schemaLocation="http://maven.apache.org/XDOC/2.0 http://maven.apache.org/xsd/xdoc-2.0.xsd">
	<properties>
	<title>VoIP and SIP Integration</title>
	<author email="dev@openmeetings.apache.org">Apache OpenMeetings Team</author>
	</properties>
	<body>
	<section name="Not implemented">
	<div class="bd-callout bd-callout-danger">
	Please NOTE: this functionality is not yet implemented in 5.0.x
	</div>
	</section>

	<section name="VoIP and SIP Integration">
	<p>
	There are multiple ways to integrate with VoIP and or SIP.
	OpenMeetings does not provide out of the box a ready to run VoIP
	integration / integration to cell phone or usual land lane.
	The
	nature of such integrations is that it depends heavily on the
	infrastructure that you are using and where you would like to
	integrate OpenMeetings into.
	<br />
	<br />
	It also depends on a number of factors of which OpenMeetings is
	impossible to set up for you, for example setting up your VoIP
	server or provide you with a range of telephone numbers reserved for
	conference calls in your national phone network.
	Such an integration
	project is likely to become a consulting job for a
	telecommunications consultant.
	<br />
	<br />
	To get help on the integration you can contact the
	<a href="mailing-lists.html">mailing lists</a>
	or for example somebody from the list of
	<a href="commercial-support.html">commercial support</a>.
	<br/><br/>
	</p>
	</section>
	<section name="SIP-Transport Integration">
	<p>You need Apache OpenMeetings <strong>version 4.0+</strong> to apply this guide!</p>
	<p>You need Asterisk <strong>version 13+</strong> to apply this guide!</p>
	<p>Here is instruction how-to set up red5sip transport integration with OpenMeetings on Ubuntu 16.04.</p>
	</section>
	<section name="Prerequisites">
	<div>
	Run the commands
	<source>
	<![CDATA[
	sudo apt update && sudo apt upgrade
	]]>
	</source>
	</div>
	</section>
	<section name="Building and setting up Asterisk">
	<div>
	Run the commands
	<source>
	<![CDATA[
	sudo mkdir /usr/src/asterisk && cd /usr/src/asterisk
	sudo wget http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-13.17.0.tar.gz
	sudo tar -xvzf asterisk-13.17.0.tar.gz
	cd ./asterisk-13.17.0
	sudo make clean
	sudo contrib/scripts/install_prereq install
	sudo ./configure
	sudo make menuconfig
	]]>
	</source>
	Make sure you have selected <tt>Add-ons -> res_config_mysql</tt>, Press F12 to save
	<source>
	<![CDATA[
	sudo make
	sudo make install
	sudo make samples
	sudo make config
	sudo service asterisk start
	]]>
	</source>
	</div>
	</section>
	<section name="Configure Asterisk">
	<div>
	Enable asterisk MySQL module:<br /><br />
	Modify "[modules]" section of <tt>/etc/asterisk/modules.conf</tt> as follows:<br />
	<strong>Add/uncomment the following lines</strong>
	<source>
	<![CDATA[
	preload => res_config_mysql.so
	]]>
	</source>
	</div><br />
	<div>
	Configure MySQL module:<br /><br />
	Set valid data for MySQL in <tt>/etc/asterisk/res_config_mysql.conf</tt> :<br />
	<strong>Example</strong>
	<source>
	<![CDATA[
	[general]
	dbhost = 127.0.0.1
	dbname = openmeetings
	dbuser = root
	dbpass =
	dbport = 3306
	dbsock = /var/lib/mysql/mysql.sock
	dbcharset = utf8
	requirements=warn
	]]>
	</source>
	</div><br />
	<div>
	Modify <tt>/etc/asterisk/sip.conf</tt><br />
	<strong>Add/uncomment the following line</strong>:<br />
	<source>
	<![CDATA[
	videosupport=yes
	rtcachefriends=yes
	]]>
	</source>
	<strong>Increase maxexpiry value to 43200</strong>:<br />
	<source>
	<![CDATA[
	maxexpiry=43200
	]]>
	</source>
	<strong>Add user for the "SIP Transport"</strong>:<br />
	<source>
	<![CDATA[
	[red5sip_user]
	type=friend
	secret=12345
	disallow=all
	allow=ulaw
	allow=h263
	host=dynamic
	nat=force_rport,comedia
	context=rooms-red5sip
	]]>
	</source>
	</div><br />
	<div>
	Add next lines into the <tt>/etc/asterisk/extconfig.conf</tt>:
	<source>
	<![CDATA[
	[settings]
	sippeers => mysql,general,sipusers
	]]>
	</source>
	</div><br />
	<div>
	Modify <tt>/etc/asterisk/extensions.conf</tt><br />
	<strong>Add the following section</strong>:<br />
	<source>
	<![CDATA[
	; *****************************************************
	; The below dial plan is used to dial into a Openmeetings Conference room
	; The first line DB_EXISTS(openmeetings/room/ does not belong to the openmeetings application but is the name of astDB containing the astDB family/key pair and values
	; To Check if your astDB has been created do the following in a terminal window type the following:
	; asterisk –rx “database show”
	; If you do not receive an output with that resembles openmeetings/rooms/400## where “##” will equal the extension assigned when you created your room
	; If you do not receive the above output check your parameters in /opt/om/webapps/openmeetings/WEB-INF/classes/applicationContext.xml
	; Go back into the Administrator Panel and remove the PIN number in each room save the record with no PIN number and then re-enter the pin again resave the record.
	; *****************************************************

	[rooms]
	exten => _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavail)
	exten => _400X!,n(ok),SET(PIN=${DB(openmeetings/rooms/${EXTEN})})
	exten => _400X!,n,Set(CONFBRIDGE(user,template)=sip_user)
	exten => _400X!,n,Set(CONFBRIDGE(user,pin)=${PIN})
	exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,)
	exten => _400X!,n,Hangup
	exten => _400X!,n(notavail),Answer()
	exten => _400X!,n,Playback(invalid)
	exten => _400X!,n,Hangup

	[rooms-originate]
	exten => _400X!,1,Confbridge(${EXTEN},default_bridge,sip_user)
	exten => _400X!,n,Hangup

	[rooms-out]
	; *****************************************************
	; Extensions for outgoing calls from Openmeetings room.
	; *****************************************************

	[rooms-red5sip]
	exten => _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavail)
	exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,red5sip_user)
	exten => _400X!,n(notavail),Hangup
	]]>
	</source>
	</div><br />
	<div>
	Modify <tt>/etc/asterisk/confbridge.conf</tt><br />
	<strong>Add/Modify the following secions</strong>:<br />
	<source>
	<![CDATA[
	[general]

	[red5sip_user]
	type=user
	marked=yes
	dsp_drop_silence=yes
	denoise=true

	[sip_user]
	type=user
	end_marked=yes
	wait_marked=yes
	music_on_hold_when_empty=yes
	dsp_drop_silence=yes
	denoise=true

	[default_bridge]
	type=bridge
	video_mode=follow_talker
	]]>
	</source>
	</div><br />
	<div>
	To enable Asterisk Manager API modify <tt>/etc/asterisk/manager.conf</tt><br />
	<strong>Add/Modify the following sections</strong>:<br />
	<source>
	<![CDATA[
	[general]
	enabled = yes
	webenabled = no
	port = 5038
	bindaddr = 127.0.0.1

	[openmeetings]
	secret = 12345
	deny=0.0.0.0/0.0.0.0
	permit=127.0.0.1/255.255.255.0
	read = all
	write = all
	]]>
	</source>
	</div><br />
	<div>
	Update OpenMeetings with credentials for Asterisk manager.
	Modify <tt>/opt/om/webapps/openmeetings/WEB-INF/classes/applicationContext.xml</tt><br />
	find <strong><bean id="sipDao" class="org.apache.openmeetings.db.dao.room.SipDao"></strong>
	uncomment its parameters and set it to your custom values.<br/>
	set value for <tt>uid</tt> property to unique secret value (can be generated here <a href="https://www.uuidtools.com">https://www.uuidtools.com</a>)
	and sync it with <tt>settings.properties</tt> of red5sip (see below)
	<p style="font-size: larger; color: blue;">
	IMPORTANT: this step should be done <strong>BEFORE</strong> system install/restore
	otherwise all SIP related room information will be lost
	</p>
	</div><br />
	<div>
	Restart asterisk:
	<source>
	<![CDATA[
	service asterisk restart
	]]>
	</source>
	</div><br />
	</section>

	<section name="Setup red5sip transport">
	<ul>
	<li>Download red5sip from <tt>https://github.com/openmeetings/red5sip</tt>
	<source>
	<![CDATA[
	git clone https://github.com/openmeetings/red5sip.git
	]]>
	</source>
	</li>
	<li>Build with Apache Maven
	<source>
	<![CDATA[
	cd red5sip
	mvn clean package
	]]>
	</source>
	</li>
	<li>All necessary files will be available in <tt>target</tt> folder, copy/move it to /opt/red5sip/</li>
	<li>Insert proper values to the <tt>/opt/red5sip/settings.properties</tt>
	<source>
	<![CDATA[
	red5.host=127.0.0.1 # red5 server address
	om.context=openmeetings # Openmeetings context
	red5.codec=asao
	red5.codec.rate=22 # should correlate with mic setting in Admin->Config `flash.mic.rate`
	sip.obproxy=127.0.0.1 # asterisk adderss
	sip.phone=red5sip_user # sip phone number
	sip.authid=red5sip_user # sip auth id
	sip.secret=12345 # sip password
	sip.realm=asterisk # sip realm
	sip.proxy=127.0.0.1 # address of sip proxy
	rooms.forceStart=no # TBD
	uid=87dddad4-9ca5-475b-860f-2e0825d02b76 #can be generated here: https://www.uuidtools.com/
	rooms=1 # TBD (not in use)
	]]>
	</source>
	</li>
	<li>Set correct permissions on red5sip files:
	<source>
	<![CDATA[
	sudo chown -R nobody:nogroup /opt/red5sip
	]]>
	</source>
	</li>
	<li>Add red5sip to autostart:
	<source>
	<![CDATA[
	sudo cp /opt/red5sip/red5sip /etc/init.d/
	sudo chmod a+x /etc/init.d/red5sip
	sudo update-rc.d red5sip defaults
	]]>
	</source>
	</li>
	<li>Start openmeetings
	<source>
	<![CDATA[
	service red5 start
	]]>
	</source>
	</li>
	<li>
	Enable <tt>SIP</tt> in openmeetings: <br/>
	<tt>Administration->Configuration->red5sip.enable == yes</tt>
	</li>
	<li>
	Enable SIP for particular room(s): <br/>
	<tt>Administration->Conference rooms->Room->Enable SIP transport in the room == checked</tt><br/>
	(SIP number will be assigned to room if everything is OK)
	</li>
	<li>Start red5sip
	<source>
	<![CDATA[
	service red5sip start
	]]>
	</source>
	</li>
	</ul>
	</section>
	</body>
	</document>