openmeetings-server/src/site/xdoc/red5sip-integration_3.0.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>SIP-Transport Integration</title>
 		<author email="dev@openmeetings.apache.org">Apache OpenMeetings Team</author>
 	</properties>
 	<body>
 		<section name="SIP-Transport Integration">
 			<p>You need Apache OpenMeetings <strong>version 3.0+</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 12.10.</p>
 		</section>
 		<section name="Prerequisites">
 			<div>
 				Run the commands
 				<source>
 <![CDATA[
 sudo apt-get update && sudo apt-get upgrade
 ]]>
 				</source>
 			</div>
 		</section>
 		<section name="Building and setting up Asterisk">
 			<div>
 				Run the commands
 				<source>
 <![CDATA[
 sudo mkdir /usr/src/asterisk &amp;&amp; cd /usr/src/asterisk
 sudo wget http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-13.9.0.tar.gz
 sudo tar -xvzf asterisk-13.9.0.tar.gz
 cd ./asterisk-13.9.0
 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/red5/webapps/openmeetings/WEB-INF/classes/openmeetings-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 creadentials for Asterisk manager.
 				Modify <tt>/opt/red5/webapps/openmeetings/WEB-INF/classes/openmeetings-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.
 				<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">
 			<div>
 				Download red5sip from <tt>https://github.com/solomax/red5phone/tree/red5sip_3.0</tt>, switch to the branch red5sip_3.0
 			</div>
 			<div>
 				switch to the branch red5sip_3.0
 				<source>
 <![CDATA[
 git clone https://github.com/solomax/red5phone.git
 git checkout red5sip_3.0
 ]]>
 				</source>
 			</div>
 			<div>
 				Build with Apache Ant
 				<source>
 <![CDATA[
 ant
 ]]>
 				</source>
 			</div>
 			<div>
 				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 settings in public/config.xml
 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
 rooms=1 # TBD (not in use)
 ]]>
 				</source>
 			</div>
 			<div>
 				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>
 			</div>
 			<div>
 				Start openmeetings
 				<source>
 <![CDATA[
 service red5 start
 ]]>
 				</source>
 			</div>
 			<div>
 				Start red5sip
 				<source>
 <![CDATA[
 service red5sip start
 ]]>
 				</source>
 			</div>
 		</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>SIP-Transport Integration</title>
	<author email="dev@openmeetings.apache.org">Apache OpenMeetings Team</author>
	</properties>
	<body>
	<section name="SIP-Transport Integration">
	<p>You need Apache OpenMeetings <strong>version 3.0+</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 12.10.</p>
	</section>
	<section name="Prerequisites">
	<div>
	Run the commands
	<source>
	<![CDATA[
	sudo apt-get update && sudo apt-get 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-13.9.0.tar.gz
	sudo tar -xvzf asterisk-13.9.0.tar.gz
	cd ./asterisk-13.9.0
	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/red5/webapps/openmeetings/WEB-INF/classes/openmeetings-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 creadentials for Asterisk manager.
	Modify <tt>/opt/red5/webapps/openmeetings/WEB-INF/classes/openmeetings-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.
	<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">
	<div>
	Download red5sip from <tt>https://github.com/solomax/red5phone/tree/red5sip_3.0</tt>, switch to the branch red5sip_3.0
	</div>
	<div>
	switch to the branch red5sip_3.0
	<source>
	<![CDATA[
	git clone https://github.com/solomax/red5phone.git
	git checkout red5sip_3.0
	]]>
	</source>
	</div>
	<div>
	Build with Apache Ant
	<source>
	<![CDATA[
	ant
	]]>
	</source>
	</div>
	<div>
	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 settings in public/config.xml
	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
	rooms=1 # TBD (not in use)
	]]>
	</source>
	</div>
	<div>
	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>
	</div>
	<div>
	Start openmeetings
	<source>
	<![CDATA[
	service red5 start
	]]>
	</source>
	</div>
	<div>
	Start red5sip
	<source>
	<![CDATA[
	service red5sip start
	]]>
	</source>
	</div>
	</section>
	</body>
	</document>