====== Install Jitsi on CentOS 7 ====== This page gives the needed steps to install and configure Jitsi on a CentOS server. If like me, you're not a big Docker fan, and you're happier with EL based systems, it might be useful to you. In this guide, you'll learn : * How to build latest versions of all components * Get a working install with prosody, jicofo, meet, videobridge * Configure jigasi and integrate it with an Asterisk/FreePBX server to have both inbound and outbound phone call from a Jitsi conference * Integrate Etherpad We deploy all this with ansible, see [[https://git.fws.fr/fws/ansible-roles/src/branch/master/roles/prosody|prosody]] [[https://git.fws.fr/fws/ansible-roles/src/branch/master/roles/jitsi|jitsi]] and [[https://git.fws.fr/fws/ansible-roles/src/branch/master/roles/jitsi_videobridge|jitsi videobridge]] This page are just some notes to help you setting this up if you don't want to play with ansible. But ansible is our recommanded way to deploy it Jitsi is composed of several components, and also relies on 3rd party ones. Here is a quick overview of which are using for what : * An XMPP server is needed to route messages between all the components. We will use prosody for this * Videobridge is the SFU((Selective Forwarding Unit)). It will receive all the video and audio streams, and handle relay to the other participants * Jicofo is the component which will handle stream negociation and room management * Meet is the user interface of Jitsi. It's also available as an mobile app and an Electron desktop app (but here we'll install the web interface only) * Jigasi is a SIP gateway. It can register on a SIP server and bridge phones and Jitsi rooms. Both outbound (call phone numbers from Jitsi) and inbound (join Jitsi rooms from a phone) are possible * ConferenceMapper API is a small daemon needed for inbound calls to work. It'll associate a PIN to each Jitsi room and is needed so that a phone call can be routed to the correct Jitsi room In this example, we will use visio.fws.fr as jitsi domain name. You'll need to adapt this This how to assume you already have a valid SSL cert in /etc/prosody/certs/jitsi.crt with its private key /etc/prosody/certs/jitsi.key. ===== Enable EPEL repo ===== If not already done yum install epel-release ===== Install prosody ===== Prosody is available in EPEL, so we can install it easily mkdir -p /opt/prosody/modules yum install prosody lua-ldap lua-cyrussasl Jitsi can also use some 3rd party prosody modules for MOD in ext_events.lib.lua \ util.lib.lua \ mod_speakerstats.lua \ mod_speakerstats_component.lua \ mod_turncredentials.lua \ mod_conference_duration.lua \ mod_conference_duration_component.lua; do wget -P /opt/prosody/modules \ https://raw.githubusercontent.com/jitsi/jitsi-meet/master/resources/prosody-plugins/$MOD done wget -P /opt/prosody/modules \ https://raw.githubusercontent.com/prosody-modules/mod_auth_ldap/master/mod_auth_ldap.lua Now, lets configure it. Edit /etc/prosody/prosody.cfg.lua plugin_paths = { "/opt/prosody/modules" } admins = { } modules_enabled = { "roster"; "saslauth"; "tls"; "dialback"; "disco"; "carbons"; "pep"; "private"; "blocklist"; "vcard4"; "vcard_legacy"; "version"; "uptime"; "time"; "ping"; "register"; "admin_adhoc"; "bosh"; "pubsub"; } modules_disabled = { } allow_registration = false c2s_require_encryption = true s2s_require_encryption = true s2s_secure_auth = false c2s_ports = { 5222, } s2s_port = { 5269, } http_port = { 5280, } component_ports = { 5347, } component_interface = "0.0.0.0" authentication = "internal_hashed" log = { info = "*syslog"; error = "*syslog"; } certificates = "/etc/pki/prosody/"; pidfile = "/run/prosody/prosody.pid"; daemonize = false; VirtualHost "localhost" Include "conf.d/*.cfg.lua" Now edit **/etc/prosody/conf.d/jitsi.cfg.lua** muc_mapper_domain_base = "visio.fws.fr"; admins = { "focus@auth.visio.fws.fr" } http_default_host = "visio.fws.fr" -- If you have a turn server, you can configure it here -- turncredentials_secret = "TURN_SECRET"; -- turncredentials = { -- { -- type = "turns", -- host = "turn.example.net", -- port = "3478", -- transport = "udp" -- } -- }; cross_domain_bosh = false; cross_domain_websocket = true; consider_bosh_secure = true; VirtualHost "visio.fws.fr" authentication = "anonymous" ssl = { key = "/etc/prosody/certs/jitsi.key"; certificate = "/etc/prosody/certs/jitsi.crt"; } modules_enabled = { "bosh"; "pubsub"; "ping"; "websocket"; "turncredentials"; "speakerstats"; "conference_duration"; } c2s_require_encryption = false allow_unencrypted_plain_auth = true speakerstats_component = "speakerstats.visio.fws.fr" conference_duration_component = "conferenceduration.visio.fws.fr" VirtualHost "auth.visio.fws.fr" ssl = { key = "/etc/prosody/certs/jitsi.key"; certificate = "/etc/prosody/certs/jitsi.crt"; } authentication = "internal_hashed" c2s_require_encryption = false Component "conference.visio.fws.fr" "muc" storage = "memory" modules_enabled = { "ping"; } muc_room_locking = false muc_room_default_public_jids = true Component "internal.auth.visio.fws.fr" "muc" storage = "memory" modules_enabled = { "ping"; } muc_room_cache_size = 1000 Component "focus.visio.fws.fr" component_secret = FOCUS_COMPONENT_SECRET" Component "speakerstats.visio.fws.fr" "speakerstats_component" muc_component = "conference.visio.fws.fr" Component "conferenceduration.visio.fws.fr" "conference_duration_component" muc_component = "conference.visio.fws.fr" Now we can start and enable the daemon systemctl enable --now prosody And we have to create some xmpp user accounts which will be used by Jitsi (adapt the passwords of course) prosodyctl register jvb auth.visio.fws.fr JVB_XMPP_PASS prosodyctl register focus auth.visio.fws.fr FOCUS_XMPP_PASS prosodyctl register jigasi auth.visio.fws.fr JIGASI_XMPP_PASS ===== Install a recent maven ===== Maven is available with yum, but its version is too old to build videobridge. So we'll install a newer one yum install java-1.8.0-openjdk java-1.8.0-openjdk-devel mkdir -p /opt/maven/apache-maven/ wget https://miroir.univ-lorraine.fr/apache/maven/maven-3/3.6.3/binaries/apache-maven-3.6.3-bin.tar.gz tar xvzf apache-maven-3.6.3-bin.tar.gz rsync -rvP --del apache-maven-3.6.3/ /opt/maven/apache-maven/ rm -rf apache-maven-3.6.3-bin.tar.gz apache-maven-3.6.3/ cat <<_EOF > /etc/profile.d/maven.sh #!/bin/sh export JAVA_HOME=/usr/lib/jvm/jre-openjdk export M2_HOME=/opt/maven/apache-maven export MAVEN_HOME=/opt/maven/apache-maven export PATH=${M2_HOME}/bin:${PATH} _EOF chmod +x /etc/profile.d/maven.sh exec bash ===== Create a jitsi user ===== useradd -d /opt/jitsi jitsi ===== Install Videobridge ===== yum install git mkdir /opt/jitsi/{src,videobridge} cd /opt/jitsi/src git clone https://github.com/jitsi/jitsi-videobridge.git cd jitsi-videobridge /opt/maven/apache-maven/bin/mvn package -DskipTests -Dassembly.skipAssembly=false unzip target/jitsi-videobridge-2.1-SNAPSHOT-archive.zip -d /tmp/ rsync -rvP --del /tmp/jitsi-videobridge-2.1-SNAPSHOT/ /opt/jitsi/videobridge/ rm -rf /tmp/jitsi-videobridge-2.1-SNAPSHOT/ Now we have to configure videobridge mkdir -p /opt/jitsi/etc/videobridge cat <<_EOF > /opt/jitsi/etc/videobridge/videobridge.conf JVB_OPTS="--apis=rest" JAVA_SYS_PROPS="-Dnet.java.sip.communicator.SC_HOME_DIR_LOCATION=/opt/jitsi/etc -Dnet.java.sip.communicator.SC_HOME_DIR_NAME=videobridge" _EOF cat <<_EOF > /opt/jitsi/etc/videobridge/sip-communicator.properties org.jitsi.impl.neomedia.transform.srtp.SRTPCryptoContext.checkReplay=false org.jitsi.videobridge.SINGLE_PORT_HARVESTER_PORT=10000 org.jitsi.videobridge.TCP_HARVESTER_PORT=4443 org.jitsi.videobridge.DISABLE_TCP_HARVESTER=false org.ice4j.ipv6.DISABLED=true # If behind NAT, set your private, and public IP here # org.ice4j.ice.harvest.NAT_HARVESTER_LOCAL_ADDRESS=10.99.2.19 # org.ice4j.ice.harvest.NAT_HARVESTER_PUBLIC_ADDRESS=10.11.12.13 org.jitsi.videobridge.ENABLE_STATISTICS=true org.jitsi.videobridge.STATISTICS_TRANSPORT=muc org.jitsi.videobridge.STATISTICS_INTERVAL=5000 org.jitsi.videobridge.xmpp.user.acc1.HOSTNAME=jitsi.fws.fr org.jitsi.videobridge.xmpp.user.acc1.DOMAIN=auth.visio.fws.fr org.jitsi.videobridge.xmpp.user.acc1.USERNAME=jvb org.jitsi.videobridge.xmpp.user.acc1.PASSWORD=JVB_PASSWORD org.jitsi.videobridge.xmpp.user.acc1.MUC_JIDS=JvbBrewery@internal.auth.visio.fws.fr # This is just a nickname for the videobridge. # If you run several videobridge instances, make sure each one uses a unique name org.jitsi.videobridge.xmpp.user.acc1.MUC_NICKNAME=jitsi.fws.fr _EOF Now we'll create a systemd unit for the videobridge service mkdir -p /etc/systemd/system cat <<_EOF > /etc/systemd/system/jitsi-videobridge.service [Unit] Description=Jitsi Videobridge After=network.target [Service] Type=simple SuccessExitStatus=143 EnvironmentFile=/opt/jitsi/etc/videobridge/videobridge.conf User=jitsi Group=jitsi PrivateTmp=true PrivateDevices=true ProtectHome=true ProtectSystem=full ReadOnlyDirectories=/opt/jitsi/etc /opt/jitsi/videobridge Restart=on-failure StartLimitInterval=0 RestartSec=30 # more threads for this process TasksMax=65000 # allow more open files for this process LimitNPROC=65000 LimitNOFILE=65000 ExecStart=/opt/jitsi/videobridge/jvb.sh ${JVB_OPTS} [Install] WantedBy=multi-user.target _EOF systemctl daemon-reload systemctl enable --now jitsi-videobridge You have to open ports TCP/4443 and UDP/10000. Those ports must be reachable by participants when they join a room. Unless you use a TURN server ===== Install Jicofo ===== cd /opt/jitsi/src git clone https://github.com/jitsi/jicofo.git cd jicofo /opt/maven/apache-maven/bin/mvn package -DskipTests -Dassembly.skipAssembly=false unzip target/jicofo-1.1-SNAPSHOT-archive.zip -d /tmp mkdir -p /opt/jitsi/jicofo rsync -rvP --del /tmp/jicofo-1.1-SNAPSHOT/ /opt/jitsi/jicofo/ Now that jicofo is installed, it must be configured mkdir -p /opt/jitsi/etc/jicofo cat <<_EOF > /opt/jitsi/etc/jicofo/jicofo.conf JICOFO_HOST=jitsi.fws.fr JICOFO_DOMAIN=visio.fws.fr JICOFO_USER=focus JICOFO_USERDOMAIN=auth.visio.fws.fr JICOFO_SECRET='FOCUS_COMPONENT_SECRET' JICOFO_USER_PASS='FOCUS_XMPP_PASS' JICOFO_OPTS='' JAVA_SYS_PROPS="-Dnet.java.sip.communicator.SC_HOME_DIR_LOCATION=/opt/jitsi/etc -Dnet.java.sip.communicator.SC_HOME_DIR_NAME=jicofo" _EOF cat <<_EOF > /opt/jitsi/etc/jicofo/sip-communicator.properties org.jitsi.jicofo.BRIDGE_MUC=JvbBrewery@internal.auth.visio.fws.fr # Comment this line if you do not intend to use Jigasi org.jitsi.jicofo.jigasi.BREWERY=JigasiBrewery@internal.auth.visio.fws.fr _EOF Now we can create a systemd unit and start jicofo cat <<_EOF > /etc/systemd/system/jitsi-jicofo.service [Unit] Description=Jitsi Conference Focus After=network.target [Service] Type=simple SuccessExitStatus=143 EnvironmentFile=/opt/jitsi/etc/jicofo/jicofo.conf User=jitsi Group=jitsi PrivateTmp=true PrivateDevices=true ProtectHome=true ProtectSystem=full ReadOnlyDirectories=/opt/jitsi/etc /opt/jitsi/jicofo Restart=on-failure StartLimitInterval=0 RestartSec=30 ExecStart=/opt/jitsi/jicofo/jicofo.sh \ --host=${JICOFO_HOST} \ --domain=${JICOFO_DOMAIN} \ --secret=${JICOFO_SECRET} \ --user_domain=${JICOFO_USERDOMAIN} \ --user_name=${JICOFO_USER} \ --user_password=${JICOFO_USER_PASS} \ ${JICOFO_OPT} [Install] WantedBy=multi-user.target _EOF systemctl daemon-reload systemctl enable --now jitsi-jicofo You should check your logs now to be sure jicofo discover your videobridge. ===== Install Meet ===== It's time to install the Meet interface now. cat <<_EOF > /etc/yum.repos.d/nodejs.repo [nodejs] baseurl = https://rpm.nodesource.com/pub_12.x/el/7/$basearch gpgcheck = 1 gpgkey = https://rpm.nodesource.com/pub/el/NODESOURCE-GPG-SIGNING-KEY-EL name = Node.js Packages for Enterprise Linux _EOF yum install nodejs cd /opt/jitsi/src/ git clone https://github.com/jitsi/jitsi-meet.git cd jitsi-meet npm i make This should build Jitsi Meet. We now can put it somewhere to be served by a web server : mkdir -p /opt/jitsi/meet rm -rf /opt/jitsi/meet/* mkdir -p /opt/jitsi/meet/css cp -r *.js *.html connection_optimization favicon.ico fonts images libs static sounds LICENSE lang /opt/jitsi/meet/ cp css/all.css /opt/jitsi/meet/css/ Serving this from your webserver is a bit out of scope for this how to because it can be done in a lot of different ways depending on your infra. Here's a sample nginx conf : server { listen 443 ssl http2; server_name visio.fws.fr; ssl_certificate_key /etc/prosody/certs/jitsi.key; ssl_certificate /etc/prosody/certs/jitsi.crt; if ($request_method !~ ^(GET|POST|HEAD)$ ) { return 405; } root /opt/jitsi/meet; index index.html; # conferenceMapper endpoint location ~ ^/(phoneNumberList|conferenceMapper) { proxy_pass http://localhost:8823; proxy_socket_keepalive on; } # BOSH endpoint location /http-bind { proxy_socket_keepalive on; proxy_pass http://localhost:5280/http-bind; proxy_set_header X-Forwarded-For $remote_addr; proxy_set_header Host $http_host; } # Websocket endpoint location /xmpp-websocket { proxy_pass http://localhost:5280/xmpp-websocket?$args; proxy_http_version 1.1; proxy_set_header Upgrade $http_upgrade; proxy_set_header Connection "upgrade"; proxy_set_header Host $http_host; proxy_set_header X-Forwarded-For $remote_addr; tcp_nodelay on; } # Conference rooms location ~ ^/([a-zA-Z0-9=\?]+)$ { rewrite ^/(.*)$ / break; } location / { ssi on; limit_req zone=limit_req_std burst=100 nodelay; limit_conn limit_conn_std 80; } allow 0.0.0.0/0; deny all; } You also have to edit /opt/jitsi/meet/config.js and adapt it to your needs. var config = { "bosh": "//visio.fws.fr/http-bind", "channelLastN": -1, "clientNode": "http://jitsi.org/jitsimeet", "desktopSharingChromeExtId": null, "desktopSharingChromeMinExtVersion": 0.1, "desktopSharingChromeSources": [ "screen", "window", "tab" ], "dialInConfCodeUrl": "https://visio.fws.fr/conferenceMapper", "dialInNumbersUrl": "https://visio.fws.fr/phoneNumberList", "disableAudioLevels": true, "disableThirdPartyRequests": true, "enableCalendarIntegration": false, "enableLayerSuspension": true, "enableNoAudioDetection": true, "enableNoisyMicDetection": false, "enableWelcomePage": true, // Uncomment to enable Etherpad integration //"etherpad_base": "https://etherpad.fws.fr/p/", "focusUserJid": "focus@auth.visio.fws.fr", "hosts": { "domain": "visio.fws.fr", "muc": "conference.visio.fws.fr" }, "localRecording": { "enabled": true }, "p2p": { "enabled": true, "preferH264": true, "useStunTurn": true }, "requireDisplayName": false, "resolution": 480, "testing": { "p2pTestMode": false }, "useStunTurn": true, "websocket": "wss://visio.fws.fr/xmpp-websocket" }; Now you should be able to reach the Meet interface and join a conference. ===== Install Jigasi ===== Now that we have a working Jitsi install, we may want to integrate it with our telephony system. With this, we'll be able to join phone numbers to jitsi conference. Jigasi is the component doing this bridge. It connects to your SIP server just as a phone, and also on prosody as an XMPP user. It can send connect calls to Jitsi Videobridge. For outbound calls, it's quite easy. Once configured and enabled, you'll get a small + in Meet interface from where you can type the number you want to call. The call will be made by Jigasi and routed by your SIP server. If the phone answer, it'll be imédiatly joined in the conf. Inbound calls are a bit trickier, and the general workflow is the following * Jitsi will associate a PIN for every conference, and register the mapping in the confmapper API server (see below) * You need a dedicated number for Jitsi (it can be done with an internal number only, but wouldn't have a lot of sense) * Calls to this number must be routed to a special IVR which will ask the caller to enter the PIN of the conference he wants to join * Asterisk will query the confmapper API to get the name of the conference matching the PIN * If a match is found, Asterisk adds the name of the conference to join in a SIP header, and route the call to jigasi * Jigasi answer the call, get the room from the SIP header, and join the corresponding room But first, lets install Jigasi mkdir -p /opt/jitsi/jigasi cd /opt/jitsi/src git clone https://github.com/jitsi/jigasi.git cd jigasi /opt/maven/apache-maven/bin/mvn package -DskipTests -Dassembly.skipAssembly=false unzip jigasi/target/jigasi-linux-x64-1.1-SNAPSHOT.zip -d /tmp rsync -rvP --del /tmp/jigasi-linux-x64-1.1-SNAPSHOT/ /opt/jitsi/jigasi/ Now, we have to configure it. In this example, the SIP server on which we register is ast.fws.fr and we'll use SIP extension 304 with secret SIP_SECRET (we'll see later in this how to how to craete the extension through FreePBHX web interface) mkdir -p /opt/jitsi/etc/jigasi cat <<_EOF > /opt/jitsi/jigasi/jigasi.conf JIGASI_OPTS='' JAVA_SYS_PROPS='' _EOF cat <<_EOF > /opt/jitsi/jigasi/sip-communicator.properties # Default room to which inbound called without a Jitsi-Conference-Room header org.jitsi.jigasi.DEFAULT_JVB_ROOM_NAME=sip net.java.sip.communicator.impl.protocol.SingleCallInProgressPolicy.enabled=false # Disable packet capture net.java.sip.communicator.packetlogging.PACKET_LOGGING_ENABLED=false # Enable brewery org.jitsi.jigasi.BREWERY_ENABLED=true org.jitsi.jigasi.MUC_SERVICE_ADDRESS=conference.visio.fws.fr # SIP acount net.java.sip.communicator.impl.protocol.sip.acc=acc net.java.sip.communicator.impl.protocol.sip.acc.ACCOUNT_UID=SIP\:304 # THis is the base64 encoded SIP secret. Obtained with # echo -n SIP_SECRET | base64 net.java.sip.communicator.impl.protocol.sip.acc.PASSWORD=U0lQX1NFQ1JFVA== net.java.sip.communicator.impl.protocol.sip.acc.PROTOCOL_NAME=SIP net.java.sip.communicator.impl.protocol.sip.acc.SERVER_ADDRESS=ast.fws.fr net.java.sip.communicator.impl.protocol.sip.acc.USER_ID=304 net.java.sip.communicator.impl.protocol.sip.acc.KEEP_ALIVE_INTERVAL=25 net.java.sip.communicator.impl.protocol.sip.acc.KEEP_ALIVE_METHOD=OPTIONS net.java.sip.communicator.impl.protocol.sip.acc.VOICEMAIL_ENABLED=false net.java.sip.communicator.impl.protocol.sip.acc.OVERRIDE_ENCODINGS=false net.java.sip.communicator.impl.protocol.sip.acc.DOMAIN_BASE=visio.fws.fr net.java.sip.communicator.impl.protocol.sip.acc.PROXY_ADDRESS=ast.fws.fr net.java.sip.communicator.impl.protocol.sip.acc.PROXY_AUTO_CONFIG=false net.java.sip.communicator.impl.protocol.sip.acc.PROXY_PORT=5060 net.java.sip.communicator.impl.protocol.sip.acc.PREFERRED_TRANSPORT=UDP # XMPP account net.java.sip.communicator.impl.protocol.jabber.acc=acc net.java.sip.communicator.impl.protocol.jabber.acc.ACCOUNT_UID=Jabber:jigasi@auth.visio.fws.fr net.java.sip.communicator.impl.protocol.jabber.acc.USER_ID=jigasi@auth.visio.fws.fr net.java.sip.communicator.impl.protocol.jabber.acc.IS_SERVER_OVERRIDDEN=true net.java.sip.communicator.impl.protocol.jabber.acc.SERVER_ADDRESS=jitsi.fws.fr # This is the base64 encoded XMPP secret # obtained with echo -n JIGASI_XMPP_PASS | base64 net.java.sip.communicator.impl.protocol.jabber.acc.PASSWORD=SklHQVNJX1hNUFBfUEFTUw== net.java.sip.communicator.impl.protocol.jabber.acc.RESOURCE_PRIORITY=30 net.java.sip.communicator.impl.protocol.jabber.acc.BREWERY=JigasiBrewery@internal.auth.visio.fws.fr net.java.sip.communicator.impl.protocol.jabber.acc.DOMAIN_BASE=visio.fws.fr org.jitsi.jigasi.xmpp.acc.USER_ID=jigasi@auth.visio.fws.fr org.jitsi.jigasi.xmpp.acc.PASS=JIGASI_XMPP_PASS org.jitsi.jigasi.xmpp.acc.ANONYMOUS_AUTH=false org.jitsi.jigasi.xmpp.acc.IS_SERVER_OVERRIDDEN=true org.jitsi.jigasi.xmpp.acc.SERVER_ADDRESS=jitsi.fws.fr org.jitsi.jigasi.xmpp.acc.JINGLE_NODES_ENABLED=false org.jitsi.jigasi.xmpp.acc.AUTO_DISCOVER_STUN=false org.jitsi.jigasi.xmpp.acc.IM_DISABLED=true org.jitsi.jigasi.xmpp.acc.SERVER_STORED_INFO_DISABLED=true org.jitsi.jigasi.xmpp.acc.IS_FILE_TRANSFER_DISABLED=true _EOF Now we can create a systemd unit and start the service cat <<_EOF > /etc/systemd/system/jitsi-jigasi.service [Unit] Description=Jitsi Gateway to SIP After=network.target [Service] Type=simple SuccessExitStatus=143 EnvironmentFile=/opt/jitsi/etc/jigasi/jigasi.conf User=jitsi Group=jitsi PrivateTmp=true PrivateDevices=true ProtectHome=true ProtectSystem=full Restart=on-failure StartLimitInterval=0 RestartSec=30 ExecStart=/opt/jitsi/jigasi/jigasi.sh \ --configdir=/opt/jitsi/etc \ --configdirname=jigasi \ --nocomponent=true \ ${JIGASI_OPT} [Install] WantedBy=multi-user.target _EOF systemctl daemon-reload systemctl enable --now jitsi-jigasi Jigasi is now running, but it's not yet ready to be used. ===== Install confmapper daemon ===== The confmapper daemon is a small tool to register Jitsi room name <-> PIN. We'll use https://github.com/gronke/jitsi-conferencemapper-api as it's a simple and lightweigt daemon in python, using an SQLite database to store the mappings yum install python3 mkdir -p /opt/jitsi/{data,confmapper} chown jitsi:jitsi /opt/jitsi/data chmod 700 /opt/jitsi/data wget -P /opt/jitsi/confmapper/ \ https://raw.githubusercontent.com/gronke/jitsi-conferencemapper-api/master/daemon.py chmod 755 /opt/jitsi/confmapper/daemon.py Now, lets configure it cat <<_EOF > /opt/jitsi/confmapper/config.json { "db_file": "/opt/jitsi/data/confmapper.sqlite", "expire_seconds": 86400, "host": "0.0.0.0", "id_max_length": 4, "numbers": { "FR": [ "0510101010" ] }, "port": 8823 } _EOF Here : * The daemon listens on 0.0.0.0:8823. You should configure a reverse proxy to make it accessible over HTTPS. The sample nginx configuration does it * Room mappings will be kept for 1 day * PIN will be created with 4 digits * The number 0510101010 is dedicated and will be announced in Jitsi interface. You can set several numbers, including different numbers for different countries In any case, make sure requests to https://visio.fws.fr/conferenceMapper and https://visio.fws.fr/phoneNumberList are routed to this daemon (because those are the URL configured in Jitsi meet Now, we can create a systemd unit and start the service cat <<_EOF > /etc/systemd/system/jitsi-confmapper.service [Unit] Description=Jitsi Conference Mapper After=network.target [Service] Type=simple User=jitsi Group=jitsi PrivateTmp=true PrivateDevices=true ProtectHome=true ProtectSystem=full Restart=on-failure StartLimitInterval=0 RestartSec=30 ExecStart=/opt/jitsi/confmapper/daemon.py [Install] WantedBy=multi-user.target _EOF systemctl daemon-reload systemctl enable --now jitsi-confmapper ===== Configure Asterisk/FreePBX ===== ==== Create a SIP extension ==== Now, we have to configure Asterisk. First step is to create an SIP extension for Jigasi. So we create a PJSIP extension, with ID **304** and secret **SIP_SECRET** (this is what we've configured in jigasi). In the advanced tab of the extension, there's a few things we can change * You might want to change the Outbound CID to advertize the 0510101010 number * You might change the context to **outbound-allroutes** if you don't want conference user to be able to call internal numbers. The default context is **from-internal** * Restrict codecs to alaw and ulaw (I had sound issues with opus and g722, so better to restrict this to known working codecs). For this, type **all** in the **Disallowed Codecs** field, and **alaw&ulaw** in the **Allowed Codec** field * You should also disable the **Direct Media** option ==== Create a custom IVR ==== Now, we have to create a custom IVR which will ask callers the PIN of the room they want to join. you can put it in **/etc/asterisk/extension_custom.conf** [jitsi-ivr] exten => s,1,Answer exten => s,n,Set(IVR_MSG=conf-getpin) exten => s,n,Set(TIMEOUT(digit)=3) exten => s,n,Read(JITSI_PIN,${IVR_MSG}) ; Fetch the conf name from the PIN entered exten => s,n,AGI(jitsi_conf_pin,"https://visio.fws.fr/conferenceMapper",${JITSI_PIN} ; If we got a result, dial JIGASI SIP account, else, loop and ask again exten => s,n,GotoIf($["${JITSI_ROOM}" != "error"]?jitsi,1) exten => s,n(error),Playback(conf-invalid) exten => s,n,Goto(s,1) ; We got a result, lets join jitsi room exten => jitsi,1,Verbose(PIN ${JITSI_PIN} maps to Jitsi room ${JITSI_ROOM}) exten => jitsi,n,Dial(PJSIP/304,,b(jitsi-conference-room-header^addheader^1(${JITSI_ROOM}))) ==== Create an AGI script to lookup roomname from their PIN ==== We have to create an AGI script so that asterisk can query the confmapper daemon to get the name of a room from the PIN. For this, create the script **/usr/share/asterisk/agi-bin/jitsi_conf_pin** with the following content : #!/usr/bin/perl use warnings; use strict; use LWP::UserAgent; use JSON; my $ret = 'error'; my $url = $ARGV[0] . '?id=' . $ARGV[1]; my $ua = LWP::UserAgent->new(timeout => 10); $ua->env_proxy; my $response = $ua->get($url); if ($response->is_success){ my $json = from_json($response->content); if (defined $json and defined $json->{conference}){ $ret = $json->{conference}; $ret =~ s/@.*//; } } print "SET VARIABLE JITSI_ROOM $ret\n"; The script must be executable chmod +x /usr/share/asterisk/agi-bin/jitsi_conf_pin ==== Create a Custom Destination pointing on your custom IVR ==== OK, now we need to way to route calls to our new custom IVR. For this, we'll create a **Custom Destination** in FreePBX. Just set the target to **jitsi-ivr,s,1** {{ :tuto:linux_divers:jitsi_custom_dest.png |}} This **Custom Destination** make the IVR available in all the FreePBX routing logic. ==== Assign an internal number to the IVR to test ==== We can assign it an internal number to test it with a new **Misc Application** : {{ :tuto:linux_divers:jitsi_misc_app.png |}} You can now try it. Create a new room in Jitsi, and if you click on the small **i** button (bottom right), you should see a popup with the number to dial and the PIN {{ :tuto:linux_divers:jitsi_conf_ping.png |}} So, this conf has PIN 4845 Now, call your internal test number, which points on the custom IVR (381 in the previous screenshot). You should be prompted to enter a PIN. Once typed, asterisk will lookup on the confmapper daemon to find to which room this PIN maps. If found, you'll join the conference right away. If a wrong PI is entered, you'll be prompted again to enter the PIN. Now, all you have to do is to define a new **Inbound Route** which points on the same **Custom Destination** {{ :tuto:linux_divers:jitsi_inbound_route.png |}} ===== Integrate with Etherpad ===== Deploying an Etherpad instance is out of scope for this guide (but we also have an [[https://git.fws.fr/fws/ansible-roles/src/branch/master/roles/etherpad|ansible role]] for this). But, say you have it available at https://etherpad.fws.fr. All you have to do is to indicate it in **/opt/jitsi/meet/config.js** [...] "etherpad_base": "https://etherpad.fws.fr/p/", [...] The trailing **/** is important ===== Note on reverse proxy and Content-Security-Policy ===== We use a reverse proxy to serve all the web resources, and this reverse proxy insert CSP headers to response. In this case, we have to allow a few things to get everything working : * In img-src you have to add https://img.youtube.com and https://i.ytimg.com * In script-src you have to add https://www.youtube.com and https://s.ytimg.com * In frame-src you have to add https://etherpad.fws.fr wss://etherpad.fws.fr and https://www.youtube.com * In connect-src you have to add https://storage.googleapis.com