Browse Source
Jitsi Meet (and runit) types See merge request ungleich-public/cdist-contrib!21master
39 changed files with 2495 additions and 0 deletions
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|
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#!/bin/sh -e |
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EXPORTER_VERSION_FILE="/usr/local/bin/.prometheus-jitsi-meet-exporter.cdist.version" |
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if [ -f "${EXPORTER_VERSION_FILE}" ]; then |
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cat "${EXPORTER_VERSION_FILE}" |
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fi |
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#!/bin/sh -e |
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# This can be obtained with debconf-get-selections on a host with jitsi |
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# (and also analysing the deb-src) |
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if false; then |
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# We are currently not using these, just here as documentation |
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DEBCONF_SETTINGS="$(cat <<EOF |
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# Jicofo user password: |
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jicofo jicofo/jicofo-authpassword password STH |
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jitsi-meet-prosody jicofo/jicofo-authpassword password STH |
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# The secret used to connect to xmpp server as component |
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jitsi-meet-prosody jitsi-videobridge/jvbsecret password STH |
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jitsi-videobridge jitsi-videobridge/jvbsecret password STH |
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jitsi-videobridge2 jitsi-videobridge/jvbsecret password STH |
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# Jicofo Component secret: |
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jicofo jicofo/jicofosecret password STH |
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jitsi-meet-prosody jicofo/jicofosecret password STH |
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# Jicofo username: |
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jicofo jicofo/jicofo-authuser string focus |
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jitsi-meet-prosody jicofo/jicofo-authuser string focus |
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# The hostname of the current installation: |
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jitsi-meet-turnserver jitsi-meet-turnserver/jvb-hostname string ${JITSI_HOST} |
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# Full local server path to the SSL certificate file: |
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jitsi-meet-web-config jitsi-meet/cert-path-crt string |
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# Full local server path to the SSL key file: |
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jitsi-meet-web-config jitsi-meet/cert-path-key string |
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EOF |
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)" |
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fi |
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DEBCONF_SETTINGS="$(cat <<EOF |
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# The hostname of the current installation: |
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jitsi-meet-web-config jitsi-meet/jvb-hostname string ${JITSI_HOST} |
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# Hostname: |
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jicofo jitsi-videobridge/jvb-hostname string ${JITSI_HOST} |
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jitsi-meet-prosody jitsi-videobridge/jvb-hostname string ${JITSI_HOST} |
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jitsi-meet-turnserver jitsi-videobridge/jvb-hostname string ${TURN_SERVER} |
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jitsi-meet-web-config jitsi-videobridge/jvb-hostname string ${JITSI_HOST} |
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jitsi-videobridge jitsi-videobridge/jvb-hostname string ${JITSI_HOST} |
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jitsi-videobridge2 jitsi-videobridge/jvb-hostname string ${JITSI_HOST} |
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# The hostname of the current installation: |
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jitsi-meet-prosody jitsi-meet-prosody/jvb-hostname string ${JITSI_HOST} |
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# SSL certificate for the Jitsi Meet instance |
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# Choices: Generate a new self-signed certificate (You will later get a chance to obtain a Let's encrypt certificate), I want to use my own certificate |
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jitsi-meet-web-config jitsi-meet/cert-choice select Generate a new self-signed certificate (You will later get a chance to obtain a Let's encrypt certificate) |
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EOF |
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)" |
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if [ -n "${TURN_SECRET}" ]; then |
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DEBCONF_SETTINGS="$(cat <<EOF |
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${DEBCONF_SETTINGS} |
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# The turn server secret |
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jitsi-meet-prosody jitsi-meet-prosody/turn-secret string ${TURN_SECRET} |
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EOF |
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)" |
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fi |
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#!/bin/sh -e |
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# Setup firewall |
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__ufw |
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# Allow jitsi ports |
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require="__ufw" __ufw_rule http --rule 'allow 80/tcp' |
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require="__ufw" __ufw_rule https --rule 'allow 443/tcp' |
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require="__ufw" __ufw_rule jitsi_prom --rule 'allow 9888/tcp' |
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require="__ufw" __ufw_rule avfallback --rule 'allow 4443/tcp' |
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require="__ufw" __ufw_rule avdefault --rule 'allow 10000/udp' |
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#!/bin/sh -e |
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|
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if grep -qE "^__file/etc/nginx" "${__messages_in}"; then |
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echo "service nginx reload" |
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fi |
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cdist-type__jitsi_meet(7) |
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================================= |
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NAME |
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---- |
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cdist-type__jitsi_meet - Setup the server-side of Jitsi-Meet. |
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DESCRIPTION |
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----------- |
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This (singleton) type installs and configures jitsi-meet automatically. |
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It does so by following loosely the official quick-install instructions and |
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eXO's notes for installing and managing Jitsi Meet instances. |
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This type also sets up nginx in a way that is compatible with |
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`__letsencrypt_cert` and assumes that it will only serve Jitsi instances. |
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You will also need the `__jitsi_meet_domain` type in order to finish setting up |
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the web frontend (including TLS certificates) and its settings. |
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|
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You may want to use the `files/ufw` example manifest for a `__ufw`-based |
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firewall compatible with this type. |
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This file does not include rules for TCP port 9888, which exposes the |
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prometheus exporter if not disabled. |
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You should apply your own rules here. |
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This type only works on De{bi,vu}an systems. |
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NOTE: This type currently does not deal with setting up coturn. |
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For that, you might want to check `__coturn` in |
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https://code.ungleich.ch/ungleich-public/cdist-contrib |
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In that case, this type should run *after* `__coturn`. |
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OPTIONAL PARAMETERS |
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------------------- |
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turn-secret |
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The shared secret for the TURN server. |
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turn-server |
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The hostname of the TURN server. |
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This will assume that it is listening with TLS on port 443. |
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|
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BOOLEAN PARAMETERS |
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------------------ |
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disable-prometheus-exporter |
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This type enables a prometheus exporter for jitsi by default, if you would |
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rather not have that, pass this parameter. |
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The explorer is based on: |
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https://github.com/systemli/prometheus-jitsi-meet-exporter |
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EXAMPLES |
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-------- |
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.. code-block:: sh |
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# Setup the firewall |
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. "${__global}/type/__jitsi_meet/files/ufw" |
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export require="__ufw" |
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# Setup Jitsi on this host |
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__jitsi_meet \ |
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--turn-server "turn.exo.cat" \ |
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--turn-secret "WeNeedGoodSecurity" |
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SEE ALSO |
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-------- |
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- `__jitsi_meet_domain` |
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AUTHORS |
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------- |
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Evilham <contact@evilham.com> |
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COPYING |
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------- |
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Copyright \(C) 2020 Evilham. |
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@ -0,0 +1,197 @@
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#!/bin/sh -e |
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os="$(cat "${__global}/explorer/os")" |
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init="$(cat "${__global}/explorer/init")" |
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case "${os}" in |
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devuan|debian) |
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;; |
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*) |
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echo "Your OS '${os}' is currently not supported." > /dev/stderr |
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exit 1 |
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;; |
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esac |
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JITSI_HOST="${__target_host}" |
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TURN_SERVER="$(cat "${__object}/parameter/turn-server")" |
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TURN_SECRET="$(cat "${__object}/parameter/turn-secret")" |
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if [ -z "${TURN_SERVER}" ]; then |
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TURN_SERVER="${JITSI_HOST}" |
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fi |
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PROMETHEUS_JITSI_EXPORTER_IS_VERSION="$(cat "${__object}/explorer/prometheus-jitsi-meet-explorer-version")" |
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# The rest is loosely based on Jitsi's documentation |
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# https://jitsi.github.io/handbook/docs/devops-guide/devops-guide-quickstart |
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# Setup repositories |
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## First the signing keys |
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__package gnupg2 |
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require="__package/gnupg2" __apt_key_uri jitsi_meet \ |
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--name 'Jitsi <dev@jitsi.org>' \ |
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--uri https://download.jitsi.org/jitsi-key.gpg.key \ |
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--state present |
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## Now the repositories (they are a tad weird, so distribution is 'stable/') |
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require="__apt_key_uri/jitsi_meet" __apt_source jitsi_meet \ |
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--uri 'https://download.jitsi.org' \ |
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--distribution 'stable/' \ |
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--state present |
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## Ensure apt cache is up-to-date |
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require="__apt_source/jitsi_meet" __apt_update_index |
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export require="${require} __apt_source/jitsi_meet __apt_update_index" |
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# Pre-feed debconf settings, so Jitsi's installation has a good config |
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# shellcheck source=type/__jitsi_meet/files/debconf_settings.sh |
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. "${__type}/files/debconf_settings.sh" # This defines DEBCONF_SETTINGS |
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__debconf_set_selections jitsi_meet --file - <<EOF |
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${DEBCONF_SETTINGS} |
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EOF |
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export require="${require} __debconf_set_selections/jitsi_meet" |
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# Install and upgrade packages as needed |
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__package jitsi-meet |
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# Proceed only after installation/upgrade has finished |
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export require="__package/jitsi-meet" |
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# TODO: generalise and move out |
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# Prep nginx for acme settings |
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NGINX_ETC="/etc/nginx" |
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# |
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# Setup the acme-challenge snippet |
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# |
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__directory "${NGINX_ETC}/snippets" --state present |
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require="__directory${NGINX_ETC}/snippets" __file "${NGINX_ETC}/snippets/acme-challenge.conf" \ |
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--mode 644 \ |
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--source - << EOF |
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# This file is managed remotely, all changes will be lost |
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# This was heavily inspired by debops.org. |
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# Automatic Certificate Management Environment (ACME) support. |
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# https://tools.ietf.org/html/draft-ietf-acme-acme-01 |
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# https://en.wikipedia.org/wiki/Automated_Certificate_Management_Environment |
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# Return the ACME challenge present in the server public root. |
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# If not found, switch to global web server root. |
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location ^~ /.well-known/acme-challenge/ { |
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default_type "text/plain"; |
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try_files \$uri @well-known-acme-challenge; |
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} |
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# Return the ACME challenge present in the global server public root. |
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# If not present, redirect request to a specified domain. |
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location @well-known-acme-challenge { |
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root /usr/share/jitsi-meet; |
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default_type "text/plain"; |
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try_files \$uri @redirect-acme-challenge; |
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} |
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# Redirect the ACME challenge to a different host. If a redirect loop is |
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# detected, return 404. |
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location @redirect-acme-challenge { |
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if (\$arg_redirect) { |
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return 404; |
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} |
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return 307 \$scheme://${ACME_DOMAIN}\$request_uri?redirect=yes; |
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} |
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# Return 404 if ACME challenge well known path is accessed directly. |
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location = /.well-known/acme-challenge/ { |
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return 404; |
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} |
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EOF |
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|
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__directory "${NGINX_ETC}/sites-available" --state present |
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require="__directory${NGINX_ETC}/sites-available" __file "${NGINX_ETC}/sites-available/default" \ |
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--mode 644 \ |
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--source - << EOF |
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# This file is managed remotely, all changes will be lost |
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# nginx server configuration for: |
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server { |
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# Listen on IPv4 |
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listen 80; |
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# Note: there is an ipv6only=off flag, but it is Linux-only |
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# incidentally, that defaults to "on", which is what causes |
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# not having the double listen to listen on IPv6-only |
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listen [::]:80; |
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server_name welcome; |
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root /srv/www/sites/welcome/public; |
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include snippets/acme-challenge.conf; |
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location / { |
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return 301 https://\$host\$request_uri; |
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} |
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} |
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EOF |
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# These two should be changed on new release |
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PROMETHEUS_JITSI_EXPORTER_SHOULD_VERSION="1.1.2" |
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PROMETHEUS_JITSI_EXPORTER_CHECKSUM="sha256:c64852b1cf63efd1632ec07ac9fcead09baccc1bd8c7b21ca830a0d82d00fb3d" |
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PROMETHEUS_JITSI_EXPORTER_URL="https://github.com/systemli/prometheus-jitsi-meet-exporter/releases/download/${PROMETHEUS_JITSI_EXPORTER_VERSION}/prometheus-jitsi-meet-exporter-linux-amd64" |
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PROMETHEUS_JITSI_EXPORTER_VERSION_FILE="/usr/local/bin/.prometheus-jitsi-meet-exporter.cdist.version" |
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if [ ! -f "${__object}/parameter/disable-prometheus-exporter" ]; then |
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case "${init}" in |
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init|sysvinit) |
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__runit |
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require="__runit" __runit_service \ |
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prometheus-jitsi-meet-exporter --log --source - <<EOF |
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#!/bin/sh -e |
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cd /tmp |
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exec chpst -u "nobody:nogroup" env HOME="/tmp" \\ |
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prometheus-jitsi-meet-exporter \\ |
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-videobridge-url 'http://localhost:8888/stats' \\ |
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-web.listen-address ':9888' 2>&1 |
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EOF |
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export require="__runit_service/prometheus-jitsi-meet-exporter" |
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JITSI_MEET_EXPORTER_SERVICE="sv %s prometheus-jitsi-meet-exporter" |
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;; |
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systemd) |
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__systemd_unit prometheus-jitsi-meet-exporter.service \ |
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--source "-" \ |
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--enablement-state "enabled" <<EOF |
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[Unit] |
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Description=Metrics Exporter for Jitsi Meet |
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After=network.target |
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[Service] |
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Type=simple |
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DynamicUser=yes |
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ExecStart=/usr/local/bin/prometheus-jitsi-meet-exporter -videobridge-url 'http://localhost:8888/stats' -web.listen-address ':9888' |
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Restart=always |
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[Install] |
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WantedBy=multi-user.target |
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EOF |
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export require="__systemd_unit/prometheus-jitsi-meet-exporter.service" |
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JITSI_MEET_EXPORTER_SERVICE="service prometheus-jitsi-meet-exporter %s" |
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;; |
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esac |
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if [ "${PROMETHEUS_JITSI_EXPORTER_SHOULD_VERSION}" != \ |
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"${PROMETHEUS_JITSI_EXPORTER_IS_VERSION}" ]; then |
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# shellcheck disable=SC2059 |
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__download \ |
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/tmp/prometheus-jitsi-meet-exporter \ |
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--url "${PROMETHEUS_JITSI_EXPORTER_URL}" \ |
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--download remote \ |
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--sum "${PROMETHEUS_JITSI_EXPORTER_CHECKSUM}" \ |
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--onchange "$(printf "${JITSI_MEET_EXPORTER_SERVICE}" "stop") || true; chmod 555 /tmp/prometheus-jitsi-meet-exporter && mv /tmp/prometheus-jitsi-meet-exporter /usr/local/bin/prometheus-jitsi-meet-exporter && $(printf "${JITSI_MEET_EXPORTER_SERVICE}" "restart")" |
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printf "%s" "${PROMETHEUS_JITSI_EXPORTER_SHOULD_VERSION}" | \ |
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require="${require} __download/tmp/prometheus-jitsi-meet-exporter" __file \ |
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"${PROMETHEUS_JITSI_EXPORTER_VERSION_FILE}" \ |
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--source "-" |
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fi |
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fi |
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# TODO: disable the exporter if it is deployed and then admin changes their mind |
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@ -0,0 +1,700 @@
|
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#!/bin/sh -e |
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|
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# shellcheck disable=SC2034 # This is intended to be included |
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JITSI_CONFIG_JS="$(cat <<EOF |
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/* eslint-disable no-unused-vars, no-var */ |
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|
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var config = { |
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// Connection |
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// |
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|
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hosts: { |
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// XMPP domain. |
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domain: '${JITSI_HOST}', |
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|
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// When using authentication, domain for guest users. |
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// anonymousdomain: 'guest.example.com', |
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|
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// Domain for authenticated users. Defaults to <domain>. |
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// authdomain: '${JITSI_HOST}', |
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|
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// Call control component (Jigasi). |
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// call_control: 'callcontrol.${JITSI_HOST}', |
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|
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// Focus component domain. Defaults to focus.<domain>. |
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// focus: 'focus.${JITSI_HOST}', |
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|
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// XMPP MUC domain. FIXME: use XEP-0030 to discover it. |
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muc: 'conference.<!--# echo var="subdomain" default="" -->${JITSI_HOST}' |
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}, |
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|
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// BOSH URL. FIXME: use XEP-0156 to discover it. |
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bosh: '//${JITSI_HOST}/http-bind', |
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|
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// Websocket URL |
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// websocket: 'wss://${JITSI_HOST}/xmpp-websocket', |
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|
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// The name of client node advertised in XEP-0115 'c' stanza |
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clientNode: 'http://jitsi.org/jitsimeet', |
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|
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// The real JID of focus participant - can be overridden here |
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// Do not change username - FIXME: Make focus username configurable |
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// https://github.com/jitsi/jitsi-meet/issues/7376 |
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// focusUserJid: 'focus@auth.${JITSI_HOST}', |
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|
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|
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// Testing / experimental features. |
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// |
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|
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testing: { |
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// Disables the End to End Encryption feature. Useful for debugging |
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// issues related to insertable streams. |
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// disableE2EE: false, |
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|
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// P2P test mode disables automatic switching to P2P when there are 2 |
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// participants in the conference. |
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p2pTestMode: false |
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|
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// Enables the test specific features consumed by jitsi-meet-torture |
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// testMode: false |
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|
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// Disables the auto-play behavior of *all* newly created video element. |
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// This is useful when the client runs on a host with limited resources. |
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// noAutoPlayVideo: false |
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|
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// Enable / disable 500 Kbps bitrate cap on desktop tracks. When enabled, |
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// simulcast is turned off for the desktop share. If presenter is turned |
||||
// on while screensharing is in progress, the max bitrate is automatically |
||||
// adjusted to 2.5 Mbps. This takes a value between 0 and 1 which determines |
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// the probability for this to be enabled. |
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// capScreenshareBitrate: 1 // 0 to disable |
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|
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// Enable callstats only for a percentage of users. |
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// This takes a value between 0 and 100 which determines the probability for |
||||
// the callstats to be enabled. |
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// callStatsThreshold: 5 // enable callstats for 5% of the users. |
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}, |
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|
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// Disables ICE/UDP by filtering out local and remote UDP candidates in |
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// signalling. |
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// webrtcIceUdpDisable: false, |
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|
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// Disables ICE/TCP by filtering out local and remote TCP candidates in |
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// signalling. |
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// webrtcIceTcpDisable: false, |
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|
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|
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// Media |
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// |
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|
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// Audio |
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|
||||
// Disable measuring of audio levels. |
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disableAudioLevels: $(if [ -n "${DISABLE_AUDIO_LEVELS}" ]; then printf "true"; else printf "false"; fi), |
||||
// audioLevelsInterval: 200, |
||||
|
||||
// Enabling this will run the lib-jitsi-meet no audio detection module which |
||||
// will notify the user if the current selected microphone has no audio |
||||
// input and will suggest another valid device if one is present. |
||||
enableNoAudioDetection: true, |
||||
|
||||
// Enabling this will run the lib-jitsi-meet noise detection module which will |
||||
// notify the user if there is noise, other than voice, coming from the current |
||||
// selected microphone. The purpose it to let the user know that the input could |
||||
// be potentially unpleasant for other meeting participants. |
||||
enableNoisyMicDetection: true, |
||||
|
||||
// Start the conference in audio only mode (no video is being received nor |
||||
// sent). |
||||
// startAudioOnly: false, |
||||
|
||||
// Every participant after the Nth will start audio muted. |
||||
// startAudioMuted: 10, |
||||
|
||||
// Start calls with audio muted. Unlike the option above, this one is only |
||||
// applied locally. FIXME: having these 2 options is confusing. |
||||
// startWithAudioMuted: false, |
||||
|
||||
// Enabling it (with #params) will disable local audio output of remote |
||||
// participants and to enable it back a reload is needed. |
||||
// startSilent: false |
||||
|
||||
// Sets the preferred target bitrate for the Opus audio codec by setting its |
||||
// 'maxaveragebitrate' parameter. Currently not available in p2p mode. |
||||
// Valid values are in the range 6000 to 510000 |
||||
// opusMaxAverageBitrate: 20000, |
||||
|
||||
// Enables redundancy for Opus |
||||
// enableOpusRed: false |
||||
|
||||
// Video |
||||
|
||||
// Sets the preferred resolution (height) for local video. Defaults to 720. |
||||
// resolution: 720, |
||||
|
||||
// How many participants while in the tile view mode, before the receiving video quality is reduced from HD to SD. |
||||
// Use -1 to disable. |
||||
// maxFullResolutionParticipants: 2, |
||||
|
||||
// w3c spec-compliant video constraints to use for video capture. Currently |
||||
// used by browsers that return true from lib-jitsi-meet's |
||||
// util#browser#usesNewGumFlow. The constraints are independent from |
||||
// this config's resolution value. Defaults to requesting an ideal |
||||
// resolution of 720p. |
||||
// constraints: { |
||||
// video: { |
||||
// height: { |
||||
// ideal: 720, |
||||
// max: 720, |
||||
// min: 240 |
||||
// } |
||||
// } |
||||
// }, |
||||
$(if [ -n "${VIDEO_CONSTRAINTS}" ]; then echo "${VIDEO_CONSTRAINTS},"; fi) |
||||
|
||||
// Enable / disable simulcast support. |
||||
// disableSimulcast: false, |
||||
|
||||
// Enable / disable layer suspension. If enabled, endpoints whose HD |
||||
// layers are not in use will be suspended (no longer sent) until they |
||||
// are requested again. |
||||
// enableLayerSuspension: false, |
||||
|
||||
// Every participant after the Nth will start video muted. |
||||
startVideoMuted: ${START_VIDEO_MUTED}, |
||||
|
||||
// Start calls with video muted. Unlike the option above, this one is only |
||||
// applied locally. FIXME: having these 2 options is confusing. |
||||
// startWithVideoMuted: false, |
||||
|
||||
// If set to true, prefer to use the H.264 video codec (if supported). |
||||
// Note that it's not recommended to do this because simulcast is not |
||||
// supported when using H.264. For 1-to-1 calls this setting is enabled by |
||||
// default and can be toggled in the p2p section. |
||||
// This option has been deprecated, use preferredCodec under videoQuality section instead. |
||||
// preferH264: true, |
||||
|
||||
// If set to true, disable H.264 video codec by stripping it out of the |
||||
// SDP. |
||||
// disableH264: false, |
||||
|
||||
// Desktop sharing |
||||
|
||||
// Optional desktop sharing frame rate options. Default value: min:5, max:5. |
||||
// desktopSharingFrameRate: { |
||||
// min: 5, |
||||
// max: 5 |
||||
// }, |
||||
|
||||
// Try to start calls with screen-sharing instead of camera video. |
||||
// startScreenSharing: false, |
||||
|
||||
// Recording |
||||
|
||||
// Whether to enable file recording or not. |
||||
// fileRecordingsEnabled: false, |
||||
// Enable the dropbox integration. |
||||
// dropbox: { |
||||
// appKey: '<APP_KEY>' // Specify your app key here. |
||||
// // A URL to redirect the user to, after authenticating |
||||
// // by default uses: |
||||
// // 'https://${JITSI_HOST}/static/oauth.html' |
||||
// redirectURI: |
||||
// 'https://${JITSI_HOST}/subfolder/static/oauth.html' |
||||
// }, |
||||
// When integrations like dropbox are enabled only that will be shown, |
||||
// by enabling fileRecordingsServiceEnabled, we show both the integrations |
||||
// and the generic recording service (its configuration and storage type |
||||
// depends on jibri configuration) |
||||
// fileRecordingsServiceEnabled: false, |
||||
// Whether to show the possibility to share file recording with other people |
||||
// (e.g. meeting participants), based on the actual implementation |
||||
// on the backend. |
||||
// fileRecordingsServiceSharingEnabled: false, |
||||
|
||||
// Whether to enable live streaming or not. |
||||
// liveStreamingEnabled: false, |
||||
|
||||
// Transcription (in interface_config, |
||||
// subtitles and buttons can be configured) |
||||
// transcribingEnabled: false, |
||||
|
||||
// Enables automatic turning on captions when recording is started |
||||
// autoCaptionOnRecord: false, |
||||
|
||||
// Misc |
||||
|
||||
// Default value for the channel "last N" attribute. -1 for unlimited. |
||||
channelLastN: ${CHANNEL_LAST_N}, |
||||
|
||||
// Provides a way to use different "last N" values based on the number of participants in the conference. |
||||
// The keys in an Object represent number of participants and the values are "last N" to be used when number of |
||||
// participants gets to or above the number. |
||||
// |
||||
// For the given example mapping, "last N" will be set to 20 as long as there are at least 5, but less than |
||||
// 29 participants in the call and it will be lowered to 15 when the 30th participant joins. The 'channelLastN' |
||||
// will be used as default until the first threshold is reached. |
||||
// |
||||
// lastNLimits: { |
||||
// 5: 20, |
||||
// 30: 15, |
||||
// 50: 10, |
||||
// 70: 5, |
||||
// 90: 2 |
||||
// }, |
||||
|
||||
// Specify the settings for video quality optimizations on the client. |
||||
// videoQuality: { |
||||
// // Provides a way to prevent a video codec from being negotiated on the JVB connection. The codec specified |
||||
// // here will be removed from the list of codecs present in the SDP answer generated by the client. If the |
||||
// // same codec is specified for both the disabled and preferred option, the disable settings will prevail. |
||||
// // Note that 'VP8' cannot be disabled since it's a mandatory codec, the setting will be ignored in this case. |
||||
// disabledCodec: 'H264', |
||||
// |
||||
// // Provides a way to set a preferred video codec for the JVB connection. If 'H264' is specified here, |
||||
// // simulcast will be automatically disabled since JVB doesn't support H264 simulcast yet. This will only |
||||
// // rearrange the the preference order of the codecs in the SDP answer generated by the browser only if the |
||||
// // preferred codec specified here is present. Please ensure that the JVB offers the specified codec for this |
||||
// // to take effect. |
||||
// preferredCodec: 'VP8', |
||||
// |
||||
// // Provides a way to configure the maximum bitrates that will be enforced on the simulcast streams for |
||||
// // video tracks. The keys in the object represent the type of the stream (LD, SD or HD) and the values |
||||
// // are the max.bitrates to be set on that particular type of stream. The actual send may vary based on |
||||
// // the available bandwidth calculated by the browser, but it will be capped by the values specified here. |
||||
// // This is currently not implemented on app based clients on mobile. |
||||
// maxBitratesVideo: { |
||||
// low: 200000, |
||||
// standard: 500000, |
||||
// high: 1500000 |
||||
// }, |
||||
// |
||||
// // The options can be used to override default thresholds of video thumbnail heights corresponding to |
||||
// // the video quality levels used in the application. At the time of this writing the allowed levels are: |
||||
// // 'low' - for the low quality level (180p at the time of this writing) |
||||
// // 'standard' - for the medium quality level (360p) |
||||
// // 'high' - for the high quality level (720p) |
||||
// // The keys should be positive numbers which represent the minimal thumbnail height for the quality level. |
||||
// // |
||||
// // With the default config value below the application will use 'low' quality until the thumbnails are |
||||
// // at least 360 pixels tall. If the thumbnail height reaches 720 pixels then the application will switch to |
||||
// // the high quality. |
||||
// minHeightForQualityLvl: { |
||||
// 360: 'standard, |
||||
// 720: 'high' |
||||
// } |
||||
// }, |
||||
|
||||
// // Options for the recording limit notification. |
||||
// recordingLimit: { |
||||
// |
||||
// // The recording limit in minutes. Note: This number appears in the notification text |
||||
// // but doesn't enforce the actual recording time limit. This should be configured in |
||||
// // jibri! |
||||
// limit: 60, |
||||
// |
||||
// // The name of the app with unlimited recordings. |
||||
// appName: 'Unlimited recordings APP', |
||||
// |
||||
// // The URL of the app with unlimited recordings. |
||||
// appURL: 'https://unlimited.recordings.app.com/' |
||||
// }, |
||||
|
||||
// Disables or enables RTX (RFC 4588) (defaults to false). |
||||
// disableRtx: false, |
||||
|
||||
// Disables or enables TCC (the default is in Jicofo and set to true) |
||||
// (draft-holmer-rmcat-transport-wide-cc-extensions-01). This setting |
||||
// affects congestion control, it practically enables send-side bandwidth |
||||
// estimations. |
||||
// enableTcc: true, |
||||
|
||||
// Disables or enables REMB (the default is in Jicofo and set to false) |
||||
// (draft-alvestrand-rmcat-remb-03). This setting affects congestion |
||||
// control, it practically enables recv-side bandwidth estimations. When |
||||
// both TCC and REMB are enabled, TCC takes precedence. When both are |
||||
// disabled, then bandwidth estimations are disabled. |
||||
// enableRemb: false, |
||||
|
||||
// Enables ICE restart logic in LJM and displays the page reload overlay on |
||||
// ICE failure. Current disabled by default because it's causing issues with |
||||
// signaling when Octo is enabled. Also when we do an "ICE restart"(which is |
||||
// not a real ICE restart), the client maintains the TCC sequence number |
||||
// counter, but the bridge resets it. The bridge sends media packets with |
||||
// TCC sequence numbers starting from 0. |
||||
// enableIceRestart: false, |
||||
|
||||
// Defines the minimum number of participants to start a call (the default |
||||
// is set in Jicofo and set to 2). |
||||
// minParticipants: 2, |
||||
|
||||
// Use TURN/UDP servers for the jitsi-videobridge connection (by default |
||||
// we filter out TURN/UDP because it is usually not needed since the |
||||
// bridge itself is reachable via UDP) |
||||
// useTurnUdp: false |
||||
|
||||
// Enables / disables a data communication channel with the Videobridge. |
||||
// Values can be 'datachannel', 'websocket', true (treat it as |
||||
// 'datachannel'), undefined (treat it as 'datachannel') and false (don't |
||||
// open any channel). |
||||
// openBridgeChannel: true, |
||||
openBridgeChannel: 'websocket', |
||||
|
||||
|
||||
// UI |
||||
// |
||||
|
||||
// Hides lobby button |
||||
// hideLobbyButton: false, |
||||
|
||||
// Require users to always specify a display name. |
||||
// requireDisplayName: true, |
||||
|
||||
// Whether to use a welcome page or not. In case it's false a random room |
||||
// will be joined when no room is specified. |
||||
enableWelcomePage: true, |
||||
|
||||
// Enabling the close page will ignore the welcome page redirection when |
||||
// a call is hangup. |
||||
// enableClosePage: false, |
||||
|
||||
// Disable hiding of remote thumbnails when in a 1-on-1 conference call. |
||||
// disable1On1Mode: false, |
||||
|
||||
// Default language for the user interface. |
||||
defaultLanguage: '${DEFAULT_LANGUAGE}', |
||||
|
||||
// If true all users without a token will be considered guests and all users |
||||
// with token will be considered non-guests. Only guests will be allowed to |
||||
// edit their profile. |
||||
enableUserRolesBasedOnToken: false, |
||||
|
||||
// Whether or not some features are checked based on token. |
||||
// enableFeaturesBasedOnToken: false, |
||||
|
||||
// Enable lock room for all moderators, even when userRolesBasedOnToken is enabled and participants are guests. |
||||
// lockRoomGuestEnabled: false, |
||||
|
||||
// When enabled the password used for locking a room is restricted to up to the number of digits specified |
||||
// roomPasswordNumberOfDigits: 10, |
||||
// default: roomPasswordNumberOfDigits: false, |
||||
|
||||
// Message to show the users. Example: 'The service will be down for |
||||
// maintenance at 01:00 AM GMT, |
||||
noticeMessage: '${NOTICE_MESSAGE}', |
||||
|
||||
// Enables calendar integration, depends on googleApiApplicationClientID |
||||
// and microsoftApiApplicationClientID |
||||
// enableCalendarIntegration: false, |
||||
|
||||
// When 'true', it shows an intermediate page before joining, where the user can configure their devices. |
||||
// prejoinPageEnabled: false, |
||||
|
||||
// If true, shows the unsafe room name warning label when a room name is |
||||
// deemed unsafe (due to the simplicity in the name) and a password is not |
||||
// set or the lobby is not enabled. |
||||
// enableInsecureRoomNameWarning: false, |
||||
|
||||
// Whether to automatically copy invitation URL after creating a room. |
||||
// Document should be focused for this option to work |
||||
// enableAutomaticUrlCopy: false, |
||||
|
||||
// Stats |
||||
// |
||||
|
||||
// Whether to enable stats collection or not in the TraceablePeerConnection. |
||||
// This can be useful for debugging purposes (post-processing/analysis of |
||||
// the webrtc stats) as it is done in the jitsi-meet-torture bandwidth |
||||
// estimation tests. |
||||
// gatherStats: false, |
||||
|
||||
// The interval at which PeerConnection.getStats() is called. Defaults to 10000 |
||||
// pcStatsInterval: 10000, |
||||
|
||||
// To enable sending statistics to callstats.io you must provide the |
||||
// Application ID and Secret. |
||||
// callStatsID: '', |
||||
// callStatsSecret: '', |
||||
|
||||
// Enables sending participants' display names to callstats |
||||
// enableDisplayNameInStats: false, |
||||
|
||||
// Enables sending participants' emails (if available) to callstats and other analytics |
||||
// enableEmailInStats: false, |
||||
|
||||
// Privacy |
||||
// |
||||
|
||||
// If third party requests are disabled, no other server will be contacted. |
||||
// This means avatars will be locally generated and callstats integration |
||||
// will not function. |
||||
disableThirdPartyRequests: $(if [ -z "${ENABLE_THIRD_PARTY_REQUESTS}" ]; then printf "true"; else printf "false"; fi), |
||||
|
||||
|
||||
// Peer-To-Peer mode: used (if enabled) when there are just 2 participants. |
||||
// |
||||
|
||||
p2p: { |
||||
// Enables peer to peer mode. When enabled the system will try to |
||||
// establish a direct connection when there are exactly 2 participants |
||||
// in the room. If that succeeds the conference will stop sending data |
||||
// through the JVB and use the peer to peer connection instead. When a |
||||
// 3rd participant joins the conference will be moved back to the JVB |
||||
// connection. |
||||
enabled: true, |
||||
|
||||
// The STUN servers that will be used in the peer to peer connections |
||||
stunServers: [ |
||||
|
||||
{ urls: 'stun:${TURN_SERVER}:443' } |
||||
] |
||||
|
||||
// Sets the ICE transport policy for the p2p connection. At the time |
||||
// of this writing the list of possible values are 'all' and 'relay', |
||||
// but that is subject to change in the future. The enum is defined in |
||||
// the WebRTC standard: |
||||
// https://www.w3.org/TR/webrtc/#rtcicetransportpolicy-enum. |
||||
// If not set, the effective value is 'all'. |
||||
// iceTransportPolicy: 'all', |
||||
|
||||
// If set to true, it will prefer to use H.264 for P2P calls (if H.264 |
||||
// is supported). This setting is deprecated, use preferredCodec instead. |
||||
// preferH264: true |
||||
|
||||
// Provides a way to set the video codec preference on the p2p connection. Acceptable |
||||
// codec values are 'VP8', 'VP9' and 'H264'. |
||||
// preferredCodec: 'H264', |
||||
|
||||
// If set to true, disable H.264 video codec by stripping it out of the |
||||
// SDP. This setting is deprecated, use disabledCodec instead. |
||||
// disableH264: false, |
||||
|
||||
// Provides a way to prevent a video codec from being negotiated on the p2p connection. |
||||
// disabledCodec: '', |
||||
|
||||
// How long we're going to wait, before going back to P2P after the 3rd |
||||
// participant has left the conference (to filter out page reload). |
||||
// backToP2PDelay: 5 |
||||
}, |
||||
|
||||
analytics: { |
||||
// The Google Analytics Tracking ID: |
||||
// googleAnalyticsTrackingId: 'your-tracking-id-UA-123456-1' |
||||
|
||||
// Matomo configuration: |
||||
// matomoEndpoint: 'https://your-matomo-endpoint/', |
||||
// matomoSiteID: '42', |
||||
|
||||
// The Amplitude APP Key: |
||||
// amplitudeAPPKey: '<APP_KEY>' |
||||
|
||||
// Configuration for the rtcstats server: |
||||
// By enabling rtcstats server every time a conference is joined the rtcstats |
||||
// module connects to the provided rtcstatsEndpoint and sends statistics regarding |
||||
// PeerConnection states along with getStats metrics polled at the specified |
||||
// interval. |
||||
// rtcstatsEnabled: true, |
||||
|
||||
// In order to enable rtcstats one needs to provide a endpoint url. |
||||
// rtcstatsEndpoint: wss://rtcstats-server-pilot.jitsi.net/, |
||||
|
||||
// The interval at which rtcstats will poll getStats, defaults to 1000ms. |
||||
// If the value is set to 0 getStats won't be polled and the rtcstats client |
||||
// will only send data related to RTCPeerConnection events. |
||||
// rtcstatsPolIInterval: 1000 |
||||
|
||||
// Array of script URLs to load as lib-jitsi-meet "analytics handlers". |
||||
// scriptURLs: [ |
||||
// "libs/analytics-ga.min.js", // google-analytics |
||||
// "https://example.com/my-custom-analytics.js" |
||||
// ], |
||||
}, |
||||
|
||||
// Logs that should go be passed through the 'log' event if a handler is defined for it |
||||
// apiLogLevels: ['warn', 'log', 'error', 'info', 'debug'], |
||||
|
||||
// Information about the jitsi-meet instance we are connecting to, including |
||||
// the user region as seen by the server. |
||||
deploymentInfo: { |
||||
// shard: "shard1", |
||||
// region: "europe", |
||||
// userRegion: "asia" |
||||
}, |
||||
|
||||
// Decides whether the start/stop recording audio notifications should play on record. |
||||
// disableRecordAudioNotification: false, |
||||
|
||||
// Information for the chrome extension banner |
||||
// chromeExtensionBanner: { |
||||
// // The chrome extension to be installed address |
||||
// url: 'https://chrome.google.com/webstore/detail/jitsi-meetings/kglhbbefdnlheedjiejgomgmfplipfeb', |
||||
|
||||
// // Extensions info which allows checking if they are installed or not |
||||
// chromeExtensionsInfo: [ |
||||
// { |
||||
// id: 'kglhbbefdnlheedjiejgomgmfplipfeb', |
||||
// path: 'jitsi-logo-48x48.png' |
||||
// } |
||||
// ] |
||||
// }, |
||||
|
||||
// Local Recording |
||||
// |
||||
|
||||
// localRecording: { |
||||
// Enables local recording. |
||||
// Additionally, 'localrecording' (all lowercase) needs to be added to |
||||
// TOOLBAR_BUTTONS in interface_config.js for the Local Recording |
||||
// button to show up on the toolbar. |
||||
// |
||||
// enabled: true, |
||||
// |
||||
|
||||
// The recording format, can be one of 'ogg', 'flac' or 'wav'. |
||||
// format: 'flac' |
||||
// |
||||
|
||||
// }, |
||||
|
||||
// Options related to end-to-end (participant to participant) ping. |
||||
// e2eping: { |
||||
// // The interval in milliseconds at which pings will be sent. |
||||
// // Defaults to 10000, set to <= 0 to disable. |
||||
// pingInterval: 10000, |
||||
// |
||||
// // The interval in milliseconds at which analytics events |
||||
// // with the measured RTT will be sent. Defaults to 60000, set |
||||
// // to <= 0 to disable. |
||||
// analyticsInterval: 60000, |
||||
// }, |
||||
|
||||
// If set, will attempt to use the provided video input device label when |
||||
// triggering a screenshare, instead of proceeding through the normal flow |
||||
// for obtaining a desktop stream. |
||||
// NOTE: This option is experimental and is currently intended for internal |
||||
// use only. |
||||
// _desktopSharingSourceDevice: 'sample-id-or-label', |
||||
|
||||
// If true, any checks to handoff to another application will be prevented |
||||
// and instead the app will continue to display in the current browser. |
||||
// disableDeepLinking: false, |
||||
|
||||
// A property to disable the right click context menu for localVideo |
||||
// the menu has option to flip the locally seen video for local presentations |
||||
// disableLocalVideoFlip: false, |
||||
|
||||
// Mainly privacy related settings |
||||
|
||||
// Disables all invite functions from the app (share, invite, dial out...etc) |
||||
// disableInviteFunctions: true, |
||||
|
||||
// Disables storing the room name to the recents list |
||||
// doNotStoreRoom: true, |
||||
|
||||
// Deployment specific URLs. |
||||
// deploymentUrls: { |
||||
// // If specified a 'Help' button will be displayed in the overflow menu with a link to the specified URL for |
||||
// // user documentation. |
||||
// userDocumentationURL: 'https://docs.example.com/video-meetings.html', |
||||
// // If specified a 'Download our apps' button will be displayed in the overflow menu with a link |
||||
// // to the specified URL for an app download page. |
||||
// downloadAppsUrl: 'https://docs.example.com/our-apps.html' |
||||
// }, |
||||
|
||||
// Options related to the remote participant menu. |
||||
// remoteVideoMenu: { |
||||
// // If set to true the 'Kick out' button will be disabled. |
||||
// disableKick: true |
||||
// }, |
||||
|
||||
// If set to true all muting operations of remote participants will be disabled. |
||||
// disableRemoteMute: true, |
||||
|
||||
/** |
||||
External API url used to receive branding specific information. |
||||
If there is no url set or there are missing fields, the defaults are applied. |
||||
None of the fields are mandatory and the response must have the shape: |
||||
{ |
||||
// The hex value for the colour used as background |
||||
backgroundColor: '#fff', |
||||
// The url for the image used as background |
||||
backgroundImageUrl: 'https://example.com/background-img.png', |
||||
// The anchor url used when clicking the logo image |
||||
logoClickUrl: 'https://example-company.org', |
||||
// The url used for the image used as logo |
||||
logoImageUrl: 'https://example.com/logo-img.png' |
||||
} |
||||
*/ |
||||
brandingDataUrl: "$(if [ -n "${BRANDING_JSON}" ]; then printf "/branding.json"; fi)", |
||||
|
||||
// The URL of the moderated rooms microservice, if available. If it |
||||
// is present, a link to the service will be rendered on the welcome page, |
||||
// otherwise the app doesn't render it. |
||||
// moderatedRoomServiceUrl: 'https://moderated.${JITSI_HOST}', |
||||
|
||||
// List of undocumented settings used in jitsi-meet |
||||
/** |
||||
_immediateReloadThreshold |
||||
debug |
||||
debugAudioLevels |
||||
deploymentInfo |
||||
dialInConfCodeUrl |
||||
dialInNumbersUrl |
||||
dialOutAuthUrl |
||||
dialOutCodesUrl |
||||
disableRemoteControl |
||||
displayJids |
||||
etherpad_base |
||||
externalConnectUrl |
||||
firefox_fake_device |
||||
googleApiApplicationClientID |
||||
iAmRecorder |
||||
iAmSipGateway |
||||
microsoftApiApplicationClientID |
||||
peopleSearchQueryTypes |
||||
peopleSearchUrl |
||||
requireDisplayName |
||||
tokenAuthUrl |
||||
*/ |
||||
|
||||
/** |
||||
* This property can be used to alter the generated meeting invite links (in combination with a branding domain |
||||
* which is retrieved internally by jitsi meet) (e.g. https://meet.jit.si/someMeeting |
||||
* can become https://brandedDomain/roomAlias) |
||||
*/ |
||||
// brandingRoomAlias: null, |
||||
|
||||
// List of undocumented settings used in lib-jitsi-meet |
||||
/** |
||||
_peerConnStatusOutOfLastNTimeout |
||||
_peerConnStatusRtcMuteTimeout |
||||
abTesting |
||||
avgRtpStatsN |
||||
callStatsConfIDNamespace |
||||
callStatsCustomScriptUrl |
||||
desktopSharingSources |
||||
disableAEC |
||||
disableAGC |
||||
disableAP |
||||
disableHPF |
||||
disableNS |
||||
enableLipSync |
||||
enableTalkWhileMuted |
||||
forceJVB121Ratio |
||||
hiddenDomain |
||||
ignoreStartMuted |
||||
nick |
||||
startBitrate |
||||
*/ |
||||
|
||||
|
||||
// Allow all above example options to include a trailing comma and |
||||
// prevent fear when commenting out the last value. |
||||
makeJsonParserHappy: 'even if last key had a trailing comma' |
||||
|
||||
// no configuration value should follow this line. |
||||
}; |
||||
|
||||
/* eslint-enable no-unused-vars, no-var */ |
||||
EOF |
||||
)" |
||||
@ -0,0 +1,694 @@
|
||||
/* eslint-disable no-unused-vars, no-var */ |
||||
|
||||
var config = { |
||||
// Connection |
||||
// |
||||
|
||||
hosts: { |
||||
// XMPP domain. |
||||
domain: 'jitsi-meet.example.org', |
||||
|
||||
// When using authentication, domain for guest users. |
||||
// anonymousdomain: 'guest.example.com', |
||||
|
||||
// Domain for authenticated users. Defaults to <domain>. |
||||
// authdomain: 'jitsi-meet.example.org', |
||||
|
||||
// Call control component (Jigasi). |
||||
// call_control: 'callcontrol.jitsi-meet.example.org', |
||||
|
||||
// Focus component domain. Defaults to focus.<domain>. |
||||
// focus: 'focus.jitsi-meet.example.org', |
||||
|
||||
// XMPP MUC domain. FIXME: use XEP-0030 to discover it. |
||||
muc: 'conference.<!--# echo var="subdomain" default="" -->jitsi-meet.example.org' |
||||
}, |
||||
|
||||
// BOSH URL. FIXME: use XEP-0156 to discover it. |
||||
bosh: '//jitsi-meet.example.org/http-bind', |
||||
|
||||
// Websocket URL |
||||
// websocket: 'wss://jitsi-meet.example.org/xmpp-websocket', |
||||
|
||||
// The name of client node advertised in XEP-0115 'c' stanza |
||||
clientNode: 'http://jitsi.org/jitsimeet', |
||||
|
||||
// The real JID of focus participant - can be overridden here |
||||
// Do not change username - FIXME: Make focus username configurable |
||||
// https://github.com/jitsi/jitsi-meet/issues/7376 |
||||
// focusUserJid: 'focus@auth.jitsi-meet.example.org', |
||||
|
||||
|
||||
// Testing / experimental features. |
||||
// |
||||
|
||||
testing: { |
||||
// Disables the End to End Encryption feature. Useful for debugging |
||||
// issues related to insertable streams. |
||||
// disableE2EE: false, |
||||
|
||||
// P2P test mode disables automatic switching to P2P when there are 2 |
||||
// participants in the conference. |
||||
p2pTestMode: false |
||||
|
||||
// Enables the test specific features consumed by jitsi-meet-torture |
||||
// testMode: false |
||||
|
||||
// Disables the auto-play behavior of *all* newly created video element. |
||||
// This is useful when the client runs on a host with limited resources. |
||||
// noAutoPlayVideo: false |
||||
|
||||
// Enable / disable 500 Kbps bitrate cap on desktop tracks. When enabled, |
||||
// simulcast is turned off for the desktop share. If presenter is turned |
||||
// on while screensharing is in progress, the max bitrate is automatically |
||||
// adjusted to 2.5 Mbps. This takes a value between 0 and 1 which determines |
||||
// the probability for this to be enabled. |
||||
// capScreenshareBitrate: 1 // 0 to disable |
||||
|
||||
// Enable callstats only for a percentage of users. |
||||
// This takes a value between 0 and 100 which determines the probability for |
||||
// the callstats to be enabled. |
||||
// callStatsThreshold: 5 // enable callstats for 5% of the users. |
||||
}, |
||||
|
||||
// Disables ICE/UDP by filtering out local and remote UDP candidates in |
||||
// signalling. |
||||
// webrtcIceUdpDisable: false, |
||||
|
||||
// Disables ICE/TCP by filtering out local and remote TCP candidates in |
||||
// signalling. |
||||
// webrtcIceTcpDisable: false, |
||||
|
||||
|
||||
// Media |
||||
// |
||||
|
||||
// Audio |
||||
|
||||
// Disable measuring of audio levels. |
||||
// disableAudioLevels: false, |
||||
// audioLevelsInterval: 200, |
||||
|
||||
// Enabling this will run the lib-jitsi-meet no audio detection module which |
||||
// will notify the user if the current selected microphone has no audio |
||||
// input and will suggest another valid device if one is present. |
||||
enableNoAudioDetection: true, |
||||
|
||||
// Enabling this will run the lib-jitsi-meet noise detection module which will |
||||
// notify the user if there is noise, other than voice, coming from the current |
||||
// selected microphone. The purpose it to let the user know that the input could |
||||
// be potentially unpleasant for other meeting participants. |
||||
enableNoisyMicDetection: true, |
||||
|
||||
// Start the conference in audio only mode (no video is being received nor |
||||
// sent). |
||||
// startAudioOnly: false, |
||||
|
||||
// Every participant after the Nth will start audio muted. |
||||
// startAudioMuted: 10, |
||||
|
||||
// Start calls with audio muted. Unlike the option above, this one is only |
||||
// applied locally. FIXME: having these 2 options is confusing. |
||||
// startWithAudioMuted: false, |
||||
|
||||
// Enabling it (with #params) will disable local audio output of remote |
||||
// participants and to enable it back a reload is needed. |
||||
// startSilent: false |
||||
|
||||
// Sets the preferred target bitrate for the Opus audio codec by setting its |
||||
// 'maxaveragebitrate' parameter. Currently not available in p2p mode. |
||||
// Valid values are in the range 6000 to 510000 |
||||
// opusMaxAverageBitrate: 20000, |
||||
|
||||
// Enables redundancy for Opus |
||||
// enableOpusRed: false |
||||
|
||||
// Video |
||||
|
||||
// Sets the preferred resolution (height) for local video. Defaults to 720. |
||||
// resolution: 720, |
||||
|
||||
// How many participants while in the tile view mode, before the receiving video quality is reduced from HD to SD. |
||||
// Use -1 to disable. |
||||
// maxFullResolutionParticipants: 2, |
||||
|
||||
// w3c spec-compliant video constraints to use for video capture. Currently |
||||
// used by browsers that return true from lib-jitsi-meet's |
||||
// util#browser#usesNewGumFlow. The constraints are independent from |
||||
// this config's resolution value. Defaults to requesting an ideal |
||||
// resolution of 720p. |
||||
// constraints: { |
||||
// video: { |
||||
// height: { |
||||
// ideal: 720, |
||||
// max: 720, |
||||
// min: 240 |
||||
// } |
||||
// } |
||||
// }, |
||||
|
||||
// Enable / disable simulcast support. |
||||
// disableSimulcast: false, |
||||
|
||||
// Enable / disable layer suspension. If enabled, endpoints whose HD |
||||
// layers are not in use will be suspended (no longer sent) until they |
||||
// are requested again. |
||||
// enableLayerSuspension: false, |
||||
|
||||
// Every participant after the Nth will start video muted. |
||||
// startVideoMuted: 10, |
||||
|
||||
// Start calls with video muted. Unlike the option above, this one is only |
||||
// applied locally. FIXME: having these 2 options is confusing. |
||||
// startWithVideoMuted: false, |
||||
|
||||
// If set to true, prefer to use the H.264 video codec (if supported). |
||||
// Note that it's not recommended to do this because simulcast is not |
||||
// supported when using H.264. For 1-to-1 calls this setting is enabled by |
||||
// default and can be toggled in the p2p section. |
||||
// This option has been deprecated, use preferredCodec under videoQuality section instead. |
||||
// preferH264: true, |
||||
|
||||
// If set to true, disable H.264 video codec by stripping it out of the |
||||
// SDP. |
||||
// disableH264: false, |
||||
|
||||
// Desktop sharing |
||||
|
||||
// Optional desktop sharing frame rate options. Default value: min:5, max:5. |
||||
// desktopSharingFrameRate: { |
||||
// min: 5, |
||||
// max: 5 |
||||
// }, |
||||
|
||||
// Try to start calls with screen-sharing instead of camera video. |
||||
// startScreenSharing: false, |
||||
|
||||
// Recording |
||||
|
||||
// Whether to enable file recording or not. |
||||
// fileRecordingsEnabled: false, |
||||
// Enable the dropbox integration. |
||||
// dropbox: { |
||||
// appKey: '<APP_KEY>' // Specify your app key here. |
||||
// // A URL to redirect the user to, after authenticating |
||||
// // by default uses: |
||||
// // 'https://jitsi-meet.example.org/static/oauth.html' |
||||
// redirectURI: |
||||
// 'https://jitsi-meet.example.org/subfolder/static/oauth.html' |
||||
// }, |
||||
// When integrations like dropbox are enabled only that will be shown, |
||||
// by enabling fileRecordingsServiceEnabled, we show both the integrations |
||||
// and the generic recording service (its configuration and storage type |
||||
// depends on jibri configuration) |
||||
// fileRecordingsServiceEnabled: false, |
||||
// Whether to show the possibility to share file recording with other people |
||||
// (e.g. meeting participants), based on the actual implementation |
||||
// on the backend. |
||||
// fileRecordingsServiceSharingEnabled: false, |
||||
|
||||
// Whether to enable live streaming or not. |
||||
// liveStreamingEnabled: false, |
||||
|
||||
// Transcription (in interface_config, |
||||
// subtitles and buttons can be configured) |
||||
// transcribingEnabled: false, |
||||
|
||||
// Enables automatic turning on captions when recording is started |
||||
// autoCaptionOnRecord: false, |
||||
|
||||
// Misc |
||||
|
||||
// Default value for the channel "last N" attribute. -1 for unlimited. |
||||
channelLastN: -1, |
||||
|
||||
// Provides a way to use different "last N" values based on the number of participants in the conference. |
||||
// The keys in an Object represent number of participants and the values are "last N" to be used when number of |
||||
// participants gets to or above the number. |
||||
// |
||||
// For the given example mapping, "last N" will be set to 20 as long as there are at least 5, but less than |
||||
// 29 participants in the call and it will be lowered to 15 when the 30th participant joins. The 'channelLastN' |
||||
// will be used as default until the first threshold is reached. |
||||
// |
||||
// lastNLimits: { |
||||
// 5: 20, |
||||
// 30: 15, |
||||
// 50: 10, |
||||
// 70: 5, |
||||
// 90: 2 |
||||
// }, |
||||
|
||||
// Specify the settings for video quality optimizations on the client. |
||||
// videoQuality: { |
||||
// // Provides a way to prevent a video codec from being negotiated on the JVB connection. The codec specified |
||||
// // here will be removed from the list of codecs present in the SDP answer generated by the client. If the |
||||
// // same codec is specified for both the disabled and preferred option, the disable settings will prevail. |
||||
// // Note that 'VP8' cannot be disabled since it's a mandatory codec, the setting will be ignored in this case. |
||||
// disabledCodec: 'H264', |
||||
// |
||||
// // Provides a way to set a preferred video codec for the JVB connection. If 'H264' is specified here, |
||||
// // simulcast will be automatically disabled since JVB doesn't support H264 simulcast yet. This will only |
||||
// // rearrange the the preference order of the codecs in the SDP answer generated by the browser only if the |
||||
// // preferred codec specified here is present. Please ensure that the JVB offers the specified codec for this |
||||
// // to take effect. |
||||
// preferredCodec: 'VP8', |
||||
// |
||||
// // Provides a way to configure the maximum bitrates that will be enforced on the simulcast streams for |
||||
// // video tracks. The keys in the object represent the type of the stream (LD, SD or HD) and the values |
||||
// // are the max.bitrates to be set on that particular type of stream. The actual send may vary based on |
||||
// // the available bandwidth calculated by the browser, but it will be capped by the values specified here. |
||||
// // This is currently not implemented on app based clients on mobile. |
||||
// maxBitratesVideo: { |
||||
// low: 200000, |
||||
// standard: 500000, |
||||
// high: 1500000 |
||||
// }, |
||||
// |
||||
// // The options can be used to override default thresholds of video thumbnail heights corresponding to |
||||
// // the video quality levels used in the application. At the time of this writing the allowed levels are: |
||||
// // 'low' - for the low quality level (180p at the time of this writing) |
||||
// // 'standard' - for the medium quality level (360p) |
||||
// // 'high' - for the high quality level (720p) |
||||
// // The keys should be positive numbers which represent the minimal thumbnail height for the quality level. |
||||
// // |
||||
// // With the default config value below the application will use 'low' quality until the thumbnails are |
||||
// // at least 360 pixels tall. If the thumbnail height reaches 720 pixels then the application will switch to |
||||
// // the high quality. |
||||
// minHeightForQualityLvl: { |
||||
// 360: 'standard, |
||||
// 720: 'high' |
||||
// } |
||||
// }, |
||||
|
||||
// // Options for the recording limit notification. |
||||
// recordingLimit: { |
||||
// |
||||
// // The recording limit in minutes. Note: This number appears in the notification text |
||||
// // but doesn't enforce the actual recording time limit. This should be configured in |
||||
// // jibri! |
||||
// limit: 60, |
||||
// |
||||
// // The name of the app with unlimited recordings. |
||||
// appName: 'Unlimited recordings APP', |
||||
// |
||||
// // The URL of the app with unlimited recordings. |
||||
// appURL: 'https://unlimited.recordings.app.com/' |
||||
// }, |
||||
|
||||
// Disables or enables RTX (RFC 4588) (defaults to false). |
||||
// disableRtx: false, |
||||
|
||||
// Disables or enables TCC (the default is in Jicofo and set to true) |
||||
// (draft-holmer-rmcat-transport-wide-cc-extensions-01). This setting |
||||
// affects congestion control, it practically enables send-side bandwidth |
||||
// estimations. |
||||
// enableTcc: true, |
||||
|
||||
// Disables or enables REMB (the default is in Jicofo and set to false) |
||||
// (draft-alvestrand-rmcat-remb-03). This setting affects congestion |
||||
// control, it practically enables recv-side bandwidth estimations. When |
||||
// both TCC and REMB are enabled, TCC takes precedence. When both are |
||||
// disabled, then bandwidth estimations are disabled. |
||||
// enableRemb: false, |
||||
|
||||
// Enables ICE restart logic in LJM and displays the page reload overlay on |
||||
// ICE failure. Current disabled by default because it's causing issues with |
||||
// signaling when Octo is enabled. Also when we do an "ICE restart"(which is |
||||
// not a real ICE restart), the client maintains the TCC sequence number |
||||
// counter, but the bridge resets it. The bridge sends media packets with |
||||
// TCC sequence numbers starting from 0. |
||||
// enableIceRestart: false, |
||||
|
||||
// Defines the minimum number of participants to start a call (the default |
||||
// is set in Jicofo and set to 2). |
||||
// minParticipants: 2, |
||||
|
||||
// Use TURN/UDP servers for the jitsi-videobridge connection (by default |
||||
// we filter out TURN/UDP because it is usually not needed since the |
||||
// bridge itself is reachable via UDP) |
||||
// useTurnUdp: false |
||||
|
||||
// Enables / disables a data communication channel with the Videobridge. |
||||
// Values can be 'datachannel', 'websocket', true (treat it as |
||||
// 'datachannel'), undefined (treat it as 'datachannel') and false (don't |
||||
// open any channel). |
||||
// openBridgeChannel: true, |
||||
openBridgeChannel: 'websocket', |
||||
|
||||
|
||||
// UI |
||||
// |
||||
|
||||
// Hides lobby button |
||||
// hideLobbyButton: false, |
||||
|
||||
// Require users to always specify a display name. |
||||
// requireDisplayName: true, |
||||
|
||||
// Whether to use a welcome page or not. In case it's false a random room |
||||
// will be joined when no room is specified. |
||||
enableWelcomePage: true, |
||||
|
||||
// Enabling the close page will ignore the welcome page redirection when |
||||
// a call is hangup. |
||||
// enableClosePage: false, |
||||
|
||||
// Disable hiding of remote thumbnails when in a 1-on-1 conference call. |
||||
// disable1On1Mode: false, |
||||
|
||||
// Default language for the user interface. |
||||
// defaultLanguage: 'en', |
||||
|
||||
// If true all users without a token will be considered guests and all users |
||||
// with token will be considered non-guests. Only guests will be allowed to |
||||
// edit their profile. |
||||
enableUserRolesBasedOnToken: false, |
||||
|
||||
// Whether or not some features are checked based on token. |
||||
// enableFeaturesBasedOnToken: false, |
||||
|
||||
// Enable lock room for all moderators, even when userRolesBasedOnToken is enabled and participants are guests. |
||||
// lockRoomGuestEnabled: false, |
||||
|
||||
// When enabled the password used for locking a room is restricted to up to the number of digits specified |
||||
// roomPasswordNumberOfDigits: 10, |
||||
// default: roomPasswordNumberOfDigits: false, |
||||
|
||||
// Message to show the users. Example: 'The service will be down for |
||||
// maintenance at 01:00 AM GMT, |
||||
// noticeMessage: '', |
||||
|
||||
// Enables calendar integration, depends on googleApiApplicationClientID |
||||
// and microsoftApiApplicationClientID |
||||
// enableCalendarIntegration: false, |
||||
|
||||
// When 'true', it shows an intermediate page before joining, where the user can configure their devices. |
||||
// prejoinPageEnabled: false, |
||||
|
||||
// If true, shows the unsafe room name warning label when a room name is |
||||
// deemed unsafe (due to the simplicity in the name) and a password is not |
||||
// set or the lobby is not enabled. |
||||
// enableInsecureRoomNameWarning: false, |
||||
|
||||
// Whether to automatically copy invitation URL after creating a room. |
||||
// Document should be focused for this option to work |
||||
// enableAutomaticUrlCopy: false, |
||||
|
||||
// Stats |
||||
// |
||||
|
||||
// Whether to enable stats collection or not in the TraceablePeerConnection. |
||||
// This can be useful for debugging purposes (post-processing/analysis of |
||||
// the webrtc stats) as it is done in the jitsi-meet-torture bandwidth |
||||
// estimation tests. |
||||
// gatherStats: false, |
||||
|
||||
// The interval at which PeerConnection.getStats() is called. Defaults to 10000 |
||||
// pcStatsInterval: 10000, |
||||
|
||||
// To enable sending statistics to callstats.io you must provide the |
||||
// Application ID and Secret. |
||||
// callStatsID: '', |
||||
// callStatsSecret: '', |
||||
|
||||
// Enables sending participants' display names to callstats |
||||
// enableDisplayNameInStats: false, |
||||
|
||||
// Enables sending participants' emails (if available) to callstats and other analytics |
||||
// enableEmailInStats: false, |
||||
|
||||
// Privacy |
||||
// |
||||
|
||||
// If third party requests are disabled, no other server will be contacted. |
||||
// This means avatars will be locally generated and callstats integration |
||||
// will not function. |
||||
// disableThirdPartyRequests: false, |
||||
|
||||
|
||||
// Peer-To-Peer mode: used (if enabled) when there are just 2 participants. |
||||
// |
||||
|
||||
p2p: { |
||||
// Enables peer to peer mode. When enabled the system will try to |
||||
// establish a direct connection when there are exactly 2 participants |
||||
// in the room. If that succeeds the conference will stop sending data |
||||
// through the JVB and use the peer to peer connection instead. When a |
||||
// 3rd participant joins the conference will be moved back to the JVB |
||||
// connection. |
||||
enabled: true, |
||||
|
||||
// The STUN servers that will be used in the peer to peer connections |
||||
stunServers: [ |
||||
|
||||
// { urls: 'stun:jitsi-meet.example.org:3478' }, |
||||
{ urls: 'stun:meet-jit-si-turnrelay.jitsi.net:443' } |
||||
] |
||||
|
||||
// Sets the ICE transport policy for the p2p connection. At the time |
||||
// of this writing the list of possible values are 'all' and 'relay', |
||||
// but that is subject to change in the future. The enum is defined in |
||||
// the WebRTC standard: |
||||
// https://www.w3.org/TR/webrtc/#rtcicetransportpolicy-enum. |
||||
// If not set, the effective value is 'all'. |
||||
// iceTransportPolicy: 'all', |
||||
|
||||
// If set to true, it will prefer to use H.264 for P2P calls (if H.264 |
||||
// is supported). This setting is deprecated, use preferredCodec instead. |
||||
// preferH264: true |
||||
|
||||
// Provides a way to set the video codec preference on the p2p connection. Acceptable |
||||
// codec values are 'VP8', 'VP9' and 'H264'. |
||||
// preferredCodec: 'H264', |
||||
|
||||
// If set to true, disable H.264 video codec by stripping it out of the |
||||
// SDP. This setting is deprecated, use disabledCodec instead. |
||||
// disableH264: false, |
||||
|
||||
// Provides a way to prevent a video codec from being negotiated on the p2p connection. |
||||
// disabledCodec: '', |
||||
|
||||
// How long we're going to wait, before going back to P2P after the 3rd |
||||
// participant has left the conference (to filter out page reload). |
||||
// backToP2PDelay: 5 |
||||
}, |
||||
|
||||
analytics: { |
||||
// The Google Analytics Tracking ID: |
||||
// googleAnalyticsTrackingId: 'your-tracking-id-UA-123456-1' |
||||
|
||||
// Matomo configuration: |
||||
// matomoEndpoint: 'https://your-matomo-endpoint/', |
||||
// matomoSiteID: '42', |
||||
|
||||
// The Amplitude APP Key: |
||||
// amplitudeAPPKey: '<APP_KEY>' |
||||
|
||||
// Configuration for the rtcstats server: |
||||
// By enabling rtcstats server every time a conference is joined the rtcstats |
||||
// module connects to the provided rtcstatsEndpoint and sends statistics regarding |
||||
// PeerConnection states along with getStats metrics polled at the specified |
||||
// interval. |
||||
// rtcstatsEnabled: true, |
||||
|
||||
// In order to enable rtcstats one needs to provide a endpoint url. |
||||
// rtcstatsEndpoint: wss://rtcstats-server-pilot.jitsi.net/, |
||||
|
||||
// The interval at which rtcstats will poll getStats, defaults to 1000ms. |
||||
// If the value is set to 0 getStats won't be polled and the rtcstats client |
||||
// will only send data related to RTCPeerConnection events. |
||||
// rtcstatsPolIInterval: 1000 |
||||
|
||||
// Array of script URLs to load as lib-jitsi-meet "analytics handlers". |
||||
// scriptURLs: [ |
||||
// "libs/analytics-ga.min.js", // google-analytics |
||||
// "https://example.com/my-custom-analytics.js" |
||||
// ], |
||||
}, |
||||
|
||||
// Logs that should go be passed through the 'log' event if a handler is defined for it |
||||
// apiLogLevels: ['warn', 'log', 'error', 'info', 'debug'], |
||||
|
||||
// Information about the jitsi-meet instance we are connecting to, including |
||||
// the user region as seen by the server. |
||||
deploymentInfo: { |
||||
// shard: "shard1", |
||||
// region: "europe", |
||||
// userRegion: "asia" |
||||
}, |
||||
|
||||
// Decides whether the start/stop recording audio notifications should play on record. |
||||
// disableRecordAudioNotification: false, |
||||
|
||||
// Information for the chrome extension banner |
||||
// chromeExtensionBanner: { |
||||
// // The chrome extension to be installed address |
||||
// url: 'https://chrome.google.com/webstore/detail/jitsi-meetings/kglhbbefdnlheedjiejgomgmfplipfeb', |
||||
|
||||
// // Extensions info which allows checking if they are installed or not |
||||
// chromeExtensionsInfo: [ |
||||
// { |
||||
// id: 'kglhbbefdnlheedjiejgomgmfplipfeb', |
||||
// path: 'jitsi-logo-48x48.png' |
||||
// } |
||||
// ] |
||||
// }, |
||||
|
||||
// Local Recording |
||||
// |
||||
|
||||
// localRecording: { |
||||
// Enables local recording. |
||||
// Additionally, 'localrecording' (all lowercase) needs to be added to |
||||
// TOOLBAR_BUTTONS in interface_config.js for the Local Recording |
||||
// button to show up on the toolbar. |
||||
// |
||||