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plugin.json
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plugin.json
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{
"id": "com.mattermost.calls",
"name": "Calls",
"description": "Integrates real-time voice communication in Mattermost",
"homepage_url": "https://github.com/mattermost/mattermost-plugin-calls/",
"support_url": "https://github.com/mattermost/mattermost-plugin-calls/issues",
"min_server_version": "9.5.0",
"server": {
"executables": {
"linux-amd64": "server/dist/plugin-linux-amd64",
"linux-arm64": "server/dist/plugin-linux-arm64",
"freebsd-amd64": "server/dist/plugin-freebsd-amd64",
"openbsd-amd64": "server/dist/plugin-openbsd-amd64"
}
},
"webapp": {
"bundle_path": "webapp/dist/main.js"
},
"settings_schema": {
"header": "Calls plugin enables voice calls with screensharing in channels. See [documentation](https://docs.mattermost.com/channels/make-calls.html) to learn more.",
"footer": "",
"settings": [
{
"key": "DefaultEnabled",
"display_name": "Test mode",
"type": "custom",
"help_text": "When test mode is enabled, only system admins are able to start calls in channels. This allows testing to confirm calls are working as expected.",
"hosting": "on-prem"
},
{
"key": "UDPServerAddress",
"display_name": "RTC Server Address (UDP)",
"type": "text",
"help_text": "The local IP address used by the RTC server to listen on for UDP connections.",
"default": "",
"placeholder": "127.0.0.1",
"hosting": "on-prem"
},
{
"key": "TCPServerAddress",
"display_name": "RTC Server Address (TCP)",
"type": "text",
"help_text": "The local IP address used by the RTC server to listen on for TCP connections.",
"default": "",
"placeholder": "127.0.0.1",
"hosting": "on-prem"
},
{
"key": "UDPServerPort",
"display_name": "RTC Server Port (UDP)",
"type": "number",
"help_text": "The UDP port the RTC server will listen on.",
"default": 8443,
"placeholder": "8443",
"hosting": "on-prem"
},
{
"key": "TCPServerPort",
"display_name": "RTC Server Port (TCP)",
"type": "number",
"help_text": "The TCP port the RTC server will listen on.",
"default": 8443,
"placeholder": "8443",
"hosting": "on-prem"
},
{
"key": "ICEHostOverride",
"display_name": "ICE Host Override",
"type": "text",
"help_text": "(Optional) The IP (or hostname) to be used as the host ICE candidate. If empty, it defaults to resolving via STUN.",
"default": "",
"hosting": "on-prem"
},
{
"key": "ICEHostPortOverride",
"display_name": "ICE Host Port Override",
"type": "number",
"help_text": "(Optional) A port number to be used as an override for host candidates in place of the one used to listen on.\nNote: this port will apply to both UDP and TCP host candidates",
"hosting": "on-prem"
},
{
"key": "RTCDServiceURL",
"display_name": "RTCD service URL",
"type": "text",
"help_text": "(Optional) The URL to a running RTCD service instance that should host the calls. When set (non empty) all calls will be handled by the external service.",
"placeholder": "https://rtcd.example.com",
"hosting": "on-prem"
},
{
"key": "MaxCallParticipants",
"display_name": "Max call participants",
"type": "number",
"help_text": "The maximum number of participants that can join a call. If left empty, or set to 0, it means unlimited.",
"default": 0,
"hosting": "on-prem"
},
{
"key": "ICEServersConfigs",
"display_name": "ICE Servers Configurations",
"type": "longtext",
"help_text": "(Optional) A list of ICE servers (STUN/TURN) configurations to use. This field should contain a valid JSON array.",
"default": "[{\"urls\":[\"stun:stun.global.calls.mattermost.com:3478\"]}]",
"placeholder": "[{\n \"urls\":[\"turn:turnserver.example.org:3478\"],\n \"username\": \"webrtc\",\n \"credential\": \"turnpassword\"\n}]",
"hosting": "on-prem"
},
{
"key": "TURNStaticAuthSecret",
"display_name": "TURN Static Auth Secret",
"type": "text",
"default": "",
"help_text": "(Optional) The secret key used to generate TURN short-lived authentication credentials.",
"hosting": "on-prem"
},
{
"key": "TURNCredentialsExpirationMinutes",
"display_name": "TURN Credentials Expiration (minutes)",
"type": "number",
"default": 1440,
"help_text": "(Optional) The number of minutes that the generated TURN credentials will be valid for.",
"hosting": "on-prem"
},
{
"key": "ServerSideTURN",
"display_name": "Server Side TURN",
"type": "bool",
"default": false,
"help_text": "(Optional) When set to on it will pass and use configured TURN candidates to server initiated connections.",
"hosting": "on-prem"
},
{
"key": "AllowScreenSharing",
"display_name": "Allow screen sharing",
"type": "bool",
"default": true,
"help_text": "When set to true it allows call participants to share their screen."
},
{
"key": "EnableSimulcast",
"display_name": "Enable simulcast for screen sharing (Experimental)",
"type": "bool",
"default": false,
"help_text": "When set to true it enables simulcast for screen sharing. This can help to improve screen sharing quality."
},
{
"key": "EnableRecordings",
"display_name": "Enable call recordings (Beta)",
"type": "bool",
"default": false,
"help_text": "(Optional) When set to true, call recordings are enabled."
},
{
"key": "JobServiceURL",
"display_name": "Job service URL",
"type": "text",
"help_text": "The URL to a running calls job service instance used for call recordings.",
"placeholder": "https://calls-job-service.example.com"
},
{
"key": "MaxRecordingDuration",
"display_name": "Maximum call recording duration",
"type": "number",
"default": 60,
"help_text": "The maximum duration (in minutes) for call recordings. Value must be in the range [15, 180]."
},
{
"key": "RecordingQuality",
"display_name": "Call recording quality",
"type": "dropdown",
"default": "medium",
"help_text": "The audio and video quality of call recordings.\n Note: this setting can affect the overall performance of the job service and the number of concurrent recording jobs that can be run.",
"options": [
{
"display_name": "Low",
"value": "low"
},
{
"display_name": "Medium",
"value": "medium"
},
{
"display_name": "High",
"value": "high"
}
],
"hosting": "on-prem"
},
{
"key": "EnableTranscriptions",
"display_name": "Enable call transcriptions (Experimental)",
"type": "bool",
"default": false,
"help_text": "(Optional) When set to true, post-call transcriptions are enabled."
},
{
"key": "TranscriberModelSize",
"display_name": "Call transcriber model size",
"type": "dropdown",
"default": "base",
"help_text": "The speech-to-text model size to use for post-call transcriptions. Heavier models will produce more accurate results at the expense of processing time and resources usage.",
"options": [
{
"display_name": "Tiny",
"value": "tiny"
},
{
"display_name": "Base",
"value": "base"
},
{
"display_name": "Small",
"value": "small"
}
],
"hosting": "on-prem"
},
{
"key": "TranscriberNumThreads",
"display_name": "Call transcriber threads",
"type": "number",
"default": 2,
"help_text": "The number of threads used by the post-call transcriber. This must be in the range [1, numCPUs]."
},
{
"key": "EnableLiveCaptions",
"display_name": "Enable live captions (Experimental)",
"type": "bool",
"default": false,
"help_text": "(Optional) When set to true, live captions are enabled."
},
{
"key": "LiveCaptionsModelSize",
"display_name": "Live captions: Model size",
"type": "dropdown",
"default": "tiny",
"help_text": "The speech-to-text model size to use for live captions. Heavier models will produce more accurate results at the expense of processing time and resources usage.",
"options": [
{
"display_name": "Tiny",
"value": "tiny"
},
{
"display_name": "Base",
"value": "base"
},
{
"display_name": "Small",
"value": "small"
}
],
"hosting": "on-prem"
},
{
"key": "LiveCaptionsNumTranscribers",
"display_name": "Live captions: Number of transcribers used per call",
"type": "number",
"default": 1,
"help_text": "The number of separate live-captions transcribers for each call. Each transcribes one audio stream at a time. The product of LiveCaptionsNumTranscribers * LiveCaptionsNumThreadsPerTranscriber must be in the range [1, numCPUs]."
},
{
"key": "LiveCaptionsNumThreadsPerTranscriber",
"display_name": "Live captions: Number of threads per transcriber",
"type": "number",
"default": 2,
"help_text": "The number of threads per live-captions transcriber. The product of LiveCaptionsNumTranscribers * LiveCaptionsNumThreadsPerTranscriber must be in the range [1, numCPUs]."
},
{
"key": "LiveCaptionsLanguage",
"display_name": "Live captions language",
"type": "text",
"default": "en",
"help_text": "The language passed to the live captions transcriber. Should be a 2-letter ISO 639 Set 1 language code, e.g. 'en'. If blank, will be set to English 'en' as default."
},
{
"key": "EnableIPv6",
"display_name": "(Experimental) Enable IPv6 support",
"type": "bool",
"help_text": "When set to true the RTC service will work in dual-stack mode, listening for IPv6 connections and generating candidates in addition to IPv4 ones.",
"default": false,
"hosting": "on-prem"
},
{
"key": "EnableRinging",
"display_name": "Enable call ringing (Beta)",
"type": "bool",
"default": false,
"help_text": "When set to true, ringing functionality is enabled: participants in DM and GM channels will receive a desktop alert and a ringing notification when a call is started. Changing this setting requires a plugin restart."
}
]
},
"props": {
"min_rtcd_version": "v0.12.0",
"min_offloader_version": "v0.8.0",
"calls_recorder_version": "v0.7.2",
"calls_transcriber_version": "v0.2.2"
}
}