{
    "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",
    "release_notes_url": "https://github.com/mattermost/mattermost-plugin-calls/releases/tag/v0.29.6",
    "icon_path": "assets/plugin_icon.svg",
    "version": "0.29.6",
    "min_server_version": "9.5.0",
    "server": {
        "executables": {
            "freebsd-amd64": "server/dist/plugin-freebsd-amd64",
            "linux-amd64": "server/dist/plugin-linux-amd64",
            "linux-arm64": "server/dist/plugin-linux-arm64",
            "openbsd-amd64": "server/dist/plugin-openbsd-amd64"
        },
        "executable": ""
    },
    "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.",
                "placeholder": "",
                "default": null,
                "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.",
                "placeholder": "127.0.0.1",
                "default": "",
                "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.",
                "placeholder": "127.0.0.1",
                "default": "",
                "hosting": "on-prem"
            },
            {
                "key": "UDPServerPort",
                "display_name": "RTC Server Port (UDP)",
                "type": "number",
                "help_text": "The UDP port the RTC server will listen on.",
                "placeholder": "8443",
                "default": 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.",
                "placeholder": "8443",
                "default": 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.",
                "placeholder": "",
                "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",
                "placeholder": "",
                "default": null,
                "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",
                "default": null,
                "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.",
                "placeholder": "",
                "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.",
                "placeholder": "[{\n \"urls\":[\"turn:turnserver.example.org:3478\"],\n \"username\": \"webrtc\",\n \"credential\": \"turnpassword\"\n}]",
                "default": "[{\"urls\":[\"stun:stun.global.calls.mattermost.com:3478\"]}]",
                "hosting": "on-prem"
            },
            {
                "key": "TURNStaticAuthSecret",
                "display_name": "TURN Static Auth Secret",
                "type": "text",
                "help_text": "(Optional) The secret key used to generate TURN short-lived authentication credentials.",
                "placeholder": "",
                "default": "",
                "hosting": "on-prem"
            },
            {
                "key": "TURNCredentialsExpirationMinutes",
                "display_name": "TURN Credentials Expiration (minutes)",
                "type": "number",
                "help_text": "(Optional) The number of minutes that the generated TURN credentials will be valid for.",
                "placeholder": "",
                "default": 1440,
                "hosting": "on-prem"
            },
            {
                "key": "ServerSideTURN",
                "display_name": "Server Side TURN",
                "type": "bool",
                "help_text": "(Optional) When set to on it will pass and use configured TURN candidates to server initiated connections.",
                "placeholder": "",
                "default": false,
                "hosting": "on-prem"
            },
            {
                "key": "AllowScreenSharing",
                "display_name": "Allow screen sharing",
                "type": "bool",
                "help_text": "When set to true it allows call participants to share their screen.",
                "placeholder": "",
                "default": true,
                "hosting": ""
            },
            {
                "key": "EnableSimulcast",
                "display_name": "Enable simulcast for screen sharing (Experimental)",
                "type": "bool",
                "help_text": "When set to true it enables simulcast for screen sharing. This can help to improve screen sharing quality.",
                "placeholder": "",
                "default": false,
                "hosting": ""
            },
            {
                "key": "EnableRecordings",
                "display_name": "Enable call recordings (Beta)",
                "type": "bool",
                "help_text": "(Optional) When set to true, call recordings are enabled.",
                "placeholder": "",
                "default": false,
                "hosting": ""
            },
            {
                "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",
                "default": null,
                "hosting": ""
            },
            {
                "key": "MaxRecordingDuration",
                "display_name": "Maximum call recording duration",
                "type": "number",
                "help_text": "The maximum duration (in minutes) for call recordings. Value must be in the range [15, 180].",
                "placeholder": "",
                "default": 60,
                "hosting": ""
            },
            {
                "key": "RecordingQuality",
                "display_name": "Call recording quality",
                "type": "dropdown",
                "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.",
                "placeholder": "",
                "default": "medium",
                "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",
                "help_text": "(Optional) When set to true, post-call transcriptions are enabled.",
                "placeholder": "",
                "default": false,
                "hosting": ""
            },
            {
                "key": "TranscribeAPI",
                "display_name": "Call transcriber API",
                "type": "dropdown",
                "help_text": "The speech-to-text API to use for post-call transcriptions.",
                "placeholder": "",
                "default": "whisper.cpp",
                "options": [
                    {
                        "display_name": "Whisper.CPP",
                        "value": "whisper.cpp"
                    },
                    {
                        "display_name": "Azure AI",
                        "value": "azure"
                    }
                ],
                "hosting": "on-prem"
            },
            {
                "key": "TranscriberModelSize",
                "display_name": "Call transcriber model size",
                "type": "dropdown",
                "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.",
                "placeholder": "",
                "default": "base",
                "options": [
                    {
                        "display_name": "Tiny",
                        "value": "tiny"
                    },
                    {
                        "display_name": "Base",
                        "value": "base"
                    },
                    {
                        "display_name": "Small",
                        "value": "small"
                    }
                ],
                "hosting": "on-prem"
            },
            {
                "key": "TranscribeAPIAzureSpeechKey",
                "display_name": "Azure Speech Services API Key",
                "type": "text",
                "help_text": "The API key for Azure Speech Services",
                "placeholder": "",
                "default": "",
                "hosting": "on-prem"
            },
            {
                "key": "TranscribeAPIAzureSpeechRegion",
                "display_name": "Azure Speech Services API Region",
                "type": "text",
                "help_text": "The API region for Azure Speech Services",
                "placeholder": "",
                "default": "",
                "hosting": "on-prem"
            },
            {
                "key": "TranscriberNumThreads",
                "display_name": "Call transcriber threads",
                "type": "number",
                "help_text": "The number of threads used by the post-call transcriber. This must be in the range [1, numCPUs].",
                "placeholder": "",
                "default": 2,
                "hosting": ""
            },
            {
                "key": "EnableLiveCaptions",
                "display_name": "Enable live captions (Experimental)",
                "type": "bool",
                "help_text": "(Optional) When set to true, live captions are enabled.",
                "placeholder": "",
                "default": false,
                "hosting": ""
            },
            {
                "key": "LiveCaptionsModelSize",
                "display_name": "Live captions: Model size",
                "type": "dropdown",
                "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.",
                "placeholder": "",
                "default": "tiny",
                "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",
                "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].",
                "placeholder": "",
                "default": 1,
                "hosting": ""
            },
            {
                "key": "LiveCaptionsNumThreadsPerTranscriber",
                "display_name": "Live captions: Number of threads per transcriber",
                "type": "number",
                "help_text": "The number of threads per live-captions transcriber. The product of LiveCaptionsNumTranscribers * LiveCaptionsNumThreadsPerTranscriber must be in the range [1, numCPUs].",
                "placeholder": "",
                "default": 2,
                "hosting": ""
            },
            {
                "key": "LiveCaptionsLanguage",
                "display_name": "Live captions language",
                "type": "text",
                "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.",
                "placeholder": "",
                "default": "en",
                "hosting": ""
            },
            {
                "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.",
                "placeholder": "",
                "default": false,
                "hosting": "on-prem"
            },
            {
                "key": "EnableRinging",
                "display_name": "Enable call ringing (Beta)",
                "type": "bool",
                "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.",
                "placeholder": "",
                "default": false,
                "hosting": ""
            }
        ]
    },
    "props": {
        "calls_recorder_version": "v0.7.3",
        "calls_transcriber_version": "v0.3.1",
        "min_offloader_version": "v0.8.0",
        "min_rtcd_version": "v0.12.0"
    }
}