LiveKit Real-Time Communication API

your-livekit-host.livekit.cloud

Community

LiveKit is an open-source WebRTC platform for building real-time voice, video, and data applications. Use this API to manage rooms, participants, tracks, recording/streaming via egress, external media ingress, and SIP telephony integration for AI agents and communication workflows.

[catalog]Catalog only — enable via gateway with your own API key

Enable this service

This service is in the catalog but not part of v1 launch billing. To use it, run the command below and provide your own API key in the local config page — the gateway will call it under your account.

Auth

api_key

Pricing

free

Spec version

1.0

Base URL

https://your-livekit-host.livekit.cloud

Last crawled

May 4, 2026

Health

This is a community-maintained manifest. Health monitoring is not available because this service doesn't host its own /.well-known/agent endpoint yet. Learn more about trust levels →

Top Capabilities

Most commonly used operations — ready to call without drill-down.

egress_list

POST

List all active egress sessions, optionally filtered by room name.

POST https://your-livekit-host.livekit.cloud/twirp/livekit.Egress/ListEgress

room_name (string) — Filter by room name. Returns all active egress if empty.

participants_list

POST

Retrieve a list of all participants currently in a specific room.

POST https://your-livekit-host.livekit.cloud/twirp/livekit.RoomService/ListParticipants

room (string, required) — Name of the room to list participants for.

rooms_list

POST

List all active rooms, optionally filtered by room names.

POST https://your-livekit-host.livekit.cloud/twirp/livekit.RoomService/ListRooms

names (array) — Optional list of room names to filter by. Returns all rooms if empty.

All Capabilities (27)

rooms (5)

POSTdata_send

Send data packets to participants in a room via reliable or lossy transport.

POSTrooms_create

Create a new room with configurable settings including timeout, max participants, metadata, and egress options.

POSTrooms_delete

Delete a room and disconnect all participants currently in it.

POSTrooms_list

List all active rooms, optionally filtered by room names.

POSTrooms_metadata_update

Update the metadata of an active room and broadcast the change to all participants.

egress (7)

POSTegress_layout_update

Update the web layout of an active room composite egress session.

POSTegress_list

List all active egress sessions, optionally filtered by room name.

POSTegress_room_composite_start

Start recording or streaming an entire room as a composite video/audio output to file, stream, or segments.

POSTegress_stop

Stop an active egress recording or streaming session.

POSTegress_track_composite_start

Start recording specific audio and video tracks from a participant as a composite output.

POSTegress_track_start

Start recording an individual track directly to a file or WebSocket.

POSTegress_web_start

Start recording an arbitrary web page as video/audio output.

participants (4)

POSTparticipants_get

Retrieve detailed information about a specific participant by identity in a room.

POSTparticipants_list

Retrieve a list of all participants currently in a specific room.

POSTparticipants_remove

Remove a participant from a room, disconnecting them immediately.

POSTparticipants_update

Update a participant's metadata, permissions, or display name in a room.

sip (9)

POSTsip_dispatch_rule_create

Create a dispatch rule to route incoming SIP calls to specific rooms or generate individual rooms per caller.

POSTsip_dispatch_rules_list

List all configured SIP dispatch rules.

POSTsip_inbound_trunk_create

Create an inbound SIP trunk for receiving phone calls into LiveKit rooms.

POSTsip_inbound_trunks_list

List all configured inbound SIP trunks.

POSTsip_outbound_trunk_create

Create an outbound SIP trunk for placing calls from LiveKit rooms to phone numbers.

POSTsip_outbound_trunks_list

List all configured outbound SIP trunks.

POSTsip_participant_create

Initiate an outbound SIP/PSTN call and connect it as a participant in a LiveKit room.

POSTsip_participant_transfer

Transfer an active SIP call participant to another phone number or SIP endpoint.

POSTsip_trunk_delete

Delete a SIP trunk (inbound or outbound) by its trunk ID.

tracks (2)

POSTtracks_mute

Mute or unmute a published audio or video track for a participant in a room.

POSTtracks_subscriptions_update

Subscribe or unsubscribe a participant from specific tracks in a room.

Agent Preview

This is what an AI agent sees when it discovers this service via the Gateway:

Service: LiveKit Real-Time Communication API
Description: LiveKit is an open-source WebRTC platform for building real-time voice, video, and data applications. Use this API to manage rooms, participants, tracks, recording/streaming via egress, external media ingress, and SIP telephony integration for AI agents and communication workflows.
Auth: api_key
Capabilities (27 total):
  rooms (5):
    - data_send: Send data packets to participants in a room via reliable or lossy transport.
    - rooms_create: Create a new room with configurable settings including timeout, max participants, metadata, and egress options.
    - rooms_delete: Delete a room and disconnect all participants currently in it.
    - rooms_list: List all active rooms, optionally filtered by room names.
    - rooms_metadata_update: Update the metadata of an active room and broadcast the change to all participants.
  egress (7):
    - egress_layout_update: Update the web layout of an active room composite egress session.
    - egress_list: List all active egress sessions, optionally filtered by room name.
    - egress_room_composite_start: Start recording or streaming an entire room as a composite video/audio output to file, stream, or segments.
    - egress_stop: Stop an active egress recording or streaming session.
    - egress_track_composite_start: Start recording specific audio and video tracks from a participant as a composite output.
    - egress_track_start: Start recording an individual track directly to a file or WebSocket.
    - egress_web_start: Start recording an arbitrary web page as video/audio output.
  participants (4):
    - participants_get: Retrieve detailed information about a specific participant by identity in a room.
    - participants_list: Retrieve a list of all participants currently in a specific room.
    - participants_remove: Remove a participant from a room, disconnecting them immediately.
    - participants_update: Update a participant's metadata, permissions, or display name in a room.
  sip (9):
    - sip_dispatch_rule_create: Create a dispatch rule to route incoming SIP calls to specific rooms or generate individual rooms per caller.
    - sip_dispatch_rules_list: List all configured SIP dispatch rules.
    - sip_inbound_trunk_create: Create an inbound SIP trunk for receiving phone calls into LiveKit rooms.
    - sip_inbound_trunks_list: List all configured inbound SIP trunks.
    - sip_outbound_trunk_create: Create an outbound SIP trunk for placing calls from LiveKit rooms to phone numbers.
    - sip_outbound_trunks_list: List all configured outbound SIP trunks.
    - sip_participant_create: Initiate an outbound SIP/PSTN call and connect it as a participant in a LiveKit room.
    - sip_participant_transfer: Transfer an active SIP call participant to another phone number or SIP endpoint.
    - sip_trunk_delete: Delete a SIP trunk (inbound or outbound) by its trunk ID.
  tracks (2):
    - tracks_mute: Mute or unmute a published audio or video track for a participant in a room.
    - tracks_subscriptions_update: Subscribe or unsubscribe a participant from specific tracks in a room.

Top capabilities (ready to call):
  - egress_list: POST /twirp/livekit.Egress/ListEgress
  - participants_list: POST /twirp/livekit.RoomService/ListParticipants
  - rooms_list: POST /twirp/livekit.RoomService/ListRooms