github.com/livekit/protocol@v1.39.3/protobufs/livekit_models.proto

// Copyright 2023 LiveKit, Inc.
//
// Licensed under the Apache License, Version 2.0 (the "License");
// you may not use this file except in compliance with the License.
// You may obtain a copy of the License at
//
//     http://www.apache.org/licenses/LICENSE-2.0
//
// Unless required by applicable law or agreed to in writing, software
// distributed under the License is distributed on an "AS IS" BASIS,
// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
// See the License for the specific language governing permissions and
// limitations under the License.

syntax = "proto3";

package livekit;
option go_package = "github.com/livekit/protocol/livekit";
option csharp_namespace = "LiveKit.Proto";
option ruby_package = "LiveKit::Proto";

import "google/protobuf/timestamp.proto";

import "livekit_metrics.proto";

message Pagination {
  string after_id = 1; // list entities which IDs are greater
  int32  limit = 2;
}

// ListUpdate is used for updated APIs where 'repeated string' field is modified.
message ListUpdate {
  repeated string set = 1; // set the field to a new list
}

message Room {
  string sid = 1;
  string name = 2;
  uint32 empty_timeout = 3;
  uint32 departure_timeout = 14;
  uint32 max_participants = 4;
  int64 creation_time = 5;
  int64 creation_time_ms = 15;
  string turn_password = 6;
  repeated Codec enabled_codecs = 7;
  string metadata = 8;
  uint32 num_participants = 9;
  uint32 num_publishers = 11;
  bool active_recording = 10;
  TimedVersion version = 13;

  // NEXT-ID: 16
}

message Codec {
  string mime = 1;
  string fmtp_line = 2;
}

enum AudioCodec {
  DEFAULT_AC = 0;
  OPUS = 1;
  AAC = 2;
}

enum VideoCodec {
  DEFAULT_VC = 0;
  H264_BASELINE = 1;
  H264_MAIN = 2;
  H264_HIGH = 3;
  VP8 = 4;
}

enum ImageCodec {
  IC_DEFAULT = 0;
  IC_JPEG = 1;
}

// Policy for publisher to handle subscribers that are unable to support the primary codec of a track
enum BackupCodecPolicy {
  // default behavior, the track prefer to regress to backup codec and all subscribers will receive the backup codec,
  // the sfu will try to regress codec if possible but not assured.
  PREFER_REGRESSION = 0;
  // encoding/send the primary and backup codec simultaneously
  SIMULCAST = 1;
  // force the track to regress to backup codec, this option can be used in video conference or the publisher has limited bandwidth/encoding power
  REGRESSION = 2;
}

message PlayoutDelay {
  bool enabled = 1;
  uint32 min = 2;
  uint32 max = 3;
}

message ParticipantPermission {
  // allow participant to subscribe to other tracks in the room
  bool can_subscribe = 1;
  // allow participant to publish new tracks to room
  bool can_publish = 2;
  // allow participant to publish data
  bool can_publish_data = 3;
  // sources that are allowed to be published
  repeated TrackSource can_publish_sources = 9;
  // indicates that it's hidden to others
  bool hidden = 7;
  // indicates it's a recorder instance
  // deprecated: use ParticipantInfo.kind instead
  bool recorder = 8 [deprecated=true];
  // indicates that participant can update own metadata and attributes
  bool can_update_metadata = 10;
  // indicates that participant is an agent
  // deprecated: use ParticipantInfo.kind instead
  bool agent = 11 [deprecated=true];
  // if a participant can subscribe to metrics
  bool can_subscribe_metrics = 12;

  // NEXT_ID: 13
}

message ParticipantInfo {
  enum State {
    // websocket' connected, but not offered yet
    JOINING = 0;
    // server received client offer
    JOINED = 1;
    // ICE connectivity established
    ACTIVE = 2;
    // WS disconnected
    DISCONNECTED = 3;
  }
  enum Kind {
    // standard participants, e.g. web clients
    STANDARD = 0;
    // only ingests streams
    INGRESS = 1;
    // only consumes streams
    EGRESS = 2;
    // SIP participants
    SIP = 3;
    // LiveKit agents
    AGENT = 4;

    // NEXT_ID: 7
  }
  enum KindDetail {
    CLOUD_AGENT = 0;
    FORWARDED = 1;
  }
  string sid = 1;
  string identity = 2;
  State state = 3;
  repeated TrackInfo tracks = 4;
  string metadata = 5;
  // timestamp when participant joined room, in seconds
  int64 joined_at = 6;
  // timestamp when participant joined room, in milliseconds
  int64 joined_at_ms = 17;
  string name = 9;
  uint32 version = 10;
  ParticipantPermission permission = 11;
  string region = 12;
  // indicates the participant has an active publisher connection
  // and can publish to the server
  bool is_publisher = 13;
  Kind kind = 14;
  map<string, string> attributes = 15;
  DisconnectReason disconnect_reason = 16;
  repeated KindDetail kind_details = 18;

  // NEXT_ID: 19
}

enum TrackType {
  AUDIO = 0;
  VIDEO = 1;
  DATA = 2;
}

enum TrackSource {
  UNKNOWN = 0;
  CAMERA = 1;
  MICROPHONE = 2;
  SCREEN_SHARE = 3;
  SCREEN_SHARE_AUDIO = 4;
}

message Encryption {
  enum Type {
      NONE=0;
      GCM=1;
      CUSTOM=2;
  }
}

message SimulcastCodecInfo {
  string mime_type = 1;
  string mid = 2;
  string cid = 3;
  repeated VideoLayer layers = 4;
}

message TrackInfo {
  string sid = 1;
  TrackType type = 2;
  string name = 3;
  bool muted = 4;
  // original width of video (unset for audio)
  // clients may receive a lower resolution version with simulcast
  uint32 width = 5;
  // original height of video (unset for audio)
  uint32 height = 6;
  // true if track is simulcasted
  bool simulcast = 7;
  // true if DTX (Discontinuous Transmission) is disabled for audio
  bool disable_dtx = 8;
  // source of media
  TrackSource source = 9;
  repeated VideoLayer layers = 10;
  // mime type of codec
  string mime_type = 11;
  string mid = 12;
  repeated SimulcastCodecInfo codecs = 13;
  bool stereo = 14;
  // true if RED (Redundant Encoding) is disabled for audio
  bool disable_red = 15;
  Encryption.Type encryption = 16;
  string stream = 17;
  TimedVersion version = 18;
  repeated AudioTrackFeature audio_features = 19;
  BackupCodecPolicy backup_codec_policy = 20;
}

enum VideoQuality {
  LOW = 0;
  MEDIUM = 1;
  HIGH = 2;
  OFF = 3;
}

// provide information about available spatial layers
message VideoLayer {
  // for tracks with a single layer, this should be HIGH
  VideoQuality quality = 1;
  uint32 width = 2;
  uint32 height = 3;
  // target bitrate in bit per second (bps), server will measure actual
  uint32 bitrate = 4;
  uint32 ssrc = 5;
  int32 spatial_layer = 6;
  string rid = 7;
}

// new DataPacket API
message DataPacket {
  enum Kind {
    RELIABLE = 0;
    LOSSY = 1;
  }
  Kind kind = 1 [deprecated=true];
  // participant identity of user that sent the message
  string participant_identity = 4;
  // identities of participants who will receive the message (sent to all by default)
  repeated string destination_identities = 5;
  oneof value {
    UserPacket user = 2;
    ActiveSpeakerUpdate speaker = 3 [deprecated=true];
    SipDTMF sip_dtmf = 6;
    Transcription transcription = 7;
    MetricsBatch metrics = 8;
    ChatMessage chat_message = 9;
    RpcRequest rpc_request = 10;
    RpcAck rpc_ack = 11;
    RpcResponse rpc_response = 12;
    DataStream.Header stream_header = 13;
    DataStream.Chunk stream_chunk = 14;
    DataStream.Trailer stream_trailer = 15;
  }
  // sequence number of reliable packet
  uint32 sequence = 16;
  // sid of the user that sent the message
  string participant_sid = 17;

  // NEXT_ID: 18
}

message ActiveSpeakerUpdate {
  repeated SpeakerInfo speakers = 1;
}

message SpeakerInfo {
  string sid = 1;
  // audio level, 0-1.0, 1 is loudest
  float level = 2;
  // true if speaker is currently active
  bool active = 3;
}

message UserPacket {
  // participant ID of user that sent the message
  string participant_sid = 1 [deprecated=true];
  string participant_identity = 5 [deprecated=true];
  // user defined payload
  bytes payload = 2;
  // the ID of the participants who will receive the message (sent to all by default)
  repeated string destination_sids = 3 [deprecated=true];
  // identities of participants who will receive the message (sent to all by default)
  repeated string destination_identities = 6 [deprecated=true];
  // topic under which the message was published
  optional string topic = 4;
  // Unique ID to indentify the message
  optional string id = 8;
  // start and end time allow relating the message to specific media time
  optional uint64 start_time = 9;
  optional uint64 end_time = 10;
  // added by SDK to enable de-duping of messages, for INTERNAL USE ONLY
  bytes nonce = 11;

  // NEXT_ID: 12
}

message SipDTMF {
  uint32 code = 3;
  string digit = 4;
}

message Transcription {
  // Participant that got its speech transcribed
  string transcribed_participant_identity = 2;
  string track_id = 3;
  repeated TranscriptionSegment segments = 4;

  // NEXT_ID: 6
}

message TranscriptionSegment {
  string id = 1;
  string text = 2;
  uint64 start_time = 3;
  uint64 end_time = 4;
  bool final = 5;
  string language = 6;
}

message ChatMessage {
  string id = 1; // uuid
  int64 timestamp = 2;
  optional int64 edit_timestamp = 3; // populated only if the intent is to edit/update an existing message
  string message = 4;
  bool deleted = 5; // true to remove message
  bool generated = 6; // true if the chat message has been generated by an agent from a participant's audio transcription
}

message RpcRequest {
  string id = 1;
  string method = 2;
  string payload = 3;
  uint32 response_timeout_ms = 4;
  uint32 version = 5;
}

message RpcAck {
  string request_id = 1;
}

message RpcResponse {
  string request_id = 1;
  oneof value {
    string payload = 2;
    RpcError error = 3;
  }
}

message RpcError {
  uint32 code = 1;
  string message = 2;
  string data = 3;
}

enum ConnectionQuality {
  POOR = 0;
  GOOD = 1;
  EXCELLENT = 2;
  LOST = 3;
}

message ParticipantTracks {
  // participant ID of participant to whom the tracks belong
  string participant_sid = 1;
  repeated string track_sids = 2;
}

// details about the server
message ServerInfo {
  enum Edition {
    Standard = 0;
    Cloud = 1;
  }
  Edition edition = 1;
  string version = 2;
  int32 protocol = 3;
  string region = 4;
  string node_id = 5;
  // additional debugging information. sent only if server is in development mode
  string debug_info = 6;
  int32 agent_protocol = 7;
}

// details about the client
message ClientInfo {
  enum SDK {
     UNKNOWN = 0;
     JS = 1;
     SWIFT = 2;
     ANDROID = 3;
     FLUTTER = 4;
     GO = 5;
     UNITY = 6;
     REACT_NATIVE = 7;
     RUST = 8;
     PYTHON = 9;
     CPP = 10;
     UNITY_WEB = 11;
     NODE = 12;
     UNREAL = 13;
     ESP32 = 14;
  }

  SDK sdk = 1;
  string version = 2;
  int32 protocol = 3;
  string os = 4;
  string os_version = 5;
  string device_model = 6;
  string browser = 7;
  string browser_version = 8;
  string address = 9;
  // wifi, wired, cellular, vpn, empty if not known
  string network = 10;
  // comma separated list of additional LiveKit SDKs in use of this client, with versions
  // e.g. "components-js:1.2.3,track-processors-js:1.2.3"
  string other_sdks = 11;
}

// server provided client configuration
message ClientConfiguration {
  VideoConfiguration video = 1;
  VideoConfiguration screen = 2;

  ClientConfigSetting resume_connection = 3;
  DisabledCodecs disabled_codecs = 4;
  ClientConfigSetting force_relay = 5;
}

enum ClientConfigSetting {
  UNSET = 0;
  DISABLED = 1;
  ENABLED = 2;
}

message VideoConfiguration {
  ClientConfigSetting hardware_encoder = 1;
}

message DisabledCodecs {
  // disabled for both publish and subscribe
  repeated Codec codecs = 1;
  // only disable for publish
  repeated Codec publish = 2;
}

enum DisconnectReason {
  UNKNOWN_REASON = 0;
  // the client initiated the disconnect
  CLIENT_INITIATED = 1;
  // another participant with the same identity has joined the room
  DUPLICATE_IDENTITY = 2;
  // the server instance is shutting down
  SERVER_SHUTDOWN = 3;
  // RoomService.RemoveParticipant was called
  PARTICIPANT_REMOVED = 4;
  // RoomService.DeleteRoom was called
  ROOM_DELETED = 5;
  // the client is attempting to resume a session, but server is not aware of it
  STATE_MISMATCH = 6;
  // client was unable to connect fully
  JOIN_FAILURE = 7;
  // Cloud-only, the server requested Participant to migrate the connection elsewhere
  MIGRATION = 8;
  // the signal websocket was closed unexpectedly
  SIGNAL_CLOSE = 9;
  // the room was closed, due to all Standard and Ingress participants having left
  ROOM_CLOSED = 10;
  // SIP callee did not respond in time
  USER_UNAVAILABLE = 11;
  // SIP callee rejected the call (busy)
  USER_REJECTED = 12;
  // SIP protocol failure or unexpected response
  SIP_TRUNK_FAILURE = 13;
  // server timed out a participant session
  CONNECTION_TIMEOUT = 14;
  // media stream failure or media timeout
  MEDIA_FAILURE = 15;
}

message RTPDrift {
  google.protobuf.Timestamp start_time = 1;
  google.protobuf.Timestamp end_time = 2;
  double duration = 3;

  uint64 start_timestamp = 4;
  uint64 end_timestamp = 5;
  uint64 rtp_clock_ticks = 6;
  int64 drift_samples = 7;
  double drift_ms = 8;
  double clock_rate = 9;
}

message RTPStats {
  google.protobuf.Timestamp start_time = 1;
  google.protobuf.Timestamp end_time = 2;
  double duration = 3;

  uint32 packets = 4;
  double packet_rate = 5;

  uint64 bytes = 6;
  uint64 header_bytes = 39;
  double bitrate = 7;

  uint32 packets_lost = 8;
  double packet_loss_rate = 9;
  float packet_loss_percentage = 10;

  uint32 packets_duplicate = 11;
  double packet_duplicate_rate = 12;

  uint64 bytes_duplicate = 13;
  uint64 header_bytes_duplicate = 40;
  double bitrate_duplicate = 14;

  uint32 packets_padding = 15;
  double packet_padding_rate = 16;

  uint64 bytes_padding = 17;
  uint64 header_bytes_padding = 41;
  double bitrate_padding = 18;

  uint32 packets_out_of_order = 19;

  uint32 frames = 20;
  double frame_rate = 21;

  double jitter_current = 22;
  double jitter_max = 23;

  map<int32, uint32> gap_histogram = 24;

  uint32 nacks = 25;
  uint32 nack_acks = 37;
  uint32 nack_misses = 26;
  uint32 nack_repeated = 38;

  uint32 plis = 27;
  google.protobuf.Timestamp last_pli = 28;

  uint32 firs = 29;
  google.protobuf.Timestamp last_fir = 30;

  uint32 rtt_current = 31;
  uint32 rtt_max = 32;

  uint32 key_frames = 33;
  google.protobuf.Timestamp last_key_frame = 34;

  uint32 layer_lock_plis = 35;
  google.protobuf.Timestamp last_layer_lock_pli = 36;

  RTPDrift packet_drift = 44;
  RTPDrift ntp_report_drift = 45;
  RTPDrift rebased_report_drift = 46;
  RTPDrift received_report_drift = 47;
  // NEXT_ID: 48
}

message RTCPSenderReportState {
  uint32 rtp_timestamp = 1;
  uint64 rtp_timestamp_ext = 2;
  uint64 ntp_timestamp = 3;
  int64 at = 4; // time at which this happened
  int64 at_adjusted = 5;
  uint32 packets = 6;
  uint64 octets = 7;
}

message RTPForwarderState {
  bool started = 1;
  int32 reference_layer_spatial = 2;
  int64 pre_start_time = 3;
  uint64 ext_first_timestamp = 4;
  uint64 dummy_start_timestamp_offset = 5;
  RTPMungerState rtp_munger = 6;
  oneof codec_munger {
    VP8MungerState vp8_munger = 7;
  }
  repeated RTCPSenderReportState sender_report_state = 8;
}

message RTPMungerState {
  uint64 ext_last_sequence_number = 1;
  uint64 ext_second_last_sequence_number = 2;
  uint64 ext_last_timestamp = 3;
  uint64 ext_second_last_timestamp = 4;
  bool last_marker = 5;
  bool second_last_marker = 6;
}

message VP8MungerState {
  int32 ext_last_picture_id = 1;
  bool picture_id_used = 2;
  uint32 last_tl0_pic_idx = 3;
  bool tl0_pic_idx_used = 4;
  bool tid_used = 5;
  uint32 last_key_idx = 6;
  bool key_idx_used = 7;
}

message TimedVersion {
  int64 unix_micro = 1;
  int32 ticks = 2;
}

enum ReconnectReason {
  RR_UNKNOWN = 0;
  RR_SIGNAL_DISCONNECTED = 1;
  RR_PUBLISHER_FAILED = 2;
  RR_SUBSCRIBER_FAILED = 3;
  RR_SWITCH_CANDIDATE = 4;
}

enum SubscriptionError {
  SE_UNKNOWN = 0;
  SE_CODEC_UNSUPPORTED = 1;
  SE_TRACK_NOTFOUND = 2;
}

enum AudioTrackFeature {
  TF_STEREO = 0;
  TF_NO_DTX = 1;
  TF_AUTO_GAIN_CONTROL = 2;
  TF_ECHO_CANCELLATION = 3;
  TF_NOISE_SUPPRESSION = 4;
  TF_ENHANCED_NOISE_CANCELLATION = 5;
  TF_PRECONNECT_BUFFER = 6; // client will buffer audio once available and send it to the server via bytes stream once connected
}

message DataStream {

  // enum for operation types (specific to TextHeader)
  enum OperationType {
    CREATE = 0;
    UPDATE = 1;
    DELETE = 2;
    REACTION = 3;
  }

  // header properties specific to text streams
  message TextHeader {
    OperationType operation_type = 1;
    int32 version = 2;  // Optional: Version for updates/edits
    string reply_to_stream_id = 3;  // Optional: Reply to specific message
    repeated string attached_stream_ids = 4; // file attachments for text streams
    bool generated = 5; // true if the text has been generated by an agent from a participant's audio transcription

  }

  // header properties specific to byte or file streams
  message ByteHeader {
    string name = 1;
  }

  // main DataStream.Header that contains a oneof for specific headers
  message Header {
    string stream_id = 1; // unique identifier for this data stream
    int64 timestamp = 2;  // using int64 for Unix timestamp
    string topic = 3;
    string mime_type = 4;
    optional uint64 total_length = 5;  // only populated for finite streams, if it's a stream of unknown size this stays empty
    Encryption.Type encryption_type = 7;  // defaults to NONE
    map<string, string> attributes = 8;  // user defined attributes map that can carry additional info

    // oneof to choose between specific header types
    oneof content_header {
      TextHeader text_header = 9;
      ByteHeader byte_header = 10;
    }
  }

  message Chunk {
    string stream_id = 1;  // unique identifier for this data stream to map it to the correct header
    uint64 chunk_index = 2;
    bytes content = 3; // content as binary (bytes)
    int32 version = 4; // a version indicating that this chunk_index has been retroactively modified and the original one needs to be replaced
    optional bytes iv = 5; // optional, initialization vector for AES-GCM encryption
  }

  message Trailer {
    string stream_id = 1; // unique identifier for this data stream
    string reason = 2; // reason why the stream was closed (could contain "error" / "interrupted" / empty for expected end)
    map<string, string> attributes = 3; // finalizing updates for the stream, can also include additional insights for errors or endTime for transcription
  }
}

message WebhookConfig {
  string url = 1;
  string signing_key = 2;
}