livekit-protocol 0.5.1

// 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 rpc;

option go_package = "github.com/livekit/protocol/rpc";

import "google/protobuf/empty.proto";
import "google/protobuf/duration.proto";
import "options.proto";
import "livekit_sip.proto";

service SIPInternal {
  rpc CreateSIPParticipant(InternalCreateSIPParticipantRequest) returns (InternalCreateSIPParticipantResponse) {
    option (psrpc.options).affinity_func = true;
    option (psrpc.options).topics = true;
  };
  rpc TransferSIPParticipant(InternalTransferSIPParticipantRequest) returns (google.protobuf.Empty) {
    option (psrpc.options) = {
      topics: true
      topic_params: {
        names: ["sip_call_id"]
        typed: false
      }
    };
  };
}

message InternalCreateSIPParticipantRequest {
  // Used in Cloud only
  string project_id = 18;

  string sip_call_id = 13;
  string sip_trunk_id = 19;
  // IP or hostname that SIP INVITE is sent too
  string address = 2;
  // Hostname for the 'From' SIP address in INVITE
  string hostname = 20;
  string destination_country = 30;
  livekit.SIPTransport transport = 16;

  // Number used to make the call
  string number = 3;

  // Number to call to
  string call_to = 4;

  string username = 5;
  string password = 6;

  string room_name = 7;
  string participant_identity = 8;
  string participant_name = 14;
  string participant_metadata = 15;
  map<string, string> participant_attributes = 17;

  // optional token that should be used when creating LiveKit participant
  string token = 9;

  // optional websocket url that should be used when creating LiveKit participant
  string ws_url = 10;

  // Optionally send following DTMF digits (extension codes) when making a call.
  // Character 'w' can be used to add a 0.5 sec delay.
  string dtmf = 11;

  // Optionally play dialtone in the room as an audible indicator for existing participants
  bool play_dialtone = 12;

  map<string, string> headers = 21;
  map<string, string> headers_to_attributes = 22;
  // Map LiveKit attributes to SIP X-* headers when sending BYE or REFER requests.
  // Keys are the names of attributes and values are the names of X-* headers they will be mapped to.
  map<string, string> attributes_to_headers = 26;
  // Map SIP headers from 200 OK to sip.h.* participant attributes automatically.
  //
  // When the names of required headers is known, using headers_to_attributes is strongly recommended.
  //
  // When mapping 200 OK headers to follow-up request headers with attributes_to_headers map,
  // lowercase header names should be used, for example: sip.h.x-custom-header.
  livekit.SIPHeaderOptions include_headers = 27;

  repeated livekit.SIPFeature enabled_features = 25;

  // Max time for the callee to answer the call.
  google.protobuf.Duration ringing_timeout = 23;
  // Max call duration.
  google.protobuf.Duration max_call_duration = 24;

  livekit.SIPMediaEncryption media_encryption = 28;

  // Wait for the answer for the call before returning.
  bool wait_until_answered = 29;

  // NEXT ID: 31
}

message InternalCreateSIPParticipantResponse {
  string participant_id = 1;
  string participant_identity = 2;
  string sip_call_id = 3;
}

message InternalTransferSIPParticipantRequest {
  string sip_call_id = 1;
  string transfer_to = 2;

  // Optionally play dialtone to the SIP participant as an audible indicator of being transferred
  bool play_dialtone = 3;

  // Add the following headers to the REFER SIP request.
  map<string, string> headers = 4;

  // Max time for the transfer destination to answer the call.
  google.protobuf.Duration ringing_timeout = 5;
}