@livekit/protocol 1.53.0 → 1.53.2
This diff represents the content of publicly available package versions that have been released to one of the supported registries. The information contained in this diff is provided for informational purposes only and reflects changes between package versions as they appear in their respective public registries.
- package/dist/index.cjs +286 -12
- package/dist/index.cjs.map +1 -1
- package/dist/index.d.cts +585 -11
- package/dist/index.d.mts +585 -11
- package/dist/index.d.ts +585 -11
- package/dist/index.mjs +279 -13
- package/dist/index.mjs.map +1 -1
- package/package.json +3 -2
- package/src/gen/agent/livekit_agent_session_pb.d.ts +40 -0
- package/src/gen/agent/livekit_agent_session_pb.js +18 -0
- package/src/gen/livekit_agent_dispatch_pb.d.ts +38 -0
- package/src/gen/livekit_agent_dispatch_pb.js +16 -0
- package/src/gen/livekit_agent_pb.d.ts +167 -0
- package/src/gen/livekit_agent_pb.js +42 -0
- package/src/gen/livekit_agent_simulation_pb.d.ts +184 -3
- package/src/gen/livekit_agent_simulation_pb.js +54 -2
- package/src/gen/livekit_egress_pb.d.ts +62 -1
- package/src/gen/livekit_egress_pb.js +15 -0
- package/src/gen/livekit_internal_pb.d.ts +14 -0
- package/src/gen/livekit_internal_pb.js +2 -0
- package/src/gen/livekit_metrics_pb.d.ts +9 -0
- package/src/gen/livekit_metrics_pb.js +1 -0
- package/src/gen/livekit_sip_pb.d.ts +80 -5
- package/src/gen/livekit_sip_pb.js +22 -0
- package/src/gen/livekit_webhook_pb.d.ts +1 -1
- package/src/gen/version.js +1 -1
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@@ -484,6 +484,57 @@ export declare enum SIPMediaEncryption {
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SIP_MEDIA_ENCRYPT_REQUIRE = 2,
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}
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487
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+
/**
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* Dial tone (ringback) played while a call is being placed or transferred.
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489
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* Receivers treat unknown values as SIP_DIALTONE_ETSI.
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*
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* @generated from enum livekit.SIPDialtone
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*/
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export declare enum SIPDialtone {
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/**
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* no tone
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*
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* @generated from enum value: SIP_DIALTONE_DISABLED = 0;
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*/
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SIP_DIALTONE_DISABLED = 0,
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501
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/**
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* 425 Hz, 1s on / 4s off; the tone play_dialtone used
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*
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* @generated from enum value: SIP_DIALTONE_ETSI = 1;
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*/
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SIP_DIALTONE_ETSI = 1,
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/**
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* pick by the country of the called number, ETSI if unknown
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*
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* @generated from enum value: SIP_DIALTONE_AUTO = 2;
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*/
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SIP_DIALTONE_AUTO = 2,
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+
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/**
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* North America (NANP)
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*
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* @generated from enum value: SIP_DIALTONE_US = 3;
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*/
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SIP_DIALTONE_US = 3,
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/**
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* @generated from enum value: SIP_DIALTONE_UK = 4;
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*/
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SIP_DIALTONE_UK = 4,
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/**
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* @generated from enum value: SIP_DIALTONE_JP = 5;
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*/
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SIP_DIALTONE_JP = 5,
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/**
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* @generated from enum value: SIP_DIALTONE_AU = 6;
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*/
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SIP_DIALTONE_AU = 6,
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}
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+
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/**
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* @generated from enum livekit.ProviderType
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*/
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@@ -841,6 +892,14 @@ export declare class SIPMediaConfig extends Message<SIPMediaConfig> {
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*/
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mediaTimeout?: Duration;
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/**
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* Stop sending audio RTP while outbound DTMF digits are sent.
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* Always on when the telephone-event clock rate differs from the audio codec's.
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*
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* @generated from field: bool drop_audio_during_dtmf = 5;
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*/
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dropAudioDuringDtmf: boolean;
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constructor(data?: PartialMessage<SIPMediaConfig>);
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static readonly runtime: typeof proto3;
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@@ -2667,7 +2726,7 @@ export declare class CreateSIPParticipantRequest extends Message<CreateSIPPartic
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dtmf: string;
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/**
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-
*
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* Deprecated: use dialtone. Setting play_dialtone (or play_ringtone) is equivalent to SIP_DIALTONE_ETSI when dialtone is unset.
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*
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* @generated from field: bool play_ringtone = 6 [deprecated = true];
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* @deprecated
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@@ -2675,10 +2734,18 @@ export declare class CreateSIPParticipantRequest extends Message<CreateSIPPartic
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playRingtone: boolean;
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/**
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* @generated from field: bool play_dialtone = 13;
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* @generated from field: bool play_dialtone = 13 [deprecated = true];
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* @deprecated
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*/
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playDialtone: boolean;
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/**
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* Optionally play dialtone in the room as an audible indicator for existing participants.
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*
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* @generated from field: livekit.SIPDialtone dialtone = 28;
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*/
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dialtone: SIPDialtone;
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/**
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* By default the From value (Phone number) is used for participant name/identity (if not set) and added to attributes.
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* If true, a random value for identity will be used and numbers will be omitted from attributes.
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@@ -2758,7 +2825,7 @@ export declare class CreateSIPParticipantRequest extends Message<CreateSIPPartic
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displayName?: string;
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/**
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-
* NEXT ID:
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* NEXT ID: 29
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*
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* @generated from field: optional livekit.Destination destination = 22;
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*/
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@@ -2838,12 +2905,20 @@ export declare class TransferSIPParticipantRequest extends Message<TransferSIPPa
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transferTo: string;
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/**
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*
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* Deprecated: use dialtone. Setting play_dialtone is equivalent to SIP_DIALTONE_ETSI when dialtone is unset.
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*
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* @generated from field: bool play_dialtone = 4;
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* @generated from field: bool play_dialtone = 4 [deprecated = true];
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* @deprecated
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*/
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playDialtone: boolean;
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/**
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* Optionally play dialtone to the SIP participant as an audible indicator of being transferred.
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*
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* @generated from field: livekit.SIPDialtone dialtone = 7;
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*/
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dialtone: SIPDialtone;
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/**
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* Add the following headers to the REFER SIP request.
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*
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@@ -143,6 +143,25 @@ export const SIPMediaEncryption = /*@__PURE__*/ proto3.makeEnum(
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],
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);
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/**
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* Dial tone (ringback) played while a call is being placed or transferred.
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* Receivers treat unknown values as SIP_DIALTONE_ETSI.
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*
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* @generated from enum livekit.SIPDialtone
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*/
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export const SIPDialtone = /*@__PURE__*/ proto3.makeEnum(
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"livekit.SIPDialtone",
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[
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{no: 0, name: "SIP_DIALTONE_DISABLED"},
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{no: 1, name: "SIP_DIALTONE_ETSI"},
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{no: 2, name: "SIP_DIALTONE_AUTO"},
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{no: 3, name: "SIP_DIALTONE_US"},
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{no: 4, name: "SIP_DIALTONE_UK"},
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{no: 5, name: "SIP_DIALTONE_JP"},
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{no: 6, name: "SIP_DIALTONE_AU"},
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],
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);
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/**
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* @generated from enum livekit.ProviderType
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*/
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@@ -275,6 +294,7 @@ export const SIPMediaConfig = /*@__PURE__*/ proto3.makeMessageType(
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{ no: 2, name: "codecs", kind: "message", T: SIPCodec, repeated: true },
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{ no: 3, name: "encryption", kind: "enum", T: proto3.getEnumType(SIPMediaEncryption), opt: true },
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{ no: 4, name: "media_timeout", kind: "message", T: Duration, opt: true },
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{ no: 5, name: "drop_audio_during_dtmf", kind: "scalar", T: 8 /* ScalarType.BOOL */ },
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],
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);
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@@ -782,6 +802,7 @@ export const CreateSIPParticipantRequest = /*@__PURE__*/ proto3.makeMessageType(
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{ no: 5, name: "dtmf", kind: "scalar", T: 9 /* ScalarType.STRING */ },
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{ no: 6, name: "play_ringtone", kind: "scalar", T: 8 /* ScalarType.BOOL */ },
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{ no: 13, name: "play_dialtone", kind: "scalar", T: 8 /* ScalarType.BOOL */ },
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{ no: 28, name: "dialtone", kind: "enum", T: proto3.getEnumType(SIPDialtone) },
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{ no: 10, name: "hide_phone_number", kind: "scalar", T: 8 /* ScalarType.BOOL */ },
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{ no: 16, name: "headers", kind: "map", K: 9 /* ScalarType.STRING */, V: {kind: "scalar", T: 9 /* ScalarType.STRING */} },
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{ no: 17, name: "include_headers", kind: "enum", T: proto3.getEnumType(SIPHeaderOptions) },
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@@ -819,6 +840,7 @@ export const TransferSIPParticipantRequest = /*@__PURE__*/ proto3.makeMessageTyp
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{ no: 2, name: "room_name", kind: "scalar", T: 9 /* ScalarType.STRING */ },
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{ no: 3, name: "transfer_to", kind: "scalar", T: 9 /* ScalarType.STRING */ },
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{ no: 4, name: "play_dialtone", kind: "scalar", T: 8 /* ScalarType.BOOL */ },
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{ no: 7, name: "dialtone", kind: "enum", T: proto3.getEnumType(SIPDialtone) },
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{ no: 5, name: "headers", kind: "map", K: 9 /* ScalarType.STRING */, V: {kind: "scalar", T: 9 /* ScalarType.STRING */} },
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{ no: 6, name: "ringing_timeout", kind: "message", T: Duration },
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],
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@@ -31,7 +31,7 @@ export declare class WebhookEvent extends Message<WebhookEvent> {
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/**
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* one of room_started, room_finished, participant_joined, participant_left, participant_connection_aborted,
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* track_published, track_unpublished, egress_started, egress_updated, egress_ended,
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* ingress_started, ingress_ended, agent_job_started, agent_job_ended
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* ingress_started, ingress_ended, agent_job_pending, agent_job_started, agent_job_ended
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*
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* @generated from field: string event = 1;
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*/
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package/src/gen/version.js
CHANGED
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// Generated by genversion.
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export const version = '1.53.
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export const version = '1.53.2';
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