@depup/openai 6.31.0-depup.0

package/resources/realtime/calls.d.mts

@@ -0,0 +1,158 @@
+import { APIResource } from "../../core/resource.mjs";
+import * as RealtimeAPI from "./realtime.mjs";
+import * as ResponsesAPI from "../responses/responses.mjs";
+import { APIPromise } from "../../core/api-promise.mjs";
+import { RequestOptions } from "../../internal/request-options.mjs";
+export declare class Calls extends APIResource {
+    /**
+     * Accept an incoming SIP call and configure the realtime session that will handle
+     * it.
+     *
+     * @example
+     * ```ts
+     * await client.realtime.calls.accept('call_id', {
+     *   type: 'realtime',
+     * });
+     * ```
+     */
+    accept(callID: string, body: CallAcceptParams, options?: RequestOptions): APIPromise<void>;
+    /**
+     * End an active Realtime API call, whether it was initiated over SIP or WebRTC.
+     *
+     * @example
+     * ```ts
+     * await client.realtime.calls.hangup('call_id');
+     * ```
+     */
+    hangup(callID: string, options?: RequestOptions): APIPromise<void>;
+    /**
+     * Transfer an active SIP call to a new destination using the SIP REFER verb.
+     *
+     * @example
+     * ```ts
+     * await client.realtime.calls.refer('call_id', {
+     *   target_uri: 'tel:+14155550123',
+     * });
+     * ```
+     */
+    refer(callID: string, body: CallReferParams, options?: RequestOptions): APIPromise<void>;
+    /**
+     * Decline an incoming SIP call by returning a SIP status code to the caller.
+     *
+     * @example
+     * ```ts
+     * await client.realtime.calls.reject('call_id');
+     * ```
+     */
+    reject(callID: string, body?: CallRejectParams | null | undefined, options?: RequestOptions): APIPromise<void>;
+}
+export interface CallAcceptParams {
+    /**
+     * The type of session to create. Always `realtime` for the Realtime API.
+     */
+    type: 'realtime';
+    /**
+     * Configuration for input and output audio.
+     */
+    audio?: RealtimeAPI.RealtimeAudioConfig;
+    /**
+     * Additional fields to include in server outputs.
+     *
+     * `item.input_audio_transcription.logprobs`: Include logprobs for input audio
+     * transcription.
+     */
+    include?: Array<'item.input_audio_transcription.logprobs'>;
+    /**
+     * The default system instructions (i.e. system message) prepended to model calls.
+     * This field allows the client to guide the model on desired responses. The model
+     * can be instructed on response content and format, (e.g. "be extremely succinct",
+     * "act friendly", "here are examples of good responses") and on audio behavior
+     * (e.g. "talk quickly", "inject emotion into your voice", "laugh frequently"). The
+     * instructions are not guaranteed to be followed by the model, but they provide
+     * guidance to the model on the desired behavior.
+     *
+     * Note that the server sets default instructions which will be used if this field
+     * is not set and are visible in the `session.created` event at the start of the
+     * session.
+     */
+    instructions?: string;
+    /**
+     * Maximum number of output tokens for a single assistant response, inclusive of
+     * tool calls. Provide an integer between 1 and 4096 to limit output tokens, or
+     * `inf` for the maximum available tokens for a given model. Defaults to `inf`.
+     */
+    max_output_tokens?: number | 'inf';
+    /**
+     * The Realtime model used for this session.
+     */
+    model?: (string & {}) | 'gpt-realtime' | 'gpt-realtime-1.5' | 'gpt-realtime-2025-08-28' | 'gpt-4o-realtime-preview' | 'gpt-4o-realtime-preview-2024-10-01' | 'gpt-4o-realtime-preview-2024-12-17' | 'gpt-4o-realtime-preview-2025-06-03' | 'gpt-4o-mini-realtime-preview' | 'gpt-4o-mini-realtime-preview-2024-12-17' | 'gpt-realtime-mini' | 'gpt-realtime-mini-2025-10-06' | 'gpt-realtime-mini-2025-12-15' | 'gpt-audio-1.5' | 'gpt-audio-mini' | 'gpt-audio-mini-2025-10-06' | 'gpt-audio-mini-2025-12-15';
+    /**
+     * The set of modalities the model can respond with. It defaults to `["audio"]`,
+     * indicating that the model will respond with audio plus a transcript. `["text"]`
+     * can be used to make the model respond with text only. It is not possible to
+     * request both `text` and `audio` at the same time.
+     */
+    output_modalities?: Array<'text' | 'audio'>;
+    /**
+     * Reference to a prompt template and its variables.
+     * [Learn more](https://platform.openai.com/docs/guides/text?api-mode=responses#reusable-prompts).
+     */
+    prompt?: ResponsesAPI.ResponsePrompt | null;
+    /**
+     * How the model chooses tools. Provide one of the string modes or force a specific
+     * function/MCP tool.
+     */
+    tool_choice?: RealtimeAPI.RealtimeToolChoiceConfig;
+    /**
+     * Tools available to the model.
+     */
+    tools?: RealtimeAPI.RealtimeToolsConfig;
+    /**
+     * Realtime API can write session traces to the
+     * [Traces Dashboard](/logs?api=traces). Set to null to disable tracing. Once
+     * tracing is enabled for a session, the configuration cannot be modified.
+     *
+     * `auto` will create a trace for the session with default values for the workflow
+     * name, group id, and metadata.
+     */
+    tracing?: RealtimeAPI.RealtimeTracingConfig | null;
+    /**
+     * When the number of tokens in a conversation exceeds the model's input token
+     * limit, the conversation be truncated, meaning messages (starting from the
+     * oldest) will not be included in the model's context. A 32k context model with
+     * 4,096 max output tokens can only include 28,224 tokens in the context before
+     * truncation occurs.
+     *
+     * Clients can configure truncation behavior to truncate with a lower max token
+     * limit, which is an effective way to control token usage and cost.
+     *
+     * Truncation will reduce the number of cached tokens on the next turn (busting the
+     * cache), since messages are dropped from the beginning of the context. However,
+     * clients can also configure truncation to retain messages up to a fraction of the
+     * maximum context size, which will reduce the need for future truncations and thus
+     * improve the cache rate.
+     *
+     * Truncation can be disabled entirely, which means the server will never truncate
+     * but would instead return an error if the conversation exceeds the model's input
+     * token limit.
+     */
+    truncation?: RealtimeAPI.RealtimeTruncation;
+}
+export interface CallReferParams {
+    /**
+     * URI that should appear in the SIP Refer-To header. Supports values like
+     * `tel:+14155550123` or `sip:agent@example.com`.
+     */
+    target_uri: string;
+}
+export interface CallRejectParams {
+    /**
+     * SIP response code to send back to the caller. Defaults to `603` (Decline) when
+     * omitted.
+     */
+    status_code?: number;
+}
+export declare namespace Calls {
+    export { type CallAcceptParams as CallAcceptParams, type CallReferParams as CallReferParams, type CallRejectParams as CallRejectParams, };
+}
+//# sourceMappingURL=calls.d.mts.map

package/resources/realtime/calls.d.mts.map

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package/resources/realtime/calls.d.ts

@@ -0,0 +1,158 @@
+import { APIResource } from "../../core/resource.js";
+import * as RealtimeAPI from "./realtime.js";
+import * as ResponsesAPI from "../responses/responses.js";
+import { APIPromise } from "../../core/api-promise.js";
+import { RequestOptions } from "../../internal/request-options.js";
+export declare class Calls extends APIResource {
+    /**
+     * Accept an incoming SIP call and configure the realtime session that will handle
+     * it.
+     *
+     * @example
+     * ```ts
+     * await client.realtime.calls.accept('call_id', {
+     *   type: 'realtime',
+     * });
+     * ```
+     */
+    accept(callID: string, body: CallAcceptParams, options?: RequestOptions): APIPromise<void>;
+    /**
+     * End an active Realtime API call, whether it was initiated over SIP or WebRTC.
+     *
+     * @example
+     * ```ts
+     * await client.realtime.calls.hangup('call_id');
+     * ```
+     */
+    hangup(callID: string, options?: RequestOptions): APIPromise<void>;
+    /**
+     * Transfer an active SIP call to a new destination using the SIP REFER verb.
+     *
+     * @example
+     * ```ts
+     * await client.realtime.calls.refer('call_id', {
+     *   target_uri: 'tel:+14155550123',
+     * });
+     * ```
+     */
+    refer(callID: string, body: CallReferParams, options?: RequestOptions): APIPromise<void>;
+    /**
+     * Decline an incoming SIP call by returning a SIP status code to the caller.
+     *
+     * @example
+     * ```ts
+     * await client.realtime.calls.reject('call_id');
+     * ```
+     */
+    reject(callID: string, body?: CallRejectParams | null | undefined, options?: RequestOptions): APIPromise<void>;
+}
+export interface CallAcceptParams {
+    /**
+     * The type of session to create. Always `realtime` for the Realtime API.
+     */
+    type: 'realtime';
+    /**
+     * Configuration for input and output audio.
+     */
+    audio?: RealtimeAPI.RealtimeAudioConfig;
+    /**
+     * Additional fields to include in server outputs.
+     *
+     * `item.input_audio_transcription.logprobs`: Include logprobs for input audio
+     * transcription.
+     */
+    include?: Array<'item.input_audio_transcription.logprobs'>;
+    /**
+     * The default system instructions (i.e. system message) prepended to model calls.
+     * This field allows the client to guide the model on desired responses. The model
+     * can be instructed on response content and format, (e.g. "be extremely succinct",
+     * "act friendly", "here are examples of good responses") and on audio behavior
+     * (e.g. "talk quickly", "inject emotion into your voice", "laugh frequently"). The
+     * instructions are not guaranteed to be followed by the model, but they provide
+     * guidance to the model on the desired behavior.
+     *
+     * Note that the server sets default instructions which will be used if this field
+     * is not set and are visible in the `session.created` event at the start of the
+     * session.
+     */
+    instructions?: string;
+    /**
+     * Maximum number of output tokens for a single assistant response, inclusive of
+     * tool calls. Provide an integer between 1 and 4096 to limit output tokens, or
+     * `inf` for the maximum available tokens for a given model. Defaults to `inf`.
+     */
+    max_output_tokens?: number | 'inf';
+    /**
+     * The Realtime model used for this session.
+     */
+    model?: (string & {}) | 'gpt-realtime' | 'gpt-realtime-1.5' | 'gpt-realtime-2025-08-28' | 'gpt-4o-realtime-preview' | 'gpt-4o-realtime-preview-2024-10-01' | 'gpt-4o-realtime-preview-2024-12-17' | 'gpt-4o-realtime-preview-2025-06-03' | 'gpt-4o-mini-realtime-preview' | 'gpt-4o-mini-realtime-preview-2024-12-17' | 'gpt-realtime-mini' | 'gpt-realtime-mini-2025-10-06' | 'gpt-realtime-mini-2025-12-15' | 'gpt-audio-1.5' | 'gpt-audio-mini' | 'gpt-audio-mini-2025-10-06' | 'gpt-audio-mini-2025-12-15';
+    /**
+     * The set of modalities the model can respond with. It defaults to `["audio"]`,
+     * indicating that the model will respond with audio plus a transcript. `["text"]`
+     * can be used to make the model respond with text only. It is not possible to
+     * request both `text` and `audio` at the same time.
+     */
+    output_modalities?: Array<'text' | 'audio'>;
+    /**
+     * Reference to a prompt template and its variables.
+     * [Learn more](https://platform.openai.com/docs/guides/text?api-mode=responses#reusable-prompts).
+     */
+    prompt?: ResponsesAPI.ResponsePrompt | null;
+    /**
+     * How the model chooses tools. Provide one of the string modes or force a specific
+     * function/MCP tool.
+     */
+    tool_choice?: RealtimeAPI.RealtimeToolChoiceConfig;
+    /**
+     * Tools available to the model.
+     */
+    tools?: RealtimeAPI.RealtimeToolsConfig;
+    /**
+     * Realtime API can write session traces to the
+     * [Traces Dashboard](/logs?api=traces). Set to null to disable tracing. Once
+     * tracing is enabled for a session, the configuration cannot be modified.
+     *
+     * `auto` will create a trace for the session with default values for the workflow
+     * name, group id, and metadata.
+     */
+    tracing?: RealtimeAPI.RealtimeTracingConfig | null;
+    /**
+     * When the number of tokens in a conversation exceeds the model's input token
+     * limit, the conversation be truncated, meaning messages (starting from the
+     * oldest) will not be included in the model's context. A 32k context model with
+     * 4,096 max output tokens can only include 28,224 tokens in the context before
+     * truncation occurs.
+     *
+     * Clients can configure truncation behavior to truncate with a lower max token
+     * limit, which is an effective way to control token usage and cost.
+     *
+     * Truncation will reduce the number of cached tokens on the next turn (busting the
+     * cache), since messages are dropped from the beginning of the context. However,
+     * clients can also configure truncation to retain messages up to a fraction of the
+     * maximum context size, which will reduce the need for future truncations and thus
+     * improve the cache rate.
+     *
+     * Truncation can be disabled entirely, which means the server will never truncate
+     * but would instead return an error if the conversation exceeds the model's input
+     * token limit.
+     */
+    truncation?: RealtimeAPI.RealtimeTruncation;
+}
+export interface CallReferParams {
+    /**
+     * URI that should appear in the SIP Refer-To header. Supports values like
+     * `tel:+14155550123` or `sip:agent@example.com`.
+     */
+    target_uri: string;
+}
+export interface CallRejectParams {
+    /**
+     * SIP response code to send back to the caller. Defaults to `603` (Decline) when
+     * omitted.
+     */
+    status_code?: number;
+}
+export declare namespace Calls {
+    export { type CallAcceptParams as CallAcceptParams, type CallReferParams as CallReferParams, type CallRejectParams as CallRejectParams, };
+}
+//# sourceMappingURL=calls.d.ts.map

package/resources/realtime/calls.d.ts.map

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package/resources/realtime/calls.js

@@ -0,0 +1,75 @@
+"use strict";
+// File generated from our OpenAPI spec by Stainless. See CONTRIBUTING.md for details.
+Object.defineProperty(exports, "__esModule", { value: true });
+exports.Calls = void 0;
+const resource_1 = require("../../core/resource.js");
+const headers_1 = require("../../internal/headers.js");
+const path_1 = require("../../internal/utils/path.js");
+class Calls extends resource_1.APIResource {
+    /**
+     * Accept an incoming SIP call and configure the realtime session that will handle
+     * it.
+     *
+     * @example
+     * ```ts
+     * await client.realtime.calls.accept('call_id', {
+     *   type: 'realtime',
+     * });
+     * ```
+     */
+    accept(callID, body, options) {
+        return this._client.post((0, path_1.path) `/realtime/calls/${callID}/accept`, {
+            body,
+            ...options,
+            headers: (0, headers_1.buildHeaders)([{ Accept: '*/*' }, options?.headers]),
+        });
+    }
+    /**
+     * End an active Realtime API call, whether it was initiated over SIP or WebRTC.
+     *
+     * @example
+     * ```ts
+     * await client.realtime.calls.hangup('call_id');
+     * ```
+     */
+    hangup(callID, options) {
+        return this._client.post((0, path_1.path) `/realtime/calls/${callID}/hangup`, {
+            ...options,
+            headers: (0, headers_1.buildHeaders)([{ Accept: '*/*' }, options?.headers]),
+        });
+    }
+    /**
+     * Transfer an active SIP call to a new destination using the SIP REFER verb.
+     *
+     * @example
+     * ```ts
+     * await client.realtime.calls.refer('call_id', {
+     *   target_uri: 'tel:+14155550123',
+     * });
+     * ```
+     */
+    refer(callID, body, options) {
+        return this._client.post((0, path_1.path) `/realtime/calls/${callID}/refer`, {
+            body,
+            ...options,
+            headers: (0, headers_1.buildHeaders)([{ Accept: '*/*' }, options?.headers]),
+        });
+    }
+    /**
+     * Decline an incoming SIP call by returning a SIP status code to the caller.
+     *
+     * @example
+     * ```ts
+     * await client.realtime.calls.reject('call_id');
+     * ```
+     */
+    reject(callID, body = {}, options) {
+        return this._client.post((0, path_1.path) `/realtime/calls/${callID}/reject`, {
+            body,
+            ...options,
+            headers: (0, headers_1.buildHeaders)([{ Accept: '*/*' }, options?.headers]),
+        });
+    }
+}
+exports.Calls = Calls;
+//# sourceMappingURL=calls.js.map

package/resources/realtime/calls.js.map

@@ -0,0 +1 @@
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package/resources/realtime/calls.mjs

@@ -0,0 +1,71 @@
+// File generated from our OpenAPI spec by Stainless. See CONTRIBUTING.md for details.
+import { APIResource } from "../../core/resource.mjs";
+import { buildHeaders } from "../../internal/headers.mjs";
+import { path } from "../../internal/utils/path.mjs";
+export class Calls extends APIResource {
+    /**
+     * Accept an incoming SIP call and configure the realtime session that will handle
+     * it.
+     *
+     * @example
+     * ```ts
+     * await client.realtime.calls.accept('call_id', {
+     *   type: 'realtime',
+     * });
+     * ```
+     */
+    accept(callID, body, options) {
+        return this._client.post(path `/realtime/calls/${callID}/accept`, {
+            body,
+            ...options,
+            headers: buildHeaders([{ Accept: '*/*' }, options?.headers]),
+        });
+    }
+    /**
+     * End an active Realtime API call, whether it was initiated over SIP or WebRTC.
+     *
+     * @example
+     * ```ts
+     * await client.realtime.calls.hangup('call_id');
+     * ```
+     */
+    hangup(callID, options) {
+        return this._client.post(path `/realtime/calls/${callID}/hangup`, {
+            ...options,
+            headers: buildHeaders([{ Accept: '*/*' }, options?.headers]),
+        });
+    }
+    /**
+     * Transfer an active SIP call to a new destination using the SIP REFER verb.
+     *
+     * @example
+     * ```ts
+     * await client.realtime.calls.refer('call_id', {
+     *   target_uri: 'tel:+14155550123',
+     * });
+     * ```
+     */
+    refer(callID, body, options) {
+        return this._client.post(path `/realtime/calls/${callID}/refer`, {
+            body,
+            ...options,
+            headers: buildHeaders([{ Accept: '*/*' }, options?.headers]),
+        });
+    }
+    /**
+     * Decline an incoming SIP call by returning a SIP status code to the caller.
+     *
+     * @example
+     * ```ts
+     * await client.realtime.calls.reject('call_id');
+     * ```
+     */
+    reject(callID, body = {}, options) {
+        return this._client.post(path `/realtime/calls/${callID}/reject`, {
+            body,
+            ...options,
+            headers: buildHeaders([{ Accept: '*/*' }, options?.headers]),
+        });
+    }
+}
+//# sourceMappingURL=calls.mjs.map

package/resources/realtime/calls.mjs.map

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