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