Token导航 LogoToken导航TokenDH.com
开发需要联网github未标认证来源可访问许可证需确认审计异常

sinch-voice-apisinch voice API 文档

Agent Skill

用于辅助 API 设计、接口文档、请求响应结构和服务集成说明。它适合让 Agent 梳理 endpoint、生成 OpenAPI 草稿、检查字段命名、整理错误码或辅助前后端联调。使用时需要确认真实业务语义、鉴权方式、分页和错误处理规则;涉及生成接口文档时,应避免凭空补字段,最好从现有代码、schema 或接口样例中提取事实。

总安装

1,435

周安装

61

GitHub Stars

7

下载量

503
CodexClaudeCursorGemini CLI

安装说明

本站只整理中文说明和来源信息,不托管安装包,也不代用户安装。

GitHub

来源数

2

许可证

unknown

最后核验

2026-05-01

来源状态

来源可访问

安装方式

通过对话安装

复制提示词发给支持本地命令或 Skills 的 AI 助手,先确认命令和权限,再让它执行。

请帮我安装这个 Agent Skill:sinch-voice-api(sinch voice API 文档)
来源仓库:https://github.com/sinch/skills
仓库路径:skills/sinch-voice-api
安装命令:
npx skills add https://github.com/sinch/skills --skill sinch-voice-api
安装前请先检查当前环境是否支持对应 CLI,并向我确认将要执行的命令、安装目录、联网范围和文件读写权限;确认后再执行。

命令行安装

复制命令到本机终端执行。该命令会通过 npx skills 从第三方来源获取 Skill;本站只展示命令,不托管安装包,也不自动执行。

skills.shnpx skills
npx skills add https://github.com/sinch/skills --skill sinch-voice-api

简介

用于辅助 API 设计、接口文档和请求响应结构梳理。

  • 适合在 Codex、Claude、Cursor、Gemini CLI 中生成 OpenAPI 草稿或检查字段命名。
  • 使用时需确认业务语义、鉴权方式和错误处理规则,避免凭空补字段。
  • 建议从现有代码、schema 或接口样例中提取事实,确保信息准确。
  • 安装前请核实权限范围和维护状态,避免触发不必要的网络或文件操作。

SKILL.md

Sinch Voice API

Overview

The Sinch Voice API lets you make, receive, and control voice calls programmatically via REST. It uses SVAML (Sinch Voice Application Markup Language) to define call flows through callback events.

Agent Instructions

Before generating code, you MUST ask the user:

  1. Approach — SDK or direct API calls (curl/fetch/requests)?

- Node.js SDK Reference - Python SDK Reference - Java SDK Reference - .NET SDK Reference

  1. Language — Node.js, Python, Java,.NET, curl?

When generating SDK code, fetch the corresponding SDK reference page for accurate method signatures, or use the bundled examples:

When generating direct API calls, use the Voice API Reference (Markdown) for request/response schemas.

Getting Started

Authentication

See the sinch-authentication skill. The Voice API uses Application Key + Application Secret (not project-level OAuth2).

  • Basic Auth: Authorization: Basic base64(APPLICATION_KEY:APPLICATION_SECRET)
  • Signed Requests (production): HMAC-SHA256 signing. See Authentication Guide.

Base URLs

RegionBase URL
Global (default)https://calling.api.sinch.com
North Americahttps://calling-use1.api.sinch.com
Europehttps://calling-euc1.api.sinch.com
Southeast Asia 1https://calling-apse1.api.sinch.com
Southeast Asia 2https://calling-apse2.api.sinch.com
South Americahttps://calling-sae1.api.sinch.com

Configuration endpoints (numbers, callbacks) use: https://callingapi.sinch.com

SDK Installation

See sinch-sdks for installation and client initialization across all languages.

First API Call: TTS Callout

curl -X POST \
  "https://calling.api.sinch.com/calling/v1/callouts" \
  -u "{APPLICATION_KEY}:{APPLICATION_SECRET}" \
  -H "Content-Type: application/json" \
  -d '{
    "method": "ttsCallout",
    "ttsCallout": {
      "destination": { "type": "number", "endpoint": "+14045005000" },
      "cli": "+14045001000",
      "locale": "en-US",
      "text": "Hello! This is a test call from Sinch."
    }
  }'

Node.js SDK:

import { SinchClient } from "@sinch/sdk-core";

const sinch = new SinchClient({
  applicationKey: "{APPLICATION_KEY}",
  applicationSecret: "{APPLICATION_SECRET}",
});

const response = await sinch.voice.callouts.tts({
  ttsCalloutRequestBody: {
    destination: { type: "number", endpoint: "+14045005000" },
    cli: "+14045001000",
    locale: "en-US",
    text: "Hello! This is a test call from Sinch.",
  },
});
console.log("Call ID:", response.callId);

For more examples, see Callouts Reference or bundled examples.

Key Concepts

SVAML (Sinch Voice Application Markup Language)

SVAML controls call flow. Every SVAML response has:

  • instructions (array): Multiple tasks — play audio, record, set cookies
  • action (object): Exactly ONE routing/control action

Full reference: SVAML Actions | SVAML Instructions | Bundled SVAML Reference

Actions (one per response)

ActionDescription
hangupTerminate the call
continueContinue call setup (ACE response to proceed without rerouting)
connectPstnConnect to PSTN number. Supports amd for Answering Machine Detection
connectMxpConnect to Sinch SDK (in-app) endpoint
connectConfConnect to conference room by conferenceId
connectSipConnect to SIP endpoint
connectStreamConnect to a WebSocket server for real-time audio streaming (closed beta — contact Sinch to enable)
runMenuIVR menu with DTMF collection (supports enableVoice for speech input)
parkPark (hold) the call with looping prompt

Instructions (multiple per response)

InstructionDescription
playFilesPlay audio files, TTS via #tts[], SSML via #ssml[]
saySynthesize and play text-to-speech
sendDtmfSend DTMF tones
setCookiePersist key-value state across callback events in the session
answerAnswer the call (sends a SIP 200 OK to the INVITE, which starts billing). Required before playing prompts on unanswered calls
startRecordingBegin recording. Supports transcriptionOptions for auto-transcription
stopRecordingStop an active recording

Callback Events

EventTriggerSVAML Response
ICECall received by Sinch platformYes
ACECall answered by calleeYes
DiCECall disconnectedNo (fire-and-forget, logging only)
PIEDTMF/voice input from runMenuYes
NotifyNotification (e.g., recording finished)No

See Callbacks Reference for event schemas, or bundled callbacks reference for full field tables and JSON examples.

Callout Types

MethodUse Case
ttsCalloutCall and play synthesized speech. Supports text or advanced prompts (#tts[], #ssml[], #href[])
conferenceCalloutCall and connect to a conference room
customCalloutFull SVAML control with inline ICE/ACE/PIE

Callout flags: enableAce (default false), enableDice (default false), enablePie (default false) control which callbacks fire.

REST Endpoints

Paths starting with /calling/v1/ use the regional base URL from the table above. Paths starting with /v1/configuration/ use https://callingapi.sinch.com.

MethodEndpointDescription
POST/calling/v1/calloutsPlace a callout (TTS, conference, or custom)
PATCH/calling/v1/calls/id/{callId}Update in-progress call with SVAML (PSTN/SIP only)
GET/calling/v1/calls/id/{callId}Get call info
PATCH/calling/v1/calls/id/{callId}/leg/{callLeg}Manage a call leg (PlayFiles/Say only)
GET/calling/v1/conferences/id/{conferenceId}Get conference info
DELETE/calling/v1/conferences/id/{conferenceId}Kick all participants
PATCH/calling/v1/conferences/id/{conferenceId}/{callId}Mute/unmute/hold participant
DELETE/calling/v1/conferences/id/{conferenceId}/{callId}Kick specific participant
GET/v1/configuration/numbersList numbers and capabilities
POST/v1/configuration/numbersAssign numbers to an application
DELETE/v1/configuration/numbersUn-assign a number
GET/POST/v1/configuration/callbacks/applications/{applicationkey}Get/update callback URLs

Common Patterns

IVR Menu (SVAML)

{
  "instructions": [
    { "name": "setCookie", "key": "step", "value": "ivr" }
  ],
  "action": {
    "name": "runMenu",
    "mainMenu": "main",
    "menus": [{
      "id": "main",
      "mainPrompt": "#tts[Press 1 for sales or 2 for support.]",
      "options": [
        { "dtmf": 1, "action": "return(sales)" },
        { "dtmf": 2, "action": "return(support)" }
      ]
    }]
  }
}

Conference with Recording

{
  "instructions": [
    { "name": "startRecording", "options": { "notificationEvents": true } }
  ],
  "action": {
    "name": "connectConf",
    "conferenceId": "myRoom",
    "moh": "ring"
  }
}

PSTN Forward with AMD

{
  "action": {
    "name": "connectPstn",
    "number": "+14045009000",
    "cli": "+14045001000",
    "maxDuration": 3600,
    "amd": { "enabled": true }
  }
}

Executable Scripts

Bundled Node.js scripts (no external dependencies, uses Basic Auth):

export SINCH_APPLICATION_KEY="{APPLICATION_KEY}"
export SINCH_APPLICATION_SECRET="{APPLICATION_SECRET}"
export SINCH_VOICE_REGION="global"  # optional
ScriptDescriptionExample
make_tts_call.cjsTTS calloutnode scripts/make_tts_call.cjs --to +14045005000 --text "Hello"
make_conference_call.cjsConference calloutnode scripts/make_conference_call.cjs --to +14045005000 --conference-id myRoom
get_call_info.cjsGet call detailsnode scripts/get_call_info.cjs --call-id CALL_ID
list_numbers.cjsList voice numbersnode scripts/list_numbers.cjs

Gotchas and Best Practices

  1. Callback URL must be publicly accessible. Use ngrok for local dev. Configure in Dashboard under Voice app settings.
  2. ONE action per SVAML response. Multiple instructions are fine. Chain callbacks for sequential actions (ICE → ACE → PIE).
  3. ACE not sent for in-app destinations. ACE is not issued when destination type is username, only for PSTN/SIP. Setting enableAce: true has no effect for in-app destinations.
  4. DiCE is fire-and-forget. Informational only. No SVAML response expected. Use for logging/cleanup.
  5. Regional endpoints matter. Wrong region increases latency. Conference rooms have regional scope — force all participants to the same region for cross-region conferences.
  6. Instruction ordering matters. Array order = execution order. Place answer before playFiles; place startRecording before the connecting action.
  7. Max call duration: 14400 seconds (4 hours). Set maxDuration on connectPstn/connectSip for shorter limits.
  8. Validate callback signatures in production. HMAC-SHA256 signature in Authorization header. See Callback Signing.
  9. setCookie for state. Carries key-value pairs across ICE, ACE, PIE, DiCE within a call session.
  10. connectMxp does not support recording. startRecording/stopRecording instructions are ignored with connectMxp.
  11. runMenu defaults. barge: true (input accepted during prompt). timeoutMills: 5000 ms.
  12. AMD on connectPstn. amd: {enabled: true, async: true/false} for answering machine detection.
  13. startRecording transcription. transcriptionOptions: {enabled: true, locale: "en-US"} for auto-transcription.
  14. Conference DTMF options. conferenceDtmfOptions on conferenceCallout/connectConf with modes: ignore (default), forward, detect (sends PIE).
  15. cli is required for TTS callouts to connect. The API accepts a TTS callout without a cli parameter and returns a call ID, but the call will never reach the destination. The cli is the number displayed as the incoming caller — use your verified number or your Dashboard-assigned number, in E.164 format (e.g., "+14151112223333"). To test, register on the Sinch Dashboard and use the free number assigned to your app. See Assign your number.

Links

适合场景

01

用户想查找某类 Agent Skill 时

02

需要根据任务场景推荐可安装能力包时

03

需要对比不同来源的安装命令和来源信息时

能力概览

能力 1

按任务关键词查找相关 Skills

能力 2

展示可复制的安装命令

能力 3

保留来源站点、仓库和原始说明,方便继续核验

能力 4

展示第三方安全扫描或审计结果

安装后应在对应宿主中按原始 README 的触发条件使用;具体调用方式请以来源页面和 README 为准。

平台分布

Codex

33.43%
按下载量换算168

Claude

30.19%
按下载量换算152

Cursor

19.16%
按下载量换算96

Gemini CLI

9.79%
按下载量换算49

安全审计

Gen Agent Trust Hub

通过

Socket

通过

Snyk

未通过

权限和风险

需要联网

该 Skill 可能需要联网访问来源站点、仓库或外部 API;具体网络访问范围需要结合源码和 README 复核。

安装前确认

本站仅展示第三方公开信息,不托管安装包,不提供自动安装或运行环境。安装前应自行审查源码、依赖和命令行为。来源安全扫描存在 warning/failed 结果,不能写成本站确认安全。当前只有一个来源,正式发布前建议补源仓库或其他目录站核验。

来源信息

继续浏览同类 Skills