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
2. 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:

• Node.js examples | Python examples | Java examples | .NET 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

Region	Base URL
Global (default)	https://calling.api.sinch.com
North America	https://calling-use1.api.sinch.com
Europe	https://calling-euc1.api.sinch.com
Southeast Asia 1	https://calling-apse1.api.sinch.com
Southeast Asia 2	https://calling-apse2.api.sinch.com
South America	https://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)

Action	Description
hangup	Terminate the call
continue	Continue call setup (ACE response to proceed without rerouting)
connectPstn	Connect to PSTN number. Supports amd for Answering Machine Detection
connectMxp	Connect to Sinch SDK (in-app) endpoint
connectConf	Connect to conference room by conferenceId
connectSip	Connect to SIP endpoint
connectStream	Connect to a WebSocket server for real-time audio streaming (closed beta — contact Sinch to enable)
runMenu	IVR menu with DTMF collection (supports enableVoice for speech input)
park	Park (hold) the call with looping prompt

Instructions (multiple per response)

Instruction	Description
playFiles	Play audio files, TTS via #tts[], SSML via #ssml[]
say	Synthesize and play text-to-speech
sendDtmf	Send DTMF tones
setCookie	Persist key-value state across callback events in the session
answer	Answer the call (sends a SIP 200 OK to the INVITE, which starts billing). Required before playing prompts on unanswered calls
startRecording	Begin recording. Supports transcriptionOptions for auto-transcription
stopRecording	Stop an active recording

Callback Events

Event	Trigger	SVAML Response
ICE	Call received by Sinch platform	Yes
ACE	Call answered by callee	Yes
DiCE	Call disconnected	No (fire-and-forget, logging only)
PIE	DTMF/voice input from runMenu	Yes
Notify	Notification (e.g., recording finished)	No

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

Callout Types

Method	Use Case
ttsCallout	Call and play synthesized speech. Supports text or advanced prompts (#tts[], #ssml[], #href[])
conferenceCallout	Call and connect to a conference room
customCallout	Full 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.

Method	Endpoint	Description
POST	/calling/v1/callouts	Place 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/numbers	List numbers and capabilities
POST	/v1/configuration/numbers	Assign numbers to an application
DELETE	/v1/configuration/numbers	Un-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

Script	Description	Example
make_tts_call.cjs	TTS callout	node scripts/make_tts_call.cjs --to +14045005000 --text "Hello"
make_conference_call.cjs	Conference callout	node scripts/make_conference_call.cjs --to +14045005000 --conference-id myRoom
get_call_info.cjs	Get call details	node scripts/get_call_info.cjs --call-id CALL_ID
list_numbers.cjs	List voice numbers	node 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

• Voice API Reference (Markdown)
• Voice API OpenAPI Spec (YAML)
• SVAML Actions | SVAML Instructions
• Callbacks | Callouts
• Authentication | Callback Signing
• Node.js SDK Reference
• Python SDK Reference
• Java SDK Reference
• .NET SDK Reference
• Voice Tutorials
• LLMs.txt (full docs index)