WebSocket API reference - SLNG API Documentation

WebSocket connections give you sub-100ms latency and bidirectional streaming, ideal for voice agents, live transcription, and any use case where audio flows continuously. If you don’t need real-time streaming, see HTTP vs. WebSocket instead. Prerequisites:

An SLNG API key
Basic familiarity with WebSockets (open, send, receive, close)

Protocol overview

This page documents the WebSocket protocol used by SLNG-hosted models and bridges. Third-party providers (Deepgram, KugelAudio, Sarvam) may expose their own native WebSocket formats — check the per-model API reference in the sidebar for provider-specific details.

Supported encodings, sample rates, and optional fields vary by model. For model-specific parameters, see the Text-to-Speech and Speech-to-Text tabs in the sidebar.

TTS WebSocket Protocol

For the full per-model parameter list and response schema, see the model’s page in the Text-to-Speech sidebar.

Connection

WSS wss://api.slng.ai/v1/tts/{provider}/{model}:{variant}

Message Flow

Browse all available TTS models and endpoints on the Text-to-Speech models page.

Client → Server Messages

Initialize Session

Initialize a session with model and voice configuration before sending text.

{
  "type": "init",
  "model": "aura:2",
  "voice": "aura-2-thalia-en",
  "config": {
    "sample_rate": 24000,
    "encoding": "linear16",  // 16-bit signed PCM — the most common raw audio format
    "language": "en",
    "speed": 1.0
  }
}

Parameters:

Field	Type	Required	Description
`type`	`"init"`	Yes	Message type
`model`	`string`	Yes	Model identifier
`voice`	`string`	No	Voice identifier
`config`	`object`	No	Session configuration
`config.sample_rate`	`number`	No	Audio sample rate in Hz
`config.encoding`	`"linear16"` \| `"mp3"` \| `"opus"`	No	Audio encoding format
`config.language`	`string`	No	Language code
`config.speed`	`number`	No	Speech speed multiplier

Send Text for Synthesis

Send text to synthesize into audio. Set flush: true to finalize the current segment immediately instead of waiting for more text.

{
  "type": "text",
  "text": "Hello from SLNG.",
  "flush": false
}

Field	Type	Required	Description
`type`	`"text"`	Yes	Message type
`text`	`string`	Yes	Text to synthesize
`flush`	`boolean`	No	Whether to flush remaining audio immediately

Flush Buffer

Force any buffered text/audio to be finalized and delivered.

{
  "type": "flush"
}

Clear Buffer

Clear any queued text/audio from the current session.

{
  "type": "clear"
}

Close Session

Close the current session and stop any further audio generation.

{
  "type": "close"
}

Server → Client Messages

Session Ready

Indicates the session is ready to receive messages.

{
  "type": "ready",
  "session_id": "session_123"
}

Audio Chunk

Chunk of base64-encoded audio data.

{
  "type": "audio_chunk",
  "data": "SGVsbG8gV29ybGQ=",
  "sequence": 1
}

Field	Type	Required	Description
`type`	`"audio_chunk"`	Yes	Message type
`data`	`string`	Yes	Base64-encoded audio data
`sequence`	`integer`	No	Sequence number for ordering

Audio may also arrive as raw binary WebSocket frames instead of base64 JSON. Binary frames have lower overhead.

Segment Start

Signals the start of a synthesized segment.

{
  "type": "segment_start",
  "segment_id": "seg_1"
}

Segment End

Signals the end of a synthesized segment.

{
  "type": "segment_end",
  "segment_id": "seg_1"
}

Flushed

Acknowledges that buffered output was flushed.

{
  "type": "flushed"
}

Cleared

Acknowledges that queued output was cleared.

{
  "type": "cleared"
}

Audio End

Signals the end of audio generation.

{
  "type": "audio_end",
  "duration": 0.12
}

Field	Type	Required	Description
`type`	`"audio_end"`	Yes	Message type
`duration`	`number`	No	Audio duration

Error

Indicates an error occurred during synthesis.

{
  "type": "error",
  "code": "provider_error",
  "message": "Upstream error"
}

Field	Type	Required	Description
`type`	`"error"`	Yes	Message type
`code`	`string`	Yes	Error code
`message`	`string`	Yes	Human-readable error description

Common Error Codes:

auth_error: Invalid or missing API key
config_error: Invalid configuration
rate_limit: Too many requests
provider_error: Upstream provider error

STT WebSocket Protocol

For the full per-model parameter list and response schema, see the model’s page in the Speech-to-Text sidebar.

Connection

WSS wss://api.slng.ai/v1/stt/{provider}/{model}:{variant}

Message Flow

Browse all available STT models and endpoints on the Speech-to-Text models page.

Client → Server Messages

Initialize Session

Initialize a session with recognition configuration before streaming audio.

{
  "type": "init",
  "config": {
    "language": "en",
    "sample_rate": 16000,
    "encoding": "linear16",
    "enable_vad": true,
    "enable_diarization": false,
    "enable_word_timestamps": true,
    "enable_partials": true
  }
}

Parameters:

Field	Type	Required	Description
`type`	`"init"`	Yes	Message type
`config`	`object`	No	Recognition configuration
`config.language`	`string`	No	Language code for recognition
`config.sample_rate`	`number`	No	Audio sample rate in Hz
`config.encoding`	`"linear16"` \| `"mp3"` \| `"opus"`	No	Audio encoding format
`config.enable_vad`	`boolean`	No	Enable voice activity detection
`config.enable_diarization`	`boolean`	No	Enable speaker diarization
`config.enable_word_timestamps`	`boolean`	No	Include word-level timestamps
`config.enable_partials`	`boolean`	No	Enable partial/interim transcripts

Send Audio Data

Stream an audio frame to be transcribed. After initialization, send audio in one of two formats:

Binary frames are recommended over base64-encoded JSON for lower overhead.

Binary frames — send raw PCM audio samples directly as binary WebSocket frames:

ws.send(audioBuffer); // ArrayBuffer or Uint8Array

JSON messages with base64-encoded data:

{
  "type": "audio",
  "data": "SGVsbG8gV29ybGQ="
}

Field	Type	Required	Description
`type`	`"audio"`	Yes	Message type
`data`	`string`	Yes	Base64-encoded audio data

Finalize Transcription

Force the server to finalize any buffered audio and return results. The connection stays open so you can continue streaming.

{
  "type": "finalize"
}

Close Stream

Signal that no more audio will be sent. The server processes any remaining audio, sends final results, then closes the connection.

{
  "type": "close"
}

Use finalize when you want to flush results mid-session (e.g., between utterances). Use close when you are done and want to end the session.

Keep-Alive

Send periodically during silence to prevent idle disconnection.

{
  "type": "keepalive"
}

Server → Client Messages

Session Ready

Indicates the session is ready to receive audio.

{
  "type": "ready",
  "session_id": "session_123"
}

Partial Transcript

Interim transcription result.

{
  "type": "partial_transcript",
  "transcript": "Hello",
  "confidence": 0.91
}

Field	Type	Required	Description
`type`	`"partial_transcript"`	Yes	Message type
`transcript`	`string`	Yes	Transcribed text so far
`confidence`	`number`	No	Confidence score (0-1)

Final Transcript

Final transcription result with optional metadata.

{
  "type": "final_transcript",
  "transcript": "Hello world",
  "confidence": 0.97,
  "language": "en",
  "duration": 2.5
}

Field	Type	Required	Description
`type`	`"final_transcript"`	Yes	Message type
`transcript`	`string`	Yes	Complete transcribed text
`confidence`	`number`	No	Overall confidence score (0-1)
`language`	`string`	No	Detected or specified language code
`duration`	`number`	No	Audio duration

Error

Indicates an error occurred during recognition.

{
  "type": "error",
  "code": "provider_error",
  "message": "Upstream error"
}

Field	Type	Required	Description
`type`	`"error"`	Yes	Message type
`code`	`string`	Yes	Error code
`message`	`string`	Yes	Human-readable error description

Next Steps

Integration guide

Best practices and troubleshooting

TTS examples

JavaScript and Python code for real-time TTS

STT examples

JavaScript and Python code for real-time STT

Protocol comparison

HTTP vs. WebSocket — when to use each

Getting Started

SDKs & Tools

Dashboard

Guides

Voices

Examples

Changelog

Documentation Index

​Protocol overview

​TTS WebSocket Protocol

​Connection

​Message Flow

​Client → Server Messages

​Initialize Session

​Send Text for Synthesis

​Flush Buffer

​Clear Buffer

​Close Session

​Server → Client Messages

​Session Ready

​Audio Chunk

​Segment Start

​Segment End

​Flushed

​Cleared

​Audio End

​Error

​STT WebSocket Protocol

​Connection

​Message Flow

​Client → Server Messages

​Initialize Session

​Send Audio Data

​Finalize Transcription

​Close Stream

​Keep-Alive

​Server → Client Messages

​Session Ready

​Partial Transcript

​Final Transcript

​Error

​Next Steps

Integration guide

TTS examples

STT examples

Protocol comparison

Protocol overview

TTS WebSocket Protocol

Connection

Message Flow

Client → Server Messages

Initialize Session

Send Text for Synthesis

Flush Buffer

Clear Buffer

Close Session

Server → Client Messages

Session Ready

Audio Chunk

Segment Start

Segment End

Flushed

Cleared

Audio End

Error

STT WebSocket Protocol

Connection

Message Flow

Client → Server Messages

Initialize Session

Send Audio Data

Finalize Transcription

Close Stream

Keep-Alive

Server → Client Messages

Session Ready

Partial Transcript

Final Transcript

Error

Next Steps