Speech to Text (Beta) | Murf API

Murf Speech to Text (STT) converts live audio into text in real time. You stream audio over a WebSocket and receive partial transcripts while the caller is speaking, then a final transcript with word-level timestamps when you end the turn. It is built for voice agents, IVR, and call analytics, and accepts telephony audio (8 kHz µ-law and A-law) directly.

Speech to Text (Beta) is available in the India region only. Connect to wss://in.api.murf.ai/v1/transcription/stream-input.

Combined with Falcon 2.1, Murf now covers both ends of a voice agent: hearing the caller and speaking back.

Ideal for

Voice agents and IVR: Show the caller’s words as they speak, and pass each completed turn to your LLM. When your agent decides the caller has finished speaking, send {"type": "Finalize"} (see Ending a turn with Finalize) to get that turn’s final transcript immediately.
Collections and payment reminders: Boost recognition of product names and account terms with keyword boosting.
Customer support: Transcribe calls in the caller’s own language, or detect it automatically.
Call analytics and QA: Get word-level timestamps on each final transcript for search, redaction, and review.

Key features

Real-time streaming

Stream audio over one WebSocket. partial transcripts update while the caller speaks, and a final transcript arrives when the turn ends.

12 languages + auto-detect

English (India), Hindi, Assamese, Bengali, Gujarati, Kannada, Malayalam, Marathi, Odia, Punjabi, Tamil and Telugu, or let the model detect the language.

Telephony-ready audio

Send 8 kHz µ-law or A-law straight from your telephony provider, plus PCM and Opus at rates up to 48 kHz.

Keyword boosting

Improve recognition of brand names, product terms and domain vocabulary per connection.

Multi-turn sessions

Keep one connection open for a whole call. End each turn with Finalize; timestamps stay on one session timeline.

Word-level timestamps

Every final transcript includes start and end times for each word.

How it works

Connect

Open a WebSocket to wss://in.api.murf.ai/v1/transcription/stream-input with your API key in the api_key query parameter, and optionally language and keywords.

Stream audio

Send audio as binary frames. The server detects speech and returns partial messages as the turn grows. The default format is 16-bit PCM, 16 kHz, mono; for telephony or other formats, see Audio formats.

End turns and the session

Send {"type": "Finalize"} to end a turn and get its final transcript immediately. Send {"type": "CloseStream"} when the call ends.

Coming soon

The following query parameters are accepted on the connection URL but have no effect yet:

commit_mode=auto: ends a turn automatically when the caller stops speaking. Until it is available, a turn ends only when you send Finalize.
bvc=true: removes background voices before transcription.

Pricing

Speech to Text costs $0.24 per hour of audio, billed per second. Enterprise customers can get custom pricing; contact sales.

See Pricing

Get started

Streaming

Transcribe a file or your microphone in a few minutes.

API reference

Every message, field and close code.

Audio formats

Codecs, sample rates and telephony examples.

Languages

Supported languages and automatic detection.