AI Products & Automation
AI voice & speech products
Voice agents, transcription and speech features with a real job to do.
What it is
Speech built into a product: transcription for records and search, voice interfaces for calls and hands-free use, and the latency work that makes either tolerable.
Why businesses need it
Voice is the fastest input method for a customer on a call or a worker with no free hands. It only ships when latency, interruption and accuracy are engineered, not demoed.
What MIWEBX delivers
- Transcription of calls, meetings and field recordings
- Summaries, tags and search built on top of transcripts
- Inbound and outbound voice agents for support and verification
- Voice-driven interfaces inside an existing app
- Language and accent handling for a specific user base
Core features
- Streaming transcription with defined latency targets
- Speaker separation and role labelling
- Barge-in, silence and failure handling on live calls
- Domain vocabulary tuned with a word list per client
- Redaction and retention rules for sensitive recordings
- Accuracy measured on real client audio, not public benchmarks
Technical capabilities
- Transcription and speech providers sit behind one interface, selectable per language, channel and cost
- Streaming audio over a socket with buffering, reconnect and sequence recovery
- Endpointing and turn detection tuned so interruption feels conversational rather than scripted
- Domain vocabulary and hotword lists applied at decode time, held per client
- Post-transcription normalisation of names, numbers, currency and formatting per locale
- Redaction pass over sensitive spans before storage, with retention enforced in the store
- Sampled sessions graded against transcripts of the client's own recordings, not public benchmarks
How we build it
- Find the moment where speaking beats typing, and what has to happen next
- Collect representative audio for the actual user base and its accents
- Establish a baseline on that audio before committing to an interface
- Build the streaming path with reconnect, fallback and failure states
- Layer summarisation, tagging or actions on top of the transcript
- Set consent, redaction and retention, then monitor quality on live sessions
Technology stack
- WebRTC
- Python
- Node.js
- Whisper-class ASR
- Speech APIs
- PostgreSQL
- Redis
What you receive
- The voice or transcription feature integrated in your product
- Vocabulary and formatting configuration for your domain
- Measured latency and accuracy report on your own audio
- Retention, consent and redaction settings documented
- Fallback paths for failed or interrupted sessions
Business benefits
- Calls become searchable data instead of a memory
- Hands-free work where typing is impossible
- Faster resolution without dropping the human touch
- Compliance handled at the feature level