Real-Time Audio Recording &
Live Transcription Platform
Browser-based real-time speech-to-text with WebSocket/WebRTC streaming, multi-provider integration, and enterprise-grade accuracy
Business Context
Organizations increasingly need to capture live conversations and convert speech into accurate, readable text in real time
The Problem
Traditional batch transcription solutions fail to meet real-time, interactive, and scalable requirements. Organizations needed a platform that could:
- Capture live conversations (meetings, interviews, voice notes, customer calls)
- Convert speech into accurate, readable text in real time
- Store audio recordings and transcripts securely
- Support multiple transcription engines for accuracy, cost, and redundancy
- Deliver low-latency transcription in web applications
Technical Challenges
Six critical challenges in building an enterprise-grade real-time transcription platform
Low-Latency Requirements
Achieving sub-500ms latency for live transcription while maintaining accuracy across different audio qualities and accents.
Real-Time Streaming
Implementing efficient WebSocket and WebRTC protocols for continuous audio streaming from browser to backend.
Multi-Provider Integration
Seamlessly integrating Azure Speech SDK, AssemblyAI, and ElevenLabs with fallback mechanisms for reliability.
Browser Audio Capture
Capturing high-quality audio at 16kHz mono from various devices and browsers with consistent quality.
Session Management
Managing long-running transcription sessions with graceful recovery from network interruptions and reconnections.
Audio Storage Optimization
Converting and storing audio efficiently as MP3 while maintaining quality and linking with transcript sessions.
The Solution
A browser-based, real-time audio capture and transcription system with enterprise-grade capabilities
Audio Recording & Capture
- •Direct browser microphone access with Web Audio API
- •Configured at 16 kHz, mono for optimal transcription
- •Continuous audio chunk streaming during recording
- •Final audio converted and saved as MP3
Live Transcription
- •Near real-time text output displayed as user speaks
- •Automatic punctuation and formatting
- •Support for long-running sessions
- •Speaker-friendly readable transcript generation
Multi-Provider Integration
- •Azure Speech SDK: Enterprise-grade accuracy and low latency
- •AssemblyAI: Advanced noise handling and filler-word removal
- •ElevenLabs: High-quality speech processing
- •Provider switching and fallback for reliability
Real-Time Streaming
- •WebSocket: Bi-directional audio and transcript streaming
- •WebRTC: Efficient real-time audio transport
- •Low-latency updates to UI
- •Reduced network overhead for long sessions
Audio Storage & Playback
- •Recorded audio converted to MP3 format
- •Audio files linked with transcript sessions
- •Playback, review, and export capabilities
- •Optimized storage footprint
Session Management
- •Session-based authentication
- •Graceful handling of network interruptions
- •Accurate transcript recovery after reconnects
- •Provider-level failover mechanisms
Results & Impact
Measurable outcomes demonstrating platform performance and reliability
Real-time transcription with sub-500ms latency for live speech
High accuracy using multiple speech-to-text engines
Successfully processed over 100,000 hours of audio
Support for multiple languages with Azure and AssemblyAI
Fast session initialization and audio capture startup
Reliable service with provider failover mechanisms
System Architecture
Four-layer architecture for real-time audio capture, streaming, and transcription
Frontend
- →Browser Microphone Access
- →Web Audio API for Capture
- →Live Transcript Rendering
- →WebSocket/WebRTC Streaming
Backend
- →Real-Time Streaming Services
- →Audio Stream Router
- →Transcript Aggregation
- →MP3 Conversion & Storage
Speech Providers
- →Azure Speech SDK
- →AssemblyAI
- →ElevenLabs
- →Provider Failover Logic
Storage
- →Audio Files (MP3)
- →Transcripts & Metadata
- →Session Indexing
- →User Data
Technology Stack
Frontend
- Web Audio API
- WebSocket
- WebRTC
- React
Backend
- Real-Time Streaming
- Node.js/FastAPI
- WebSocket Server
Speech-to-Text
- Azure Speech SDK
- AssemblyAI
- ElevenLabs
Infrastructure
- Azure/AWS
- MP3 Encoding
- Secure Channels
Business Impact
Real value delivered through modern, scalable, and flexible transcription infrastructure
Real-Time Performance
Low-latency transcription enabling live interactive applications and instant feedback
High Accuracy
Multiple speech engines ensure enterprise-grade accuracy with provider-level redundancy
Multi-Language Ready
Support for global use cases with strong multilingual capabilities across all providers
Scalable & Reliable
Handles high-volume sessions with graceful failover and optimized storage footprint
Use Cases
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