Stream live call audio securely and in real time from EnableX Voice to any external system for advanced processing, analytics, or AI decisioning.
EnableX Voice Streaming is an extension of the EnableX Voice API that allows live audio from ongoing phone calls to be
streamed in real time over secure WebSocket connections to third-party systems.
Stream caller and agent audio to external systems for analysis
Inject processed or generated audio back into the live call
Enable bidirectional, real-time interaction between callers and AI or automation systems
This unlocks a new class of real-time, programmable voice applications—beyond traditional IVR or post-call processing.
Send live audio from calls to your application and inject audio back into the same call. Build conversational AI agents, dynamic prompts, and real-time interventions with true two-way audio flow.
Sub-100ms end-to-end latency ensures natural conversations, real-time AI responses, and seamless caller experiences—critical for voice bots and live decisioning.
Easily integrate with: - Speech-to-Text engines - Large Language Models (LLMs) - Sentiment and intent detection engines - Voice biometrics and fraud systems - Custom ML pipelines.
TLS / SRTP encrypted media streams with token-based authentication, built on a GDPR, HIPAA, and SOC 2 compliant architecture.
Monitor call quality, latency, stream health, and performance metrics through real-time dashboards and logs.
99.99% uptime SLA with Redundant infrastructure across multiple regions.
Deploy AI voice agents that handle routine queries, assist human agents in real time, and escalate intelligently when needed. Perform live sentiment analysis to adapt tone and responses dynamically.
Stream audio to intent detection models that understand natural language in real time. Route calls instantly to the right agent—without rigid IVR trees.
Analyze voice patterns and conversational behavior in real time to detect fraud, social engineering, or account takeover attempts during the call.
| Capability | EnableX | Twilio | Plivo | Exotel |
|---|---|---|---|---|
| WebSocket audio streaming | Yes | Yes (Media Streams) | Yes (Audio Streaming) | Yes (Stream / Voicebot Applet) |
| Bidirectional (two-way) | Yes | Yes | Yes | Yes (Voicebot Applet) |
| Audio format | Telephony-grade | 8 kHz μ-law mono | 8 kHz telephony-grade | 16-bit 8 kHz PCM slin |
| Built-in speech-to-text | Yes — 100+ languages, IndicVoices | No (BYO ASR) | No (BYO ASR) | No (BYO ASR) |
| Indian-language accuracy | 95%+ on major Indian languages | Depends on your ASR | Depends on your ASR | Depends on your ASR |
| Managed AI voice agent included | Yes (AI Voice Agent) | No (BYO stack) | No (BYO stack) | Partial (Voicebot Applet + BYO AI) |
| On-prem / hybrid deployment | Yes | No | No | No |
| Full-stack CPaaS (Video+Voice+Msg+AI) | Yes | No | No | No (voice/SMS focus) |
| Meta-approved WhatsApp BSP | Yes (via partner) | Yes | Yes | Yes (via partner) |
| Primary region focus | Global (India, SEA, Middle East Focus) | Global (US-centric) | Global (US-centric) | India primary |
Enterprises trust EnableX globally
for mission-critical voice workflows
real-time latency for natural conversations
compared to alternative platforms
April 20, 2026 / Pankaj Gupta
March 26, 2026 / Pankaj Gupta
July 11, 2025 / Alison Chase
A voice streaming API lets you fork the raw audio of a live phone call and stream it in real time over a WebSocket — to a speech-to-text engine, an AI voice agent, a sentiment-analysis service, or a supervisor dashboard. It's the plumbing behind most modern AI voice experiences.
Unlike call recording (which gives you an audio file after the call ends), voice streaming delivers the audio as it happens, with latency measured in hundreds of milliseconds. Every major CPaaS now offers some version of this: EnableX Voice Streaming API, Twilio Media Streams, Plivo Audio Streaming, Exotel Stream Applet. They differ in format, bidirectional support, deployment options, and how tightly they integrate with AI.
The EnableX Voice Streaming API forks the raw audio of any voice call on the EnableX platform and streams it over a WebSocket to a destination you specify — your transcription engine, your AI voice agent, your analytics pipeline, or your own application.
It is part of the broader EnableX Voice API stack and integrates directly with the EnableX AI Voice Agent and the Dialogs Cloud Conversational AI Platform — so you can stream audio to EnableX's own AI, a third-party model (OpenAI Realtime, Google Gemini, ElevenLabs, Deepgram), or your proprietary stack.
When a call is placed, EnableX opens a WebSocket connection to your URL, streams base64-encoded audio frames as the call progresses, and closes the socket when the call ends. Your server receives the audio, processes it, and optionally sends audio back for bidirectional use cases.
Control messages (call started, call ended, DTMF events) travel on the same socket. Your WebSocket endpoint must stay up for the duration of the call. EnableX handles reconnection, sequencing, and clock sync so you don't have to rebuild that logic.
EnableX streams raw audio frames over WebSocket as base64-encoded μ-law (ulaw) at 8000 Hz mono — the same telephony-grade format as Twilio Media Streams, compatible with all major ASR engines including Google Speech-to-Text, Deepgram, AWS Transcribe, and Azure Speech, so existing speech-to-text stacks work without re-encoding.
Both directions use the same format — audio received from EnableX and audio sent back into the call are both ulaw 8000 Hz mono, Base64 encoded in a JSON payload over the WebSocket connection.
Yes — bidirectional streaming is supported, so your WebSocket application can both receive caller audio and send audio back to be played on the call. This is what enables AI voice agents, conversational IVR, and real-time voice assistants.
Twilio, Plivo, and Exotel all offer bidirectional streaming on specific applets (Twilio bidirectional Media Streams, Plivo bidirectional audio streaming, Exotel Voicebot Applet). EnableX does the same, and additionally lets you hand off to the built-in EnableX AI Voice Agent without writing the WebSocket logic yourself.
Yes. Real-time transcription is the single most common use case. Stream the audio to Google Speech-to-Text, Deepgram, AWS Transcribe, Azure Speech, OpenAI Whisper, ElevenLabs, or any ASR engine — or use EnableX's built-in speech-to-text, which supports 95%+ recognition accuracy across Indian languages using the IIT Madras IndicVoices dataset.
For Indian-language call centres, the EnableX built-in ASR often beats global engines on Hindi, Tamil, Telugu, Marathi, Bengali, and code-mixed Hinglish speech. Read our guide on building speech-to-text systems in WebRTC calling for architecture patterns.
Yes. Raw streamed audio can be fed to a voice-biometrics engine to verify caller identity in the first few seconds of a call — useful for banking IVR, high-value support, and fraud prevention in BFSI.
EnableX's on-premise deployment option is a meaningful advantage here: regulated banks often cannot send voice biometrics to US cloud regions. EnableX deploys inside your datacentre or a private cloud in India / UAE / Saudi Arabia if required.
Yes. Fork the audio of an agent call into a supervisor dashboard for live listen-in, live transcription with keyword alerts, or AI-driven whisper coaching that suggests what the agent should say next.
Because EnableX streams both sides of the call, you can surface sentiment, escalation risk, and compliance violations in real time — not hours later from recordings.
The EnableX AI Voice Agent is built on top of the Voice Streaming API. If you want a managed, turnkey voice AI experience, you don't integrate the streaming API directly — you configure the AI Voice Agent and EnableX handles the WebSocket, ASR, LLM, and TTS stack end-to-end.
If you want to use your own LLM, your own ASR, or your own TTS, use the Voice Streaming API directly and plug in your stack. Both paths are first-class supported.
Both stream raw audio over WebSocket with bidirectional support. Choose EnableX if you need on-premise deployment, Indian-language ASR built in, integrated AI Voice Agent, or a single vendor for Voice + WhatsApp Business API + Video + Conversational AI. Choose Twilio if you need global reach with US-centric cloud and a mature developer ecosystem.
Twilio Media Streams is cloud-only (US-centric regions), uses 8 kHz μ-law mono by default, and requires you to bring your own AI stack. EnableX Voice Streaming offers the same raw-stream capability plus on-prem / hybrid deployment, BSP-grade WhatsApp integration from the same account, and an optional built-in AI Voice Agent so you don't have to assemble ASR + LLM + TTS yourself. Twilio also killed its Video API in 2023, so if you need Video + Voice + Messaging on one platform, EnableX is the cleaner choice.
Plivo Audio Streaming is a solid WebSocket audio-forking service with bidirectional support and similar telephony-grade audio format. EnableX offers the same streaming capability plus full-stack CPaaS (Video, WhatsApp Business API, RCS, Conversational AI), on-premise deployment for regulated industries, and built-in Indian-language ASR.
If your need is messaging-plus-voice-streaming on cloud infrastructure, Plivo is credible. If your need is voice streaming as one piece of a larger AI voice agent or contact-centre build — especially in India, SEA, or the Middle East — EnableX is the better match because the AI Voice Agent and Conversational AI Platform are already wired into the same platform.
Sign up for an EnableX account, enable Voice Streaming in the console, generate an API key, and point your WebSocket endpoint URL in the call configuration. Working samples are available in the Developer Docs.
Most developers have a bidirectional proof-of-concept running in under a day. For AI voice agent use cases, the managed EnableX AI Voice Agent path is typically faster than DIY WebSocket + LLM integration.