How do I stream live call audio to my server?

To stream live call audio to your server, the exact method depends on where the call is happening:

• Phone/PSTN calls
• WebRTC calls
• VoIP / SIP calls
• In-app calls from your own app
• A telephony provider like Twilio, Plivo, Vonage, etc.

The general pattern

1. Capture the audio on the client or telephony platform
2. Encode it in small chunks (often 20–100 ms frames)
3. Send it over a low-latency transport:
   • WebSocket
   • WebRTC data channel / media server
   • RTP/RTCP
   • provider-specific streaming API
4. Receive it on your server
5. Decode/process/store it

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Common ways to do it

1) Using a telephony provider

If calls come through a provider, this is usually the easiest.

Twilio example: Twilio Media Streams can send live audio from a call to your WebSocket server in near real time.

Typical flow:

• User calls your Twilio number
• Twilio connects the call audio to your WebSocket endpoint
• Your server receives audio frames as JSON messages, often base64-encoded audio payloads

This is ideal for:

• transcription
• call analytics
• live monitoring
• AI agents

You’ll need:

• a public wss:// endpoint
• a Twilio webhook to start the stream
• server logic to parse media messages

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2) Using WebRTC

If you control the app/browser/mobile client, WebRTC is often best.

Options:

• Send audio to your backend through WebRTC
• Or use a media server like:
  • Janus
  • mediasoup
  • Jitsi
  • LiveKit

You can:

• receive the stream on a server
• fork the audio to a processor
• transcode to PCM if needed

This is useful for:

• browser-based calls
• app-to-app calls
• low-latency pipelines

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3) SIP/RTP audio forwarding

If you have a SIP PBX or softswitch, you can mirror/fork the RTP stream.

Typical components:

• Asterisk / FreeSWITCH / Kamailio
• RTP proxy or media fork
• custom RTP receiver

This is common in enterprise telephony setups.

You’ll handle:

• codec negotiation (G.711, Opus, etc.)
• packet loss/jitter
• RTP timestamps and sequencing

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4) Client-side streaming from a web app

If you’re building your own calling app in the browser:

• Use getUserMedia({ audio: true })
• Process audio with the Web Audio API or AudioWorklet
• Chunk and send over WebSocket to your server

This gives you direct control, but you must handle:

• resampling
• encoding
• buffering
• permissions and device selection

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Important technical details

Audio format

Decide what your server expects:

• PCM 16-bit mono is common for processing
• Opus is efficient for transport
• G.711 μ-law/A-law is common in telephony

Chunk size

For live streaming, keep chunks small:

• 20 ms is common in telephony/WebRTC
• 50–100 ms can reduce overhead but add latency

Transport

Use:

• WebSocket for simple custom streaming
• RTP for telephony standards
• WebRTC for real-time interactive apps

Latency

If you need near-real-time processing:

• keep buffers small
• avoid unnecessary transcoding
• process incrementally as frames arrive

Security

• use TLS/WSS
• authenticate streams
• avoid exposing raw audio endpoints publicly
• comply with call recording consent laws

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A simple architecture

Caller audio → provider/app → stream endpoint → your server → processing/transcription/storage

Example:

• Twilio call
• Twilio Media Streams over WebSocket
• Node.js server receives audio
• Server forwards PCM to speech-to-text API or your own model

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Minimal implementation idea

If you’re using a telephony provider with media streaming:

Server

• Run a WebSocket server
• Accept incoming audio frames
• Decode base64 audio if needed
• Write frames into a buffer or pipe to a processor

Client/provider side

• Configure the call to stream audio to your wss:// endpoint

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If you want the fastest path

Tell me:

1. What kind of call is it? (phone, browser, SIP, app)
2. Who controls the client? (you or a third-party platform)
3. What do you need the audio for? (transcription, storage, AI agent, monitoring)
4. What stack are you using? (Node, Python, Go, etc.)

And I can give you:

• the exact architecture
• sample server code
• provider-specific setup
• audio decoding/handling examples