How to Stream Asterisk Audio via WebRTC (2025 Guide)

Modern telephony systems increasingly require real-time audio access. Whether you're building:

• live speech-to-text (STT) pipelines
• AI receptionists that respond instantly
• browser-based monitoring dashboards
• or custom voice-driven applications

…the ability to stream audio from Asterisk to a browser or an AI service becomes essential.

This is where WebRTC comes in.

🔍 What Is WebRTC?

WebRTC (Web Real-Time Communication) is an open standard that enables:

• Real-time audio & video transport
• Low-latency data channels
• Browser-to-browser or browser-to-server communication
• Built-in NAT traversal, STUN/TURN, DTLS-SRTP encryption

WebRTC is supported by all major browsers without plugins. For Asterisk, it provides a secure, low-latency way to send audio directly to a web browser, Node.js service, or an AI backend.

🌟 Why Use WebRTC With Asterisk?

✔ Real-time Speech-to-Text (STT)

If you're building live STT with Whisper, Deepgram, or Google STT, you need audio frames as they happen. WebRTC provides low latency, Opus audio, and reliable jitter buffering.

✔ AI Receptionists & Voicebots

WebRTC enables AI agents to hear the caller instantly, transcribe intent, decide how to respond, and speak back naturally.

✔ Live Call Dashboards

Many businesses want real-time call monitoring and analytics. WebRTC allows secure, low-latency streaming to supervisor dashboards.

✔ Routing Audio to External Engines

You can stream Opus audio to:

• ASR engines
• ML models
• Compliance systems
• Analytics services

🛠 Step-by-Step: Enable WebRTC Audio Streaming in Asterisk

🧩 Step 1: Install WebRTC Dependencies

sudo apt install libsrtp2-dev

Ensure your Asterisk build includes:

• res_http_websocket
• chan_pjsip
• codec_opus

⚙️ Step 2: Configure WebRTC Transport in PJSIP

Edit

/etc/asterisk/pjsip.conf

transport=transport-wss
protocol=wss
bind=0.0.0.0

Create a WebRTC endpoint:

[webrtc-endpoint]
type = endpoint
transport = transport-wss
context = internal
disallow = all
allow = opus
webrtc = yes

📡 Step 3: Enable HTTP + WebSocket

Edit

/etc/asterisk/http.conf

[general]
enabled=yes
bindaddr=0.0.0.0
bindport=8088
websocket_enabled=yes

Restart Asterisk.

🌐 Step 4: Create a Stasis App

In

extensions.conf

exten => 7002,1,Stasis(webrtc_streamer)

🔌 Step 5: Forward Audio to a WebRTC Server or Browser

Example Node.js WebRTC handler:

peer.addStream(asteriskStream);

This can forward Opus audio to STT, AI bots, dashboards, or ML pipelines.

🎯 Summary

With WebRTC enabled, Asterisk becomes a modern audio streaming platform, enabling:

• Real-time transcription
• AI-powered assistants
• Live call dashboards
• Browser-based softphones
• Low-latency AI-driven telephony

WebRTC transforms Asterisk into a next-generation communication system ready for AI.