WebRTC vs WebSockets for AI Voice: Why Gabber Offers Both

WebRTC vs WebSocket for AI Voice

When you're building with AI voice, choosing the right real-time protocol matters — a lot.

Two major players dominate:

  • WebRTC: Designed for real-time peer-to-peer media.
  • WebSockets: Built for real-time messaging and data streams.

At Gabber, we support both — because different apps need different approaches:

  • Our real-time AI conversation engine runs on WebRTC.
  • Our TTS-only plug-in API uses WebSockets for easy integration.

Here’s how it breaks down — and how to pick the right tool for your AI app.


Quick Comparison: WebRTC vs WebSockets

FeatureWebRTCWebSockets
ProtocolUDP (peer-to-peer)TCP (client-server)
Setup ComplexityHigh (STUN/TURN servers)Low (single server connection)
Ideal ForLive voice conversationsStreaming TTS audio
LatencyExtremely lowLow
Media HandlingBuilt-in voice/video optimizationYou manage media streams manually

Why Gabber Uses WebRTC for Full Real-Time AI Conversations

When you’re building a dynamic voice-first AI, you need:

  • 📞 Two-way voice communication
  • ⚡ Instant interruption support (mid-sentence)
  • 🔄 Live tool-calling, memory updates, and persona switching
  • 🎙️ Extremely low jitter, low latency media exchange

WebRTC was designed for this.
It optimizes voice streams using:

  • UDP transmission (faster packet delivery)
  • Adaptive bitrate adjustments
  • Echo cancellation and NAT traversal

That’s why Gabber’s core real-time engine — powering AI companions, voice coaching, phone AI bots, and more — is built on WebRTC.


Why Gabber Offers WebSocket TTS for Plug-and-Play Simplicity

Sometimes you don’t need full conversation handling — you just want affordable, high-quality TTS.

If you already have:

  • 🛠️ A WebSocket-driven app
  • 🎮 A game that streams data over WebSockets
  • 📱 A mobile app with a persistent WebSocket layer

…then switching to WebRTC just for voice doesn't make sense.

That’s why Gabber’s TTS-only API is available over WebSockets:

  • 🔌 Simple integration (no STUN/TURN config)
  • 📡 Stream synthesized voice in real time
  • 🧩 Drop-in compatibility with WebSocket-based apps

You get all the benefits of Gabber's high-quality voices (like Orphoeus and Cartesia models) without restructuring your app’s network stack.


Which Should You Use?

Choose WebRTC if:

  • You want interactive, live AI conversations
  • You need interruptable speech and dynamic flow control
  • You’re building full apps, companions, bots, or coaching tools

Choose WebSocket TTS if:

  • You want to embed AI speech into an existing WebSocket-based app
  • You only need one-way audio (TTS output)
  • You prioritize simplicity over conversation complexity

Gabber Gives You Both

Unlike most platforms that force you into one protocol, Gabber adapts to your build:

  • 🎙️ Full conversational AI over WebRTC (voice + tool calling + memory)
  • 🗣️ Simple TTS output over WebSockets (voice only)

Same quality. Same speed. Different flexibility — based on your app’s needs.


TL;DR: One Stack, Two Paths

In 2025 and beyond, great AI apps will be:

  • Real-time
  • Voice-powered
  • Latency-optimized

Gabber gives you the flexibility to build it your way — whether you need full conversational pipelines or lightweight speech streaming.

🚀 Ready to build? Try Gabber's AI Voice Platform