VoiceAI
← Back to blog

SIP vs WebRTC: Choosing the Right Infrastructure for Your Voice Agent

Admin User · 6 May 2026 · 2 reads

Two Protocols, Two Worlds

When building a voice AI agent, one of the first infrastructure decisions you'll face is how the audio gets to your agent in the first place. Two protocols dominate: SIP (Session Initiation Protocol) and WebRTC (Web Real-Time Communication). They're not interchangeable — each has distinct strengths, and your choice will shape your entire stack.

What Is SIP?

SIP is the protocol that powers the traditional telephone network. When someone calls a phone number and your voice agent answers, that call arrives over SIP. It's the backbone of:

  • PSTN (Public Switched Telephone Network) calls
  • Business phone systems (PBX)
  • Carriers like Twilio, Telnyx, and Vonage

If you need your voice agent to handle real phone calls — inbound or outbound — you need SIP (or a carrier that handles SIP for you).

What Is WebRTC?

WebRTC is a browser-native protocol for peer-to-peer audio and video. It's what powers:

  • In-browser voice agents (embedded on your website)
  • Video call applications
  • Real-time collaboration tools

WebRTC doesn't touch the phone network — it's purely internet-based. If you want a "click to talk" button on your website that connects a visitor to a voice agent, WebRTC is the right choice.

Side-by-Side Comparison

SIP: Works with real phone numbers, reaches anyone with a phone, requires a carrier, higher setup complexity.

WebRTC: Browser-only, no phone number needed, instant setup, limited to web sessions.

Which Should You Choose?

The answer depends on your use case:

  • Outbound sales calls to prospects → SIP (they need a real phone call)
  • Inbound support on a toll-free number → SIP
  • Website chat-to-voice widget → WebRTC
  • In-app voice assistant → WebRTC

Many production deployments use both — WebRTC for web-embedded agents and SIP for telephony-based scenarios. Platforms like Vapi and LiveKit support both protocols, making the transition seamless.

Latency Considerations

Both protocols can achieve sub-200ms audio latency on good connections. SIP adds a small overhead from carrier routing, but in practice the difference is negligible for conversational AI. Your STT and LLM pipeline will dominate the total response latency far more than the transport layer.