Real-time / Web

WebRTC (Web Real-Time Communications)

WebRTC is a browser-native API and protocol stack for peer-to-peer audio, video, and data transfer. It runs in every modern browser without plugins and is the foundation for browser-based softphones and voice agents.

Stack overview

WebRTC bundles four protocol families:

• Signaling: typically SIP-over-WebSocket (RFC 7118) or a custom HTTP/WS API. The browser exchanges SDP offers/answers.
• NAT traversal: ICE (RFC 8445), STUN, TURN.
• Media: SRTP (encrypted RTP) keyed via DTLS-SRTP.
• Codecs: Opus (audio), VP8/VP9/H.264/AV1 (video), G.711 (PSTN interop).

Browser API

const pc = new RTCPeerConnection({
  iceServers: [
    { urls: "stun:stun.didhub.io:3478" },
    { urls: "turn:turn.didhub.io:443?transport=tcp", username: "u", credential: "p" },
  ],
});
const stream = await navigator.mediaDevices.getUserMedia({ audio: true });
stream.getTracks().forEach(t => pc.addTrack(t, stream));
const offer = await pc.createOffer();
await pc.setLocalDescription(offer);
// Send offer.sdp to your signaling server (SIP WSS or custom)

WebRTC to PSTN interop

WebRTC alone cannot reach the PSTN — you need a SIP gateway that bridges WebRTC media (DTLS-SRTP, Opus) to traditional SIP/RTP (G.711). DIDHub provides this bridge as part of every SIP trunk: just send your INVITE over WSS and the carrier handles transcoding.

Related terms

Related glossary terms

Asterisk (open-source PBX framework)

Asterisk is the original open-source telephony framework, started by Mark Spencer in 1999. It is a Class 5 PBX engine: it terminates SIP/IAX

Read →

Attestation Levels (A, B, C)

Attestation levels are the three trust ratings that an originating carrier assigns to outbound calls under STIR/SHAKEN. They tell the termin

Read →

Auto-Provisioning (zero-touch desk phone setup)

Auto-provisioning is how you deploy 50, 500, or 50,000 desk phones without manually configuring each one. The phone boots, fetches its confi

Read →

BYOC (Bring Your Own Carrier)

BYOC is a deployment model where you use a third-party SaaS platform (Vapi, Retell, Microsoft Teams, Zoom Phone, Twilio Flex) for the call-c

Read →