@kuralle-agents/livekit-plugin-transport-sip

SIP/RTP telephony transport that bridges Kuralle voice sessions to PBX trunks and SIP endpoints over UDP.

Install

npm install @kuralle-agents/livekit-plugin-transport-sip

Peer dependency:

npm install @kuralle-agents/livekit-plugin

What it does

Handles the UDP SIP signaling and RTP media layers for PBX/SBC integrations, so Kuralle agents can answer inbound telephone calls.

• SIPAgentServer — listens for inbound SIP INVITEs and starts a voice session per call.
• SIPSignaling — manages dialog-level SIP: INVITE, ACK, BYE, CANCEL, OPTIONS, and outbound 200 OK.
• SIPTransportAdapter — implements the transport contract for @kuralle-agents/livekit-plugin sessions.
• RtpSession — sends and receives RTP frames; includes JitterBuffer for packet reordering.
• PCMU / PCMA — G.711 mu-law and a-law codec constants (re-exported from @kuralle-agents/transport-base).
• createSipNativeAudioTransport — factory for use with provider-native realtime sessions.

Current scope: UDP only. Handles INVITE, ACK, BYE, CANCEL, OPTIONS, final 200 OK, and G.711 codec negotiation. re-INVITE, hold/resume, PRACK, REFER, SRTP, and RTCP are not yet implemented — best suited for controlled PBX/SBC deployments.

Usage

import { SIPAgentServer } from '@kuralle-agents/livekit-plugin-transport-sip';
import { createKuralleSession } from '@kuralle-agents/livekit-plugin';
import { GeminiLiveSTT, GeminiLiveTTS } from '@kuralle-agents/livekit-plugin/gemini';

const server = new SIPAgentServer({ port: 5060 });

server.onCall(async (callId, transport) => {
  const session = createKuralleSession({
    runtime: myRuntime,
    stt: new GeminiLiveSTT(),
    tts: new GeminiLiveTTS(),
  });
  await server.startSession(callId, transport, session);
});

await server.listen();

Related

• @kuralle-agents/livekit-plugin — voice session orchestrator
• @kuralle-agents/livekit-plugin-transport-twilio — Twilio Media Streams alternative