Here’s a quick rundown of how the inbound call flow works in LiveKit: 1. Call Initiation: Someone dials a phone number tied to your SIP trunking provider, which then sends an INVITE request to LiveKit SIP. 2. Authentication and Authorization: LiveKit SIP checks the call using your configured credentials and looks for a matching dispatch rule. If something doesn’t match or the rule isn’t found, the call gets rejected. 3. Call Routing and Room Creation: Once authenticated, a SIP participant is created, and the caller is placed into a LiveKit room according to the dispatch rule. Other participants might join too, depending on that rule. 4. Caller Interaction: The caller will hear a dial tone until someone else joins and starts speaking. If a PIN is needed, they’ll be asked to enter it. To get this set up, you’ll need: • A SIP trunking provider (like Twilio or Telnyx) that’s set up to send SIP traffic to LiveKit SIP. • LiveKit SIP configured with an inbound trunk and dispatch rules to route calls into rooms. • Optionally, an AI voice agent to help handle calls if you want. If you want me to walk you through any part of this or have questions, just let me know—I’m here to help! Sources: Accepting inbound calls | LiveKit Docs | pkg/sip/inbound.go | Inbound calls with Twilio Voice | LiveKit Docs | Inbound calls with Twilio Voice | LiveKit Docs | Accepting incoming calls | LiveKit Docs