LiveKit's official telephony integration is based on SIP (Session Initiation Protocol), using SIP trunks from providers like Twilio or Telnyx to bridge phone calls into LiveKit rooms. There is currently no official example or documented support for direct telephony integration using pure WebSocket instead of SIP. However, LiveKit does utilize WebSockets for certain internal functions, such as transporting SIP messages or streaming audio tracks to servers for real-time processing. Still, for full telephony (PSTN) integration, SIP remains the documented and supported approach. If your architecture requires moving away from SIP toward direct WebSocket-based telephony, you would need to implement a custom solution. The official guidance, best practices, and examples for this scenario are not covered in the current documentation. If you need specific details on SIP-based integration or using LiveKit's WebSocket transport for related purposes, let me know your requirements and I can provide targeted guidance based on those supported areas. Sources: Integrating with Telephony | LiveKit Docs | Integrating with Telephony | LiveKit Docs | Integrating with Telephony | LiveKit Docs | LiveKit Docs | Voice Assistants | livekit/livekit_composite | DeepWiki | transport/ws.go | Track Egress | LiveKit Docs | Agents telephony integration | LiveKit Docs