Hi everyone. I am working on a SIP–WebRTC integration using LiveKit and fusionPBX. We are currently facing an issue where a SIP participant does not appear to join a LiveKit room, and I would appreciate your guidance. Scenario Overview A WebRTC user initiates a call to a specific number A LiveKit room is created successfully The callee WebRTC user joins the room without issue • In parallel, a SIP leg is initiated toward our SIP trunk (FusionPBX) • The SIP call is answered successfully on FusionPBX (extension picks up) Issue Observed The SIP participant does not appear as a participant in the LiveKit room LiveKit logs show only one WebRTC participant present As a result, no audio is exchanged between WebRTC users and the SIP leg SIP signaling is successful, and media ports are open and reachable • There are no NAT or firewall issues Key Concern The main concern is that while the SIP call is successfully established, the SIP leg does not seem to be bridged or attached to the LiveKit room, which prevents media exchange. Questions Are there any known limitations or additional requirements for SIP participants joining LiveKit rooms? Is there any specific configuration or signaling step required to associate the SIP leg with an existing room?

What is the call ID on your dashboard. What room do you see the call in? Is the the same as the session ID for the WebRTC user?