Thanks — I’m happy to help. I’m asking about your goal of using FreeSWITCH instead of LiveKit’s SIP gateway so I can give practical guidance.
Could you answer a few quick questions?
• Do you want FreeSWITCH to fully replace LiveKit’s SIP gateway, or to sit in front/behind LiveKit as a proxy?
• Are you on self-hosted LiveKit or LiveKit Cloud?
• Which FreeSWITCH version and any special config (transcoding, dialplans)?
• Which transports/protocols do you need (UDP/TCP/TLS for SIP; SRTP/DTLS-SRTP for media)? Do you require ICE/STUN/TURN?
• Do you need codec transcoding between Opus and G.711 or other codecs?
• Any security/auth (SIP Digest, TLS certs) or PSTN gateway requirements?
• If you’ve tried this already, can you share logs, SDP offer/answer, or errors?
Once I have this, I’ll confirm what’s possible and outline the steps or alternatives.
, with interactive elements