Thanks — I understand your call is being dropped after ~33 seconds because the telco reports a codec mismatch (they want PCMA/PCMU, not G722). I can help you get this fixed — to make targeted instructions I need a couple quick details about how you have LiveKit connected to the telco. Could you tell me: • Are you using LiveKit’s SIP gateway (LiveKit ↔️ SIP trunk) or a custom SIP/VoIP bridge? • If using LiveKit’s SIP gateway, which version/config method are you using (config file, container env vars, or API)? • If you can, share the SIP SDP or the list of codecs you see being offered from the LiveKit side (redact any sensitive data). A short excerpt that shows the audio codecs is sufficient. What I can already say: • LiveKit supports PCMU and PCMA as audio codecs, and it can transcode between WebRTC/Opus and these codecs when bridging to SIP. • G722 being sent to a provider that refuses it will cause the telco to tear down the call; the fix is to ensure the SIP offer/answer excludes G722 and only offers PCMA/PCMU (or sets PCMA/PCMU as preferred). If you confirm the setup details above I will: 1. Show the concrete config changes or SDP manipulation steps to stop advertising/sending G722 and prefer PCMA/PCMU. 2. Provide a short checklist to validate the change (how to inspect the SDP or logs and how to confirm the telco accepts the call). 3. If needed, give sample SDP snippets or the exact env/config lines to change. Send the setup details or a small SDP excerpt and I’ll walk you through the precise steps. Sources: sip | livekit/livekit_composite | DeepWiki | client-sdk-js | livekit/livekit_composite | DeepWiki