I'm here to help you with the initial audio delay on inbound SIP calls, where the first 0-2 seconds of audio might be cut off. This can happen due to timing mismatches or other external factors. LiveKit has a built-in safeguard with a 1-second delay to make sure the start of the audio stream isn’t lost. If you’re experiencing delays longer than this, here are some friendly steps we can explore together: • Verify SIP Trunk Configuration: Let’s check that your SIP provider isn’t causing delays with early media or SDP negotiation. • Analyze Network Latency: It’s a good idea to look into any network congestion or the geographical distance that might be affecting call routing. • Agent Setup: Make sure your agent dispatch rules are set up correctly and avoid using resource-heavy methods for initial audio processing. • Play Immediate Audio Prompts: You can use audio prompts to gently mask any small delays for your callers. • Adjust Jitter Buffer Settings: If minimizing delay is crucial, consider disabling jitter buffers. Please feel free to reach out if you’d like me to walk you through any of these steps in more detail or if there’s anything else I can assist you with. I’m here to support you! Sources: test/cloud/integration_test.go | SIP troubleshooting guide | LiveKit Docs | Capturing the First Few Seconds of a Call for Processing | Make outbound calls | LiveKit Docs | CHANGELOG.md