I made an outbound call using sipparticipant api b...
# ask-ai
l
I made an outbound call using sipparticipant api but when the user is joined it is not able to hear anything the ai agent is saying why it that happening
t
Thanks for reaching out, Mohammed. I'm an AI assistant and I'm here to help. Give me a moment while I research this topic.
By default, SIP participants emit no audio while the call connects. To address this, you can set the
play_dialtone
parameter to
true
in your
CreateSIPParticipant
request. This will cause the SIP participant to play a dial tone to the room until the phone is picked up. Let me know if you need further assistance with this! Sources: Make outbound calls | LiveKit Docs
l
Not like that after the user connect to the livekit room the agent is not responding
t
Hey there! I'd love to help you troubleshoot this. Could you share a bit more about what's going on? It would be super helpful if you could tell me: • Are there any error messages or logs from the AI agent? • Is the AI agent configured to handle audio streams from the LiveKit room? • Does the issue happen in all calls or just some specific ones? The more details you can provide, the better we can track down what might be causing the problem. I'm here to help you figure this out!
l
No i am not getting any error. Yes i am configured for it Just in some specific one
t
Hey there! 👋 I totally get that troubleshooting AI agent response issues can be frustrating. Let me break down some possible reasons why your agent might be going quiet: • Sometimes the Large Language Model can get a bit tricky - if it's not set up just right when a user's turn ends, the agent might accidentally skip responding. • In manual turn detection mode, things can get a little complicated. If the agent session isn't configured correctly or the right communication methods aren't in place, responses can fall through the cracks. • Network hiccups happen to the best of us! WebRTC connection issues can temporarily interrupt your agent's ability to chat. • If something goes wrong during the SIP connection process, the participant might get disconnected, which obviously makes it tough for the agent to interact. Would you mind double-checking if any of these sound familiar to your situation? Pro tip: Taking a peek at the logs during those tricky calls can give you some super helpful insights. I'm here to help you get things running smoothly! 💡 Sources: GitHub::livekit::agents | .github/next-release/changeset-afeb4064.md | Turn detection and interruptions | LiveKit Docs | Connecting to LiveKit | LiveKit Docs | Making outbound calls | LiveKit Docs