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09/17/2025, 11:14 AMtall-airline-16756
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09/17/2025, 11:16 AMtall-airline-16756
09/17/2025, 11:17 AMconst session = new AgentSession({
stt: new DeepgramSTT(),
llm: new OpenAILLM(),
tts: new CartesiaTTS(),
vad: new SileroVAD(),
});
• Room Name Matching: Double-check that the room name you’re using in the CreateSIPParticipant API exactly matches the room the agent should join. This little detail can sometimes cause unexpected behavior.
• Agent Logic: For outbound calls, it’s normal for the agent to wait silently until the other person starts speaking. So, make sure your agent is designed to listen for user input before responding.
• Silence Detection: Adding silence monitoring can be really helpful, especially to handle situations like voicemail or long pauses. Here’s a simple way to implement it:
let userLastSpokeTime = Date.now();
const SILENCE_THRESHOLD = 5000; // 5 seconds
session.on("user_started_speaking", () => {
userLastSpokeTime = Date.now();
});
setInterval(() => {
if (Date.now() - userLastSpokeTime > SILENCE_THRESHOLD) {
console.log("Silence detected; ending call");
session.disconnect();
}
}, 1000);
• SIP Trunk Configuration: Lastly, please verify that your SIP endpoint settings are correct and align with your provider’s requirements.
If you try these steps and still run into issues, please don’t hesitate to reach out. I’m here to help, and we can dive deeper with more diagnostics or examples to get this resolved together.
Sources: Speech & Audio Processing | livekit/livekit_composite | DeepWiki | Agents telephony integration | LiveKit Docs | How to detect and handle silence during calls | Understanding "silence detected on local audio track" warning | SIP troubleshooting guide | LiveKit Docspurple-table-52276
09/17/2025, 11:27 AMtall-airline-16756
09/17/2025, 11:28 AMctx.room is null for outbound calls, it usually means the room context hasn’t been set up properly yet. In Node.js, you’ll want to create and manage the room instance explicitly for those outbound calls. Here’s a friendly guide on how to do that:
• Create the Room: Use the LiveKit SDK to explicitly create a room instance specifically for your outbound call.
• Pass the Room to the Agent: Make sure you pass this room instance to the agent session when you start it. For example:
const room = await connectLiveKitRoom({
url: 'your-livekit-server-url',
token: 'your-access-token',
});
const session = new AgentSession({
room: room,
stt: new DeepgramSTT(),
llm: new OpenAILLM(),
tts: new CartesiaTTS(),
vad: new SileroVAD(),
});
session.start();
• Verify Room Connection: Double-check that the room has connected successfully before you pass it along to the agent session.
If you’d like, I’m here to help you step-by-step with setting up the room or managing the agent session. Just let me know!
Sources: Agents telephony integration | LiveKit Docs | packages/livekit-rtc/README.md | packages/react/src/context/room-context.ts | docs/functions/LiveKitRoom.htmlpurple-table-52276
09/17/2025, 11:31 AMtall-airline-16756
09/17/2025, 11:31 AMpurple-table-52276
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09/17/2025, 11:32 AMpurple-table-52276
09/17/2025, 11:36 AMtall-airline-16756
09/17/2025, 11:36 AMCreateSIPParticipant API call. Let’s work through this together:
• Room Name Matching: Please make sure the room name used when starting the agent exactly matches the room_name parameter in the CreateSIPParticipant request. A helpful tip is to use ctx.room.name to dynamically pass the correct room name.
• Agent Dispatch: When dispatching the agent, try using the --new-room flag. This helps ensure both the agent and the outbound call connect to the same dynamically created room.
• Verify SIP Participant: After you call createSIPParticipant, keep an eye on the sip.callStatus attribute to see if the call moves to an active state.
If you’d like, I’m here to guide you through these steps or help troubleshoot any issues with room synchronization. Just let me know!
Sources: Agents telephony integration | LiveKit Docs | Making calls using SIP | LiveKit Docs | Making calls using SIP | LiveKit Docs | Making outbound calls | LiveKit Docs | packages/sip--make-a-call/README.md