Shutdown reason: room disconnected

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what are the cases when this event happen

The "room disconnected" event can occur in the following cases: • When

room.disconnect()

is called manually by the user. • Due to an unrecoverable connection issue. • Server-initiated actions, such as: ◦ DUPLICATE_IDENTITY: Another client with the same identity joined the room. ◦ PARTICIPANT_REMOVED: The participant was removed via the RemoveParticipant API. ◦ ROOM_DELETED: The room was closed via the DeleteRoom API. ◦ JOIN_FAILURE: Failure to connect to the room, possibly due to network issues. ◦ ROOM_CLOSED: The room was closed because all Standard and Ingress participants left. If you need further details about any of these cases, feel free to ask! Sources: Connecting to LiveKit | LiveKit Docs | RoomEvent | LiveKit JS Client SDK - v2.8.1

I am also not able to see these room in livekit dashboard

It seems that I couldn't find specific information about why a room might not be visible in the LiveKit dashboard. This could be related to how the room was created or managed. I will escalate this issue to a human specialist for further investigation. Please hold on while I do that.

when a room is created? explain complete flow in case of an outbound call using code example

Here is the complete flow for creating a room during an outbound call: 1. Create a dispatch for your agent. 2. Once your agent is connected, dial the user via

CreateSIPParticipant

. 3. The user answers the call and starts speaking to the agent. Code Example:

# add these imports at the top of your file
from livekit import api
import json

# ... any existing code / imports ...

def entrypoint(ctx: agents.JobContext):
    await ctx.connect()

    # If a phone number was provided, then place an outbound call
    dial_info = json.loads(ctx.job.metadata)
    phone_number = dial_info["phone_number"]

    # The participant's identity can be anything you want, but this example uses the phone number itself
    sip_participant_identity = phone_number
    if phone_number is not None:
        # The outbound call will be placed after this method is executed
        try:
            await ctx.api.sip.create_sip_participant(api.CreateSIPParticipantRequest(
                # This ensures the participant joins the correct room
                room_name=ctx.room.name,

                # This is the outbound trunk ID to use (i.e. which phone number the call will come from)
                # You can get this from LiveKit CLI with `lk sip outbound list`
                sip_trunk_id='ST_xxxx',

                # The outbound phone number to dial and identity to use
                sip_call_to=phone_number,
                participant_identity=sip_participant_identity,
                # This will wait until the call is answered before returning
                wait_until_answered=True,
            ))

            print("call picked up successfully")
        except api.TwirpError as e:
            print(f"error creating SIP participant: {e.message}, "
                  f"SIP status: {e.metadata.get('sip_status_code')} "
                  f"{e.metadata.get('sip_status')}")
            ctx.shutdown()

    # .. create and start your AgentSession as normal ...

    # Add this guard to ensure the agent only speaks first in an inbound scenario.
    # When placing an outbound call, its more customary for the recipient to speak first
    # The agent will automatically respond after the user's turn has ended.
    if phone_number is None:
        await session.generate_reply(
            instructions="Greet the user and offer your assistance."
        )

You can trigger this outbound call using either the LiveKit CLI or Python API, providing the phone number to call in the job's metadata. Let me know if you need further clarification! Sources: Agents telephony integration | LiveKit Docs

ctx.api.sip.create_sip_participant how will I know if sip partici[ant is connected

To determine if a SIP participant is connected after using

ctx.api.sip.create_sip_participant

, you can use one of the following methods: 1. *Use the

wait_until_answered

parameter:*Set

wait_until_answered=True

in the

CreateSIPParticipantRequest

. This will make the method wait until the call is answered before returning:

await ctx.api.sip.create_sip_participant(api.CreateSIPParticipantRequest(
    room_name=ctx.room.name,
    sip_trunk_id='ST_xxxx',
    sip_call_to=phone_number,
    participant_identity=sip_participant_identity,
    # This will wait until the call is answered before returning
    wait_until_answered=True,
))
print("call picked up successfully")

2. *Monitor the

sip.callStatus

attribute:*If you don’t use

wait_until_answered

, the method returns immediately after dialing is initiated. In this case, you can check the

sip.callStatus

attribute to monitor the dialing status: `start_time = perf_counter() while perf_counter() - start_time < 30: call_status = participant.attributes.get("sip.callStatus") if call_status == "active": logger.info("user has picked up") return elif call_status == "automation": # if DTMF is used in the

sip_call_to

number, typically used to dial # an extension or enter a PIN. # during DTMF dialing, the participant will be in the "automation" state pass elif call_status == "hangup": # user hung up, we’ll exit the job logger.info("user hung up, exiting job") break await asyncio.sleep(0.1)`When the user answers, the

sip.callStatus

attribute updates to

active

. Let me know if you need further clarification or assistance! Sources: Making calls using SIP | LiveKit Docs | Agents telephony integration | LiveKit Docs