Thanks — I checked the snippet you shared (you're asking whether it correctly detects an answered SIP call and triggers the agent to speak). The logic is correct and will work, with a few optional improvements you may want to add. What’s correct • Listening for the

participant_attributes_changed

event and checking

participant.kind == rtc.ParticipantKind.PARTICIPANT_KIND_SIP

targets SIP participants. • Checking

"sip.callStatus"

in

changed_attributes

and testing for

"active"

correctly detects when the callee answered. • Creating an async task for

session.say(...)

avoids blocking the event handler; the

add_done_callback

provides completion logging. • Using

nonlocal

for

call_was_answered

and

call_answered_at

is valid if those variables are defined in the enclosing scope. Optional recommendations (not required to function) • Add a guard to avoid handling duplicate events: ◦ e.g.,

if call_was_answered: return

before you set it. • Add try/except around

session.say(...)

inside

say_first_sentence()

to catch TTS/network errors and log them. • Keep a reference to tasks (e.g., append to a list) until they finish to avoid accidental GC; your

add_done_callback

helps but storing tasks is more explicit. • Include identifying info in logs (like

participant.identity

or

ctx.room.name

) for easier debugging. • If timing matters, consider recording

call_answered_at

before creating the task. If you’d like, I can: • Show a short revised snippet with the try/except and idempotency guard. • Help add task-tracking boilerplate. Which would you prefer? Sources: Participant attributes and metadata | LiveKit docs | Handling events | LiveKit docs | SIP Lifecycle Management Agent | LiveKit docs | Job lifecycle | LiveKit docs | Agents telephony integration | LiveKit docs