# SIP gateway sends 180 Ringing ~12s after agent joins room and publishes audio

**URL:** <https://community.livekit.io/t/sip-gateway-sends-180-ringing-12s-after-agent-joins-room-and-publishes-audio/323>\
**Category:** Telephony\
**Tags:** sip-trunking\
**Created:** [February 18, 2026, 5:24pm UTC](https://community.livekit.io/t/sip-gateway-sends-180-ringing-12s-after-agent-joins-room-and-publishes-audio/323 "2026-02-18T17:24:45Z")\
**Posts on this page:** 3\
**Page:** 1

<div class="post-metadata">

**Author:** ![Kidus\_Asmare](https://sea1.discourse-cdn.com/flex001/user_avatar/community.livekit.io/kidus_asmare/32/374_2.png) [@Kidus\_Asmare](https://community.livekit.io/u/Kidus_Asmare)\
**Post date:** [February 18, 2026, 5:24pm UTC](https://community.livekit.io/t/sip-gateway-sends-180-ringing-12s-after-agent-joins-room-and-publishes-audio/323/1 "2026-02-18T17:24:45Z")

</div>

Hi everyone,

We’re experiencing inbound SIP call failures due to a delay between our agent being ready in a Livekit room and the SIP gateway sending 180 Ringing to the caller.

Our upstream SIP provider has a ~20s INVITE timeout. From our pcap and agent logs on a failed call (Feb 18, 16:02 UTC), the timeline is:

- T+0.0s — INVITE arrives at Livekit, 100 Processing sent immediately ✓
- T+1.7s — Agent connects to room, participant joins
- T+6.7s — Agent publishes placeholder audio track
- T+7.7s — session.start() completes, agent begins speaking intro
- T+20.0s — Provider CANCEL (timeout, caller drops)
- T+20.1s — Livekit sends 180 Ringing (too late)

The agent was fully active and speaking in the room by ~T+8s, but 180 Ringing wasn’t sent until ~T+20s — a 12-second unexplained gap.

Additionally, after our Provider sends CANCEL, the agent room is not torn down. The agent continues running and participant\_disconnected does not fire.

Our SIP trunk uses dispatchRuleIndividual with a roomPrefix. No ringing\_timeout or early media options are configured.

Questions:

1. Can Livekit send 180 Ringing immediately upon INVITE receipt or dispatch rule match, before the agent connects? This would prevent upstream timeouts regardless of agent init time.
2. Is there a ringing\_timeout or early\_media configuration option for inbound SIP trunks?
3. Is the SIP CANCEL being propagated as a participant disconnect event? We’re not seeing it on our end.

Call details:

- Call ID: 09357eb7-4d7d-4daf-9bc8-a1ca24c0ac7b
- Time: Feb 18, 2026 ~16:02 UTC

Happy to share pcap and agent logs if helpful.

Thanks

---

<div class="post-metadata">

**Author:** ![CWilson](https://sea1.discourse-cdn.com/flex001/user_avatar/community.livekit.io/cwilson/32/135_2.png) [@CWilson](https://community.livekit.io/u/CWilson)\
**Post date:** [February 18, 2026, 11:00pm UTC](https://community.livekit.io/t/sip-gateway-sends-180-ringing-12s-after-agent-joins-room-and-publishes-audio/323/2 "2026-02-18T23:00:40Z")

</div>

Are you sure you don’t have another agent running somewhere that maybe intercepting this call?

---

<div class="post-metadata">

**Author:** ![Kidus\_Asmare](https://sea1.discourse-cdn.com/flex001/user_avatar/community.livekit.io/kidus_asmare/32/374_2.png) [@Kidus\_Asmare](https://community.livekit.io/u/Kidus_Asmare)\
**Post date:** [February 19, 2026, 1:18pm UTC](https://community.livekit.io/t/sip-gateway-sends-180-ringing-12s-after-agent-joins-room-and-publishes-audio/323/3 "2026-02-19T13:18:43Z")

</div>

100% sure. We only have one agent. We use completely different projects for dev work as well.