A lot of our livekit cloud outbound telephony call...
# ask-ai
m
A lot of our livekit cloud outbound telephony calls to one specific number are failing with
sip request timed out
. The timeout seems to be not even a second after the SIP is initiated. It does sporadically work and we can't find any pattern between the ones that are timing out and the ones that are succeeding. How can we get more information, or how can we increase the sip request timeout?
✅ 1
t
Thanks for reaching out, Amanda. I'm an AI assistant and I'm here to help. Give me a moment while I research this topic.
Thanks — I hear you: outbound calls to one number are intermittently failing with "sip request timed out" very quickly (under a second), and you want more diagnostics or a way to increase the SIP timeout. Short answer • You cannot increase a user-configurable SIP request timeout in LiveKit Cloud. • The very-short timeout strongly indicates the remote SIP endpoint or trunk provider is rejecting or not answering the request rather than a slow-network timeout. What you can do next (steps) 1. Gather LiveKit-side signals ◦ Check webhook events (especially
participant_left
and its
reason
) for the failing calls to get the LiveKit close reason. ◦ Use Agent Observability / traces to correlate the session timeline, signaling spans, and any errors for each failed call. This will show where the failure occurs in the flow. 1. Check your SIP trunk/provider ◦ Ask your SIP trunk provider for logs for the failing call attempts (they often show SIP response codes or immediate rejects). ◦ Ask them to check for IP/domain blocking, ANI/caller-ID rejections, rate limits, or geographic routing issues to that destination number. 1. Test to isolate ◦ Try the same outbound number via a different trunk/provider (if available) to see if failures persist. ◦ If you can, test using a LiveKit Phone Number (removes third‑party trunk as a variable). 1. Share example events ◦ If you can share (paste) one or two webhook event payloads or trace IDs from Agent Observability for failed calls, I can look at the exact LiveKit reason and help interpret next steps. If you’d like, start by sending one webhook payload or a trace ID for a failed attempt and one that succeeded — I’ll compare them and suggest targeted next steps. Sources: Understand and Improve Agent Latency | LiveKit | Understand and Improve Agent Latency | LiveKit | Introducing LiveKit Phone Numbers: zero to ringing in 60 seconds | LiveKit | Deployment reliability on LiveKit Cloud | LiveKit | Agents telephony integration | LiveKit Documentation, with interactive elements
m
We're using livekit cloud but the livekit agent that does the dial is self-hosted in our AWS. Would the SIP request come from our AWS server or from Livekit cloud IPs?
t
Good question — thanks for clarifying your architecture. The SIP INVITE requests will originate from your self‑hosted LiveKit agent running in your AWS environment, not from LiveKit Cloud IPs. That means the public IP (or NATed IP) of your AWS host is what the SIP provider and destination see as the source. What to check next • Confirm the agent’s public IP(s) and ensure those IPs are allowed by your SIP trunk/provider (no blocking or ACL issues). • Verify NAT/port mapping on the AWS host (SIP/UDP/TCP ports and RTP ports) and any firewall rules. • Ask your SIP provider to look up logs by your reported source IP to see immediate rejections or SIP response codes. • Use your webhook events or Agent Observability traces to correlate failures with the agent instance and timestamp (this helps the provider locate the exact attempt). If you want, share one failed webhook payload or a trace ID and the public IP of the agent (or confirm you’ve checked it) and I’ll help interpret the LiveKit-side details. , with interactive elements