We have a self hosted environment where we are getting a 477 LiveKit Community #ask-ai

We have a self-hosted environment where we are get...

magnificent-furniture-73003

04/23/2026, 10:51 AM

We have a self-hosted environment where we are getting a 477 SIP issue. Our livekit server says it is not able to find the next hop and this seems to be happening only for abruptly ended calls. This is on livekit version 1.4.4

tall-airline-16756

04/23/2026, 10:51 AM

Thanks for reaching out, Varun. I'm an AI assistant and I'm here to help. Give me a moment while I research this topic.

tall-airline-16756

04/23/2026, 10:51 AM

Thanks — I’m responding to your report about the 477 SIP error on your self‑hosted LiveKit 1.4.4 where the server reports it “cannot find the next hop,” and it happens for abruptly ended calls. I can help diagnose this — a few quick checks and some logs will narrow it down. Could you provide the following? 1. Confirmation that your LiveKit SIP service is deployed and running separately alongside the LiveKit server. 2. The dispatch rule(s) involved (mask or redacted JSON is fine), especially the

trunk_ids

and rule type (Individual / Direct / Callee). 3. The SIP trunk configuration used for those calls (provider name, transport UDP/TCP/TLS, whether IPs are whitelisted). 4. SIP signaling logs (INVITE, BYE, REFER) from the time of an abrupt disconnect — paste the relevant call flow snippets or capture (redact secrets). 5. Any network or firewall rules between the SIP provider, SIP service, and LiveKit server, and whether SIP pinning or regional pinning is enabled. What I’ll look for once you provide those: • Missing or mismatched

trunk_ids

or dispatch rule mapping causing routing to have no next hop. • Abnormal BYE/REFERRING behavior in the SIP trace when calls end abruptly (dropped disconnect signaling). • Transport or firewall drops that prevent the SIP service from reaching the next hop. • Whether SIP pinning or auto-egress/transfer rules are affecting routing. If you can share the dispatch rule and a short SIP trace for one failing call, I’ll review and give targeted steps to fix or mitigate. Sources: Understand and Improve Agent Latency | LiveKit | Agents telephony integration | LiveKit Documentation | LiveKit Phone Numbers | Connect voice agents to phone lines | LiveKit | Checklist for Regional Deployments | LiveKit | Checklist for Regional Deployments | LiveKit, with interactive elements

magnificent-furniture-73003

04/23/2026, 8:22 PM

is the jitter buffer in livekit sip 1.2.0 known to cause issues?

tall-airline-16756

04/23/2026, 8:22 PM

Thanks — you’re asking whether the jitter buffer in LiveKit SIP v1.2.0 is known to cause issues. I don’t have any confirmed reports that the jitter buffer in v1.2.0 is a known issue. To help investigate your specific case, can you share: • The symptoms you’re seeing (packet loss, audio stuttering, one-way audio, latency spikes). • Sample call timestamps and a short SIP/RTP capture or logs from the SIP service and LiveKit around those calls (redact secrets). • Your jitter buffer settings (if customized) and transport (UDP/TCP/TLS). • Whether the problem is isolated to a provider, network, or region. With those details I’ll review the traces for jitter buffer behavior and recommend targeted fixes (config changes, tuning, or troubleshooting steps). , with interactive elements

Open in Slack

Previous Next