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Over the past day, we have started observing inter...
# ask-ai
g
Over the past day, we have started observing intermittent delays in SIP provisional responses from LiveKit. Specifically: Asterisk sends INVITE → LiveKit sometimes takes ~2–5 seconds to respond with: • 100 Trying • 180 Ringing This behavior was not observed earlier and the setup had been functioning normally before. Observed behavior: • Issue is intermittent (not all calls) • During affected calls, Asterisk retransmits INVITEs before receiving 100 Trying • Once the provisional responses arrive, the call usually proceeds normally • No consistent network or SIP trunk issues observed on our side
t
Thanks for reaching out, Viswanadh. I'm an AI assistant and I'm here to help. Give me a moment while I research this topic.
Thanks — I hear you: intermittent 2–5s delays in LiveKit sending SIP provisional responses (100/180) that cause Asterisk to retransmit INVITEs. I can help troubleshoot, but I need a few details to narrow this down. Could you provide the following: • LiveKit SIP gateway version or build used. • Deployment type: self-hosted (Kubernetes/docker) or LiveKit cloud? • Approximate call volume / concurrent calls and whether the spikes correlate with delays. • Any recent changes (config, scaling, network, certificates, upstream SIP trunk) in the past 24–48 hours. • Example SIP trace (full INVITE + timestamps and the delayed 100/180 responses) or relevant logs from LiveKit and Asterisk for an affected call. Redact sensitive info if needed. • Network layout between Asterisk and LiveKit (public internet, private VPC, NAT, load balancer, SIP ALG). • Whether TLS or UDP is used for SIP transport. • Whether other SIP responses (PRACK, 200 OK) are also delayed or only provisionals. • Any CPU/memory or connection errors showing on the LiveKit host(s) around the times of affected calls. If you prefer, start by sharing one SIP trace and the LiveKit version; I’ll review it and suggest targeted next steps. , with interactive elements
g
1. using livekit cloud 2. no recent changes
t
Thanks — that helps. To narrow this down for LiveKit Cloud, please share the following so I can investigate the root cause of the intermittent 2–5s provisional response delays: • LiveKit region (or the region/edge you’re using). • Example SIP trace for an affected call (INVITE with timestamps and the delayed 100/180). Redact any sensitive headers if needed. • Timestamp (UTC) of one or two affected calls so I can correlate with LiveKit logs. • SIP transport (UDP, TCP, or TLS) and port used. • Whether calls use a SIP proxy/load balancer or go directly to LiveKit Cloud. • Approximate concurrent call volume and whether you saw any short spikes when delays occurred. • Asterisk retransmit timing you observed (e.g., first retransmit at 500 ms, 1s, etc.). • Any NAT, SIP ALG, or firewall between Asterisk and the internet that could affect UDP. • Whether RTP media started on time once the call proceeded (to confirm only signalling is affected). If you can’t share a full trace, start with the region, one UTC timestamp, transport, and retransmit timing and I’ll use that to look up related LiveKit Cloud logs and advise next steps. , with interactive elements
g
INVITE sip:Number@livekit_id.sip.livekit.cloud SIP/2.0 Via: SIP/2.0/UDP Aserisk_IP:5060;rport;branch=z9hG4bKPja1fed995-c774-4934-b59e-7a2393ed8112 From: "T NAME" sip:+91NUMBER@172.31.39.194;tag=b930463a-cf32-4b2e-a224-6148c289c97c To: sip:Number@livekit_id.sip.livekit.cloud Contact: sip:asterisk@Aserisk_IP:5060 Call-ID: 877baacb-7cf5-4c36-8083-618fbe15ae3d CSeq: 30849 INVITE Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub, histinfo Session-Expires: 1800 Min-SE: 90 Max-Forwards: 70 User-Agent: Asterisk PBX 20.6.0~dfsg+~cs6.13.40431414-2build5 Content-Type: application/sdp Content-Length: 263 v=0 o=- 1190881317 1190881317 IN IP4 Aserisk_IP s=Asterisk c=IN IP4 Aserisk_IP t=0 0 m=audio 23530 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:140 a=sendrecv <--- Transmitting SIP request (1026 bytes) to UDP143.223.91.1755060 ---> INVITE sip:Number@livekit_id.sip.livekit.cloud SIP/2.0 Via: SIP/2.0/UDP Aserisk_IP:5060;rport;branch=z9hG4bKPja1fed995-c774-4934-b59e-7a2393ed8112 From: "T NAME" sip:+91NUMBER@172.31.39.194;tag=b930463a-cf32-4b2e-a224-6148c289c97c To: sip:Number@livekit_id.sip.livekit.cloud Contact: sip:asterisk@Aserisk_IP:5060 Call-ID: 877baacb-7cf5-4c36-8083-618fbe15ae3d CSeq: 30849 INVITE Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub, histinfo Session-Expires: 1800 Min-SE: 90 Max-Forwards: 70 User-Agent: Asterisk PBX 20.6.0~dfsg+~cs6.13.40431414-2build5 Content-Type: application/sdp Content-Length: 263 v=0 o=- 1190881317 1190881317 IN IP4 Aserisk_IP s=Asterisk c=IN IP4 Aserisk_IP t=0 0 m=audio 23530 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:140 a=sendrecv
<--- Transmitting SIP request (1026 bytes) to UDP143.223.91.1755060 ---> INVITE sip:Number@livekit_id.sip.livekit.cloud SIP/2.0 Via: SIP/2.0/UDP Aserisk_IP:5060;rport;branch=z9hG4bKPja1fed995-c774-4934-b59e-7a2393ed8112 From: "T NAME" sip:+91NUMBER@172.31.39.194;tag=b930463a-cf32-4b2e-a224-6148c289c97c To: sip:Number@livekit_id.sip.livekit.cloud Contact: sip:asterisk@Aserisk_IP:5060 Call-ID: 877baacb-7cf5-4c36-8083-618fbe15ae3d CSeq: 30849 INVITE Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub, histinfo Session-Expires: 1800 Min-SE: 90 Max-Forwards: 70 User-Agent: Asterisk PBX 20.6.0~dfsg+~cs6.13.40431414-2build5 Content-Type: application/sdp Content-Length: 263 v=0 o=- 1190881317 1190881317 IN IP4 Aserisk_IP s=Asterisk c=IN IP4 Aserisk_IP t=0 0 m=audio 23530 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:140 a=sendrecv <--- Transmitting SIP request (513 bytes) to TCP143.223.91.935060 ---> OPTIONS sip:livekit_id.sip.livekit.cloud SIP/2.0 Via: SIP/2.0/TCP Aserisk_IP:5060;rport;branch=z9hG4bKPj8769f9e8-0963-45cc-924f-a9c6734683ad;alias From: sip:livekit-endpoint@172.31.39.194;tag=4aaa9fe4-4fb3-4c85-b89b-0c80128b2363 To: sip:livekit_id.sip.livekit.cloud Contact: sip:livekit-endpoint@Aserisk_IP:5060;transport=TCP Call-ID: a5cbc5ba-075c-4408-bc81-f8ef13021514 CSeq: 56 OPTIONS Max-Forwards: 70 User-Agent: Asterisk PBX 20.6.0~dfsg+~cs6.13.40431414-2build5 Content-Length: 0 <--- Received SIP response (389 bytes) from TCP143.223.91.935060 ---> SIP/2.0 200 OK Via: SIP/2.0/TCP Aserisk_IP:5060;rport;branch=z9hG4bKPj8769f9e8-0963-45cc-924f-a9c6734683ad;alias From: sip:livekit-endpoint@172.31.39.194;tag=4aaa9fe4-4fb3-4c85-b89b-0c80128b2363 To: sip:livekit_id.sip.livekit.cloud;tag=b86648e3-e654-4264-a3d3-35b3a97dbe1e Call-ID: a5cbc5ba-075c-4408-bc81-f8ef13021514 CSeq: 56 OPTIONS Content-Length: 0 Max-Forwards: 70 <--- Transmitting SIP request (1026 bytes) to UDP143.223.91.1755060 ---> INVITE sip:Number@livekit_id.sip.livekit.cloud SIP/2.0 Via: SIP/2.0/UDP Aserisk_IP:5060;rport;branch=z9hG4bKPja1fed995-c774-4934-b59e-7a2393ed8112 From: "T NAME" sip:+91NUMBER@172.31.39.194;tag=b930463a-cf32-4b2e-a224-6148c289c97c To: sip:Number@livekit_id.sip.livekit.cloud Contact: sip:asterisk@Aserisk_IP:5060 Call-ID: 877baacb-7cf5-4c36-8083-618fbe15ae3d CSeq: 30849 INVITE Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub, histinfo Session-Expires: 1800 Min-SE: 90 Max-Forwards: 70 User-Agent: Asterisk PBX 20.6.0~dfsg+~cs6.13.40431414-2build5 Content-Type: application/sdp Content-Length: 263 v=0 o=- 1190881317 1190881317 IN IP4 Aserisk_IP s=Asterisk c=IN IP4 Aserisk_IP t=0 0 m=audio 23530 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:140 a=sendrecv <--- Transmitting SIP request (480 bytes) to UDPAIRTEL IP5095 ---> OPTIONS sipAIRTEL IP5095 SIP/2.0 Via: SIP/2.0/UDP Aserisk_IP:5060;rport;branch=z9hG4bKPjc250483b-7d20-49b6-bdde-b6c58e44dac6 From: sip:airtel-trunk@172.31.39.194;tag=687178ab-9f97-44ae-8d8d-657e3102ae4c To: sip:AIRTEL_IP Contact: sip:airtel-trunk@Aserisk_IP:5060 Call-ID: 468d363b-1abe-4e5f-ab37-ef8d11eedd4e CSeq: 20182 OPTIONS Supported: path Max-Forwards: 70 User-Agent: Asterisk PBX 20.6.0~dfsg+~cs6.13.40431414-2build5 Content-Length: 0 <--- Received SIP response (353 bytes) from UDPAIRTEL IP5095 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP Aserisk_IP:5060;received=Aserisk_IP;branch=z9hG4bKPjc250483b-7d20-49b6-bdde-b6c58e44dac6;rport=5060 From: sip:airtel-trunk@172.31.39.194;tag=687178ab-9f97-44ae-8d8d-657e3102ae4c To: sip:AIRTEL_IP;tag=aprqgshnalgh3sn28-sdjpp2000gjcd Call-ID: 468d363b-1abe-4e5f-ab37-ef8d11eedd4e CSeq: 20182 OPTIONS <--- Transmitting SIP request (1026 bytes) to UDP143.223.91.1755060 ---> INVITE sip:Number@livekit_id.sip.livekit.cloud SIP/2.0 Via: SIP/2.0/UDP Aserisk_IP:5060;rport;branch=z9hG4bKPja1fed995-c774-4934-b59e-7a2393ed8112 From: "T NAME" sip:+91NUMBER@172.31.39.194;tag=b930463a-cf32-4b2e-a224-6148c289c97c To: sip:Number@livekit_id.sip.livekit.cloud Contact: sip:asterisk@Aserisk_IP:5060 Call-ID: 877baacb-7cf5-4c36-8083-618fbe15ae3d CSeq: 30849 INVITE Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub, histinfo Session-Expires: 1800 Min-SE: 90 Max-Forwards: 70 User-Agent: Asterisk PBX 20.6.0~dfsg+~cs6.13.40431414-2build5 Content-Type: application/sdp Content-Length: 263 v=0 o=- 1190881317 1190881317 IN IP4 Aserisk_IP s=Asterisk c=IN IP4 Aserisk_IP t=0 0 m=audio 23530 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:140 a=sendrecv <--- Received SIP request (350 bytes) from UDPAIRTEL IP5095 ---> OPTIONS sipAserisk IP5060 SIP/2.0 Via: SIP/2.0/UDP AIRTEL_IP:5095;branch=z9hG4bKasbc1200d0n3frm751c0 Call-ID: a32d0ce504cb4528ea30323bb8dfabfd0600022r73500@AIRTEL_IP To: sip:ping@Aserisk_IP From: sip:ping@AIRTEL_IP;tag=53be9ae9bb9609d32f4b284fab3ab8630022r73 Max-Forwards: 70 CSeq: 569807 OPTIONS Content-Length: 0 <--- Transmitting SIP response (928 bytes) to UDPAIRTEL IP5095 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP AIRTEL_IP:5095;rport=5095;received=AIRTEL_IP;branch=z9hG4bKasbc1200d0n3frm751c0 Call-ID: a32d0ce504cb4528ea30323bb8dfabfd0600022r73500@AIRTEL_IP From: sip:ping@AIRTEL_IP;tag=53be9ae9bb9609d32f4b284fab3ab8630022r73 To: sip:ping@Aserisk_IP;tag=z9hG4bKasbc1200d0n3frm751c0 CSeq: 569807 OPTIONS Accept: application/sdp, application/xpidf+xml, application/cpim-pidf+xml, application/dialog-info+xml, application/simple-message-summary, application/pidf+xml, application/pidf+xml, application/dialog-info+xml, application/simple-message-summary, message/sipfrag;version=2.0 Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub Accept-Encoding: identity Accept-Language: en Server: Asterisk PBX 20.6.0~dfsg+~cs6.13.40431414-2build5 Content-Length: 0 <--- Transmitting SIP request (1026 bytes) to UDP143.223.91.1755060 ---> INVITE sip:Number@livekit_id.sip.livekit.cloud SIP/2.0 Via: SIP/2.0/UDP Aserisk_IP:5060;rport;branch=z9hG4bKPja1fed995-c774-4934-b59e-7a2393ed8112 From: "T NAME" sip:+91NUMBER@172.31.39.194;tag=b930463a-cf32-4b2e-a224-6148c289c97c To: sip:Number@livekit_id.sip.livekit.cloud Contact: sip:asterisk@Aserisk_IP:5060 Call-ID: 877baacb-7cf5-4c36-8083-618fbe15ae3d CSeq: 30849 INVITE Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub, histinfo Session-Expires: 1800 Min-SE: 90 Max-Forwards: 70 User-Agent: Asterisk PBX 20.6.0~dfsg+~cs6.13.40431414-2build5 Content-Type: application/sdp Content-Length: 263 v=0 o=- 1190881317 1190881317 IN IP4 Aserisk_IP s=Asterisk c=IN IP4 Aserisk_IP t=0 0 m=audio 23530 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:140 a=sendrecv <--- Transmitting SIP request (1026 bytes) to UDP143.223.91.935060 ---> INVITE sip:Number@livekit_id.sip.livekit.cloud SIP/2.0 Via: SIP/2.0/UDP Aserisk_IP:5060;rport;branch=z9hG4bKPj6a904884-629f-44d0-b445-3cfea5b358ac From: "T NAME" sip:+91NUMBER@172.31.39.194;tag=b930463a-cf32-4b2e-a224-6148c289c97c To: sip:Number@livekit_id.sip.livekit.cloud Contact: sip:asterisk@Aserisk_IP:5060 Call-ID: 877baacb-7cf5-4c36-8083-618fbe15ae3d CSeq: 30850 INVITE Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub, histinfo Session-Expires: 1800 Min-SE: 90 Max-Forwards: 70 User-Agent: Asterisk PBX 20.6.0~dfsg+~cs6.13.40431414-2build5 Content-Type: application/sdp Content-Length: 263 v=0 o=- 1190881317 1190881317 IN IP4 Aserisk_IP s=Asterisk c=IN IP4 Aserisk_IP t=0 0 m=audio 23530 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:140 a=sendrecv <--- Received SIP response (604 bytes) from UDP143.223.91.935060 ---> SIP/2.0 100 Processing Record-Route: sip:10.61.98.132:40004;sipfe;transport=tcp;lr Record-Route: sip:143.223.91.93:5060;sipfe;transport=udp;lr Via: SIP/2.0/UDP Aserisk_IP:5060;rport;branch=z9hG4bKPj6a904884-629f-44d0-b445-3cfea5b358ac From: "T NAME" sip:+91NUMBER@172.31.39.194;tag=b930463a-cf32-4b2e-a224-6148c289c97c To: sip:Number@livekit_id.sip.livekit.cloud;tag=SCL_c9wngUPHRVH4 Call-ID: 877baacb-7cf5-4c36-8083-618fbe15ae3d CSeq: 30850 INVITE Content-Length: 0 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Max-Forwards: 70 <--- Received SIP response (601 bytes) from UDP143.223.91.935060 ---> SIP/2.0 180 Ringing Record-Route: sip:10.61.98.132:40004;sipfe;transport=tcp;lr Record-Route: sip:143.223.91.93:5060;sipfe;transport=udp;lr Via: SIP/2.0/UDP Aserisk_IP:5060;rport;branch=z9hG4bKPj6a904884-629f-44d0-b445-3cfea5b358ac From: "T NAME" sip:+91NUMBER@172.31.39.194;tag=b930463a-cf32-4b2e-a224-6148c289c97c To: sip:Number@livekit_id.sip.livekit.cloud;tag=SCL_c9wngUPHRVH4 Call-ID: 877baacb-7cf5-4c36-8083-618fbe15ae3d CSeq: 30850 INVITE Content-Length: 0 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Max-Forwards: 70 -- PJSIP/livekit-inbound-000001f7 is ringing <--- Transmitting SIP response (851 bytes) to UDPAIRTEL IP5095 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP AIRTEL_IP:5095;rport=5095;received=AIRTEL_IP;branch=z9hG4bKepa77r00d8dj6kjv46j0.1 Call-ID: LU-1778186401457159-1639418374@bcf.kkp30.ims.mnc045.mcc404.3gppnetwork.org From: "T NAME" sip:+91NUMBER@AIRTEL_IP;user=phone;tag=65e8d793-69fcf8a11b3fbd6e-0020-01-ic-ibo-lucentIBCF-060706 To: sip:+91Number@Aserisk_IP;user=phone;tag=a09ba6cc-9937-4321-b846-973574759095 CSeq: 1 INVITE Server: Asterisk PBX 20.6.0~dfsg+~cs6.13.40431414-2build5 Contact: sip:Aserisk_IP:5060 Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER P-Asserted-Identity: sip:+91Number@Aserisk_IP;user=phone Remote-Party-ID: sip:+91Number@Aserisk_IP;user=phone;party=called;privacy=off;screen=no Content-Length: 0
<--- Received SIP request (604 bytes) from UDPAIRTEL IP5095 ---> CANCEL sip:+91Number@Aserisk_IP:5060;user=phone;tgrp=N45N01;trunk-context=ims.mnc045.mcc404.3gppnetwork.org SIP/2.0 Via: SIP/2.0/UDP AIRTEL_IP:5095;branch=z9hG4bKepa77r00d8dj6kjv46j0.1 CSeq: 1 CANCEL From: "T NAME" sip:+91NUMBER@AIRTEL_IP:5060;transport=udp;user=phone;tag=65e8d793-69fcf8a11b3fbd6e-0020-01-ic-ibo-lucentIBCF-060706 To: sip:+91Number@Aserisk_IP:5060;transport=udp;user=phone Call-ID: LU-1778186401457159-1639418374@bcf.kkp30.ims.mnc045.mcc404.3gppnetwork.org Max-Forwards: 58 Content-Length: 0 Reason: Q.850;cause=16;text="Call Terminated" <--- Transmitting SIP response (529 bytes) to UDPAIRTEL IP5095 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP AIRTEL_IP:5095;rport=5095;received=AIRTEL_IP;branch=z9hG4bKepa77r00d8dj6kjv46j0.1 Call-ID: LU-1778186401457159-1639418374@bcf.kkp30.ims.mnc045.mcc404.3gppnetwork.org From: "T NAME" sip:+91NUMBER@AIRTEL_IP;user=phone;tag=65e8d793-69fcf8a11b3fbd6e-0020-01-ic-ibo-lucentIBCF-060706 To: sip:+91Number@Aserisk_IP;user=phone;tag=a09ba6cc-9937-4321-b846-973574759095 CSeq: 1 CANCEL Server: Asterisk PBX 20.6.0~dfsg+~cs6.13.40431414-2build5 Content-Length: 0
t
Thanks — I reviewed the SIP trace you shared (INVITE retransmits, later 100/180 from 143.223.91.93 showing Record-Route through sipfe). That confirms the delayed provisionals are coming from LiveKit front-end (sipfe) and that calls proceed normally after the provisional responses arrive. To investigate in LiveKit Cloud I need two final pieces so I can correlate your call with our logs: • LiveKit Cloud region/edge (or the region shown in your LiveKit dashboard for this SIP trunk). • One or two affected call timestamps in UTC (or the Call-ID/CSeq from the trace). For the trace above the Call-ID is: 877baacb-7cf5-4c36-8083-618fbe15ae3d and CSeq 30850 INVITE — please confirm the UTC timestamp when that INVITE was sent. Also confirm: • SIP transport used to that LiveKit IP in the trace (UDP to 143.223.91.175 and UDP/TCP seen — which one does your Asterisk primarily use?). • Whether these delayed calls correlate with spikes in concurrent calls on your side. What I’ll do once you provide region + timestamp/Call-ID: I’ll look up the cloud front-end logs around that time to check for queued processing, upstream signaling delays, or retransmit handling and then share specific next steps (packet capture/clock sync checks, recommended transport/keepalive settings, or mitigation). , with interactive elements
g
Mumbai - india region
t
Thanks — that’s helpful. To correlate with LiveKit Cloud logs in the Mumbai region, please confirm the UTC timestamp (date and time) when the INVITE with Call-ID 877baacb-7cf5-4c36-8083-618fbe15ae3d / CSeq 30850 was sent. If you don’t have the exact second, an approximate UTC time window (±2–5 minutes) is fine. Also confirm which transport your Asterisk primarily uses to LiveKit for that call (UDP or TCP). , with interactive elements
g
UDP
t
Thanks — great, UDP and Mumbai noted. Please provide the UTC timestamp (date and time) of the INVITE with Call-ID 877baacb-7cf5-4c36-8083-618fbe15ae3d / CSeq 30850 (an approximate ±2–5 minute window is fine). Once I have that I’ll check LiveKit Cloud front-end logs for that interval and report what’s causing the 2–5s provisional-response delay and recommended next steps. , with interactive elements
g
I figured out the issue
one of your node is down
t
I don't have the answer you're looking for. You could also try asking your question: • in one of the other Slack channels or • to https://deepwiki.com/livekit/livekit_composite which is trained on all LiveKit source code If you find the answer, please post it here to help others!
g
143.223.91.175 is down ad resuling in failures
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