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sure, Here is the full debug SIP logs from the ses...
# ask-ai
c
sure, Here is the full debug SIP logs from the session. ``````
Untitled
t
Thanks — I don’t see the SIP debug logs in your last message (the code block is empty). I can dig into them and point to the exact signaling or SDP problem, but I need the actual SIP capture or log snippet. Could you please paste (or attach) one of these, redacting any secrets like auth credentials, tokens, or full phone numbers if you prefer: • The full SIP INVITE → responses sequence (INVITE, 100/180/183, 401/407 challenge if present, re-INVITE with Authorization, 200 OK or CANCEL/487), including the SDP bodies. • Or a tcpdump/pcap of a single failed call (preferred). If you attach a pcap, tell me the approximate timestamp and I’ll point to the relevant packets. • The Twilio Call SID for that attempt (or timestamp) so we can correlate with Twilio logs. If you can’t share full logs, please at minimum copy these parts from the failing call: • The INVITE request headers and the SDP body your agent sends. • Any 401/407 challenge response from Twilio. • The CANCEL/487 (or final response) from Twilio. If you’d like, here’s a short checklist to make it easier: • Include timestamps for each SIP message. • Redact Authorization headers or long credentials, but keep the header names (e.g., Authorization: REDACTED). • Replace full phone numbers with +91XXXXXXXXXX (as you did earlier) if you prefer. Once you post the capture/snippet, I’ll: • Check timing between INVITE → 180/200 and any retransmits. • Validate the SDP (presence of v=, o=, s=, c=, t=, m= lines and codecs). • Identify whether Twilio’s challenge is answered correctly. • Recommend precise fixes (adjust timeouts, pre-warm agent, correct SDP lines, or trunk settings). I’ll wait for the logs and will walk through them step‑by‑step with you.
c
SIP logs.txt
SIP logs.txt
t
Thanks — I can help dig in, but I don’t see any SIP logs or pcap attached. To diagnose why calls ring a few times then disconnect unless answered quickly, I need one failing call’s SIP exchange or a packet capture. Could you please provide one of the following (redact credentials or full numbers if you prefer)? • Preferred: a pcap/tcpdump of one failing call (single file). Tell me the approximate timestamp of the call so I can find the relevant packets. ◦ Example capture command: tcpdump -i any -s 0 -w sip_capture.pcap port 5060 or host <twilio-ip> • Or: the full SIP message sequence for a failing call (INVITE, any 1xx/18x, 401/407 challenge, re-INVITE with Authorization, 200 OK or CANCEL/487), including SDP bodies. If you redact Authorization values, keep header names and timestamps. • Or at minimum: the INVITE your agent sent (headers + SDP) and the Twilio final response (CANCEL/487 or timeout lines) with timestamps. If you can’t share the full capture, please paste these specific items: • Timestamped lines showing when INVITE arrived and when LiveKit sent 180/200 (or if it didn’t). • The SDP your agent sends (or the m/c/a lines) so I can check for formatting or candidate issues. • Any Twilio Call SID or Twilio log entry timestamp for the same call. What I’ll check once you share: • Timing between INVITE → 180 → 200 and any retransmits (to confirm signaling timeout). • Whether the agent answers Twilio’s 401/407 correctly. • SDP compliance (presence of v=, o=, s=, c=, t=, m=, a=rtpmap, ICE candidates). • Any CANCEL/487 pattern that shows Twilio aborting for timeout. If you prefer, tell me which of the three options above you’ll provide and I’ll give exact instructions for extracting the snippet or pcap. I’ll stay with you and walk through the capture step‑by‑step once you post it.
c
t
Thanks — I can’t access Slack links from here, but I can definitely help if you paste the SIP logs or attach the pcap directly in this chat. What I need from you (choose one option) • Preferred: a tcpdump/pcap file of one failing call (single file). Tell me the approximate timestamp of the call so I can find the relevant packets. ◦ Example capture command: tcpdump -i any -s 0 -w sip_capture.pcap port 5060 • OR: the full SIP message sequence for one failing call (INVITE, any 1xx like 180/183, 401/407 challenge, re-INVITE with Authorization, 200 OK or CANCEL/487), including the SDP bodies. Redact credentials if needed but keep header names and timestamps. • OR (minimum): the INVITE your agent sent (headers + SDP) and the Twilio final response (CANCEL/487 or timeout lines) with timestamps. If you’re unsure how to extract the SIP messages, here are two quick ways: 1. Capture with tcpdump and attach the pcap: ◦ tcpdump -i any -s 0 -w sip_capture.pcap port 5060 1. If logs already contain SIP text, copy the block for a single call and paste it here. Replace sensitive values like tokens or full phone numbers with redacted text (e.g., +91XXXXXXXXXX). What I’ll check once you provide it • Timing between INVITE → 1xx (180) → 200 OK and any retransmits (to confirm signaling timeout). • Whether the agent answers Twilio’s 401/407 challenge correctly. • SDP compliance (presence and order of v=, o=, s=, c=, t=, m=, a=rtpmap lines) and candidate addresses. • Any CANCEL/487 patterns showing Twilio aborting for timeout. If you prefer, paste a short redacted INVITE + the Twilio response now and I’ll start the analysis immediately. I’ll stay with you until we find the root cause.
c
SIP logs.txt
SIP logs.txt
t
Thanks — I can see you’re still having trouble sharing the SIP logs (the last messages were empty or links I can’t access). I want to keep helping — here’s the simplest, lowest-effort way to get me exactly what I need and keep your credentials private. What I need to diagnose the ringing-then-disconnect behavior • One failing call’s SIP exchange (INVITE → 1xx/180 → 401/407 if present → re-INVITE with Authorization → final 200 OK or CANCEL/487), including the SDP bodies and timestamps. ◦ OR a tcpdump/pcap of a single failing call (preferred). If you attach a pcap, tell me the approximate timestamp of the call. • Redact sensitive fields if you prefer (Authorization header contents, full phone numbers). Keep header names and timestamps intact. How to provide it (pick one) 1. Attach a pcap (preferred) ◦ Run on your LiveKit/SIP agent host: tcpdump -i any -s 0 -w sip_capture.pcap port 5060 • Make one failing call, stop capture, attach the pcap file here, and tell me the call timestamp. 1. Paste the SIP message sequence (easier if you already have logs) ◦ Copy-paste the INVITE request and any responses for one failing call into this chat. ◦ Replace sensitive tokens or digits with placeholders (e.g., Authorization: REDACTED, +91XXXXXXXXXX). ◦ Keep SDP bodies intact (redact only credentials). 1. If you can’t capture, paste these exact items ◦ Time when INVITE was received by your server (timestamp). ◦ Time when your service sent 180 Ringing (if any) and when it sent 200 OK (if any). ◦ Any 401/407 challenge lines and whether your agent re-sent the INVITE with an Authorization header. ◦ The final Twilio response (CANCEL/487) with timestamp. What I’ll do once you share it • Check precise timing between INVITE → 180 → 200 and any retransmits to confirm signaling timeout. • Validate the SDP for required lines (
v=
, 
o=
, 
s=
, 
c=
, 
t=
, 
m=
) and codec attributes. • Verify the auth challenge/response flow and identify if credentials or re-INVITE behavior is the problem. • Point out the exact line(s) to change (e.g., increase a timeout, pre-warm agent, correct SDP field, or fix Authorization handling). Quick troubleshooting to try now (if you want to test immediately) • Pre-warm an agent worker (or keep one running) to eliminate cold-start delays and re-test. • Temporarily increase any per-call timeouts on the agent/RPC layer to see if the problem is just a signaling delay. • Confirm Twilio trunk allows your IP or that the digest auth credentials match (since a 407 appeared earlier). Tell me which option above you’ll use and paste or attach the file/snippet. I’ll walk through it line-by-line with you.
c
Copy code
42:38.466Z INFOsip sip/outbound.go:454 cannot play dial tone   {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdrzzVy", "callID": "SCL_GQ
42:38.467Z DEBUG   Client transaction destroyed{"nodeID": "NE_sock2aft7CoQ", "caller": "transaction.Layer", "tx": "z9hG4bK.Iip8wMzOyEaujYSK__INVITE"}
42:38.467Z INFO sip sip/outbound.go:620 SIP invite failed{"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdrzzVy", "callID": "SCL_GQ
42:38.467Z INFO sip sip/outbound.go:378 SIP call failed {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdrzzVy", "callID": "SCL_GQpq7AjLgN
42:38.467Z WARN sip sip/outbound.go:315 Closing outbound call with error {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdrzzVy", "
42:38.469Z WARN sip sip/media_port.go:260   error encountered while clearing read deadline  {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdr
42:38.469Z INFO sip sip/media_port.go:439   media timeout loop stopped  {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdrzzVy", "callID":
42:38.469Z INFO sip sip/media.go:119 call statistics {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdrzzVy", "callID": "SCL_GQpq7AjLgN
42:38.469Z INFO sip service/service.go:265  SIP call ended  {"nodeID": "NE_sock2aft7CoQ", "callID": "SCL_GQpq7AjLgNqs", "reason": "invite-failed"}
42:38.651Z DEBUG   Unhandled sip response. UnhandledResponseHandler handler not added  {"nodeID": "NE_sock2aft7CoQ", "caller": "transaction.Layer", "
42:38.668Z INFO sip sip/client.go:271Creating SIP participant {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdrzzVy", "callID":
42:38.668Z DEBUG   sip sip/media_port.go:337   listening for media on UDP  {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdrzzVy", "callID":
42:38.668Z DEBUG   sip sip/media_port.go:384   media timeout disabled  {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdrzzVy", "callID": "SCL_GQ
42:38.672Z DEBUG   sip signalling/signaltransport_websocket.go:241 first message received  {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdr
42:38.673Z DEBUG   sip v2@v2.11.4-0/engine.go:1291 received offer for subscriber   {"nodeID": "NE_sock2aft7CoQ", "sipTrun
42:38.673Z DEBUG   sip v2@v2.11.4-0/engine.go:829  sending answer for subscriber   {"nodeID": "NE_sock2aft7CoQ", "sipTrun
2:38.675Z DEBUG   sip v2@v2.11.4-0/engine.go:526  ICE connected   {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdr
42:38.675Z DEBUG   sip v2@v2.11.4-0/engine.go:474  local ICE candidate {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_
42:38.684Z INFOsip sip/room.go:405 SIP participant joined room {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdrzzVy", "callID": "SCL_GQ
42:38.673Z DEBUG   sip sip/room.go:303 skipping track publish event - subscribed flag not set  {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdr
42:38.685Z INFOsip v2@v2.11.4-0/localparticipant.go:140 published track {"nodeID": "NE_sock2aft7CoQ", "sipTrun
42:38.686Z DEBUG   sip sip/outbound.go:579 SDP offer{"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdrzzVy", "callID": "SCL_GQpq7AjLgN
42:38.705Z DEBUG   sip v2@v2.11.4-0/transport.go:378starting to negotiate   {"nodeID": "NE_sock2aft7CoQ", "sipTrun
42:38.708Z DEBUG   sip v2@v2.11.4-0/engine.go:1281 successfully set publisher answer{"nodeID": "NE_sock2aft7CoQ", 
42:38.708Z DEBUG   sip v2@v2.11.4-0/engine.go:1305 remote ICE candidate{"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_
42:38.709Z DEBUG   sip v2@v2.11.4-0/engine.go:1305 remote ICE candidate{"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_
42:38.896Z DEBUG   Client transaction destroyed{"nodeID": "NE_sock2aft7CoQ", "caller": "transaction.Layer", "tx": "z9hG4bK.oKw7E8S2bwBnjaSr__INVITE"}
42:38.896Z INFOsip sip/outbound.go:881 auth requested  {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdrzzVy", "callID": "SCL_GQpq7AjLgN
42:40.096Z DEBUG   Unhandled sip response. UnhandledResponseHandler handler not added  {"nodeID": "NE_sock2aft7CoQ", "caller": "transaction.Layer", "
42:40.634Z DEBUG   sip v2@v2.11.4-0/room.go:1113   ignoring incoming text stream due to no handler for topic {"nodeID
42:42.096Z DEBUG   Unhandled sip response. UnhandledResponseHandler handler not added  {"nodeID": "NE_sock2aft7CoQ", "caller": "transaction.Layer", "
42:42.684Z DEBUG   sip v2@v2.11.4-0/room.go:1113   ignoring incoming text stream due to no handler for topic {"nodeID
42:43.468Z INFOsip sip/outbound.go:454 cannot play dial tone   {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdrzzVy", "callID": "SCL_GQ
42:43.468Z DEBUG   Client transaction destroyed{"nodeID": "NE_sock2aft7CoQ", "caller": "transaction.Layer", "tx": "z9hG4bK.in3kj1DqJHHTTX8z__INVITE"}
42:43.468Z INFO sip sip/outbound.go:620 SIP invite failed{"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdrzzVy", "callID": "SCL_GQ
42:43.468Z INFO sip sip/outbound.go:378 SIP call failed {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdrzzVy", "callID": "SCL_GQpq7AjLgN
42:43.468Z WARN sip sip/outbound.go:315 Closing outbound call with error {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdrzzVy", "
42:43.469Z WARN sip sip/media_port.go:250   error encountered while reading UDP packets {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdr
42:43.469Z INFO sip sip/media_port.go:256   Stopped discarding packets  {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdrzzVy", "callID":
42:43.469Z WARN sip sip/media_port.go:260   error encountered while clearing read deadline  {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdr
42:43.469Z INFO sip sip/media.go:119 call statistics {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdrzzVy", "callID": "SCL_GQpq7AjLgN
42:43.469Z INFO sip service/service.go:265  SIP call ended  {"nodeID": "NE_sock2aft7CoQ", "callID": "SCL_GQpq7AjLgNqs", "reason": "invite-failed"}
42:43.469Z INFO sip sip/media_port.go:439   media timeout loop stopped  {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdrzzVy", "callID":
42:43.653Z DEBUG   Unhandled sip response. UnhandledResponseHandler handler not added  {"nodeID": "NE_sock2aft7CoQ", "caller": "transaction.Layer", "
42:43.670Z INFO sip sip/client.go:271Creating SIP participant {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdrzzVy", "callID":
42:43.670Z DEBUG   sip sip/media_port.go:337   listening for media on UDP  {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdrzzVy", "callID":
42:43.670Z DEBUG   sip sip/media_port.go:384   media timeout disabled  {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdrzzVy", "callID": "SCL_GQ
42:43.674Z DEBUG   sip signalling/signaltransport_websocket.go:241 first message received  {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdr
42:43.674Z DEBUG   sip v2@v2.11.4-0/engine.go:339  Using ICE servers{"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_
42:43.686Z INFO sip sip/room.go:405 SIP participant joined room {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdrzzVy", "callID": "SCL_GQ
42:43.675Z DEBUG   sip sip/room.go:303 skipping track publish event - subscribed flag not set  {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdr
42:43.687Z INFO sip v2@v2.11.4-0/localparticipant.go:140 published track {"nodeID": "NE_sock2aft7CoQ", "sipTrun
42:43.687Z DEBUG   sip sip/outbound.go:579 SDP offer{"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdrzzVy", "callID": "SCL_GQpq7AjLgN
42:43.706Z DEBUG   sip v2@v2.11.4-0/transport.go:378starting to negotiate   {"nodeID": "NE_sock2aft7CoQ", "sipTrun
42:43.707Z DEBUG   sip v2@v2.11.4-0/transport.go:378create offer{"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_
42:43.707Z DEBUG   sip v2@v2.11.4-0/engine.go:393  local ICE candidate {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_
42:43.708Z DEBUG   sip v2@v2.11.4-0/engine.go:393  local ICE candidate {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_
42:43.709Z DEBUG   sip v2@v2.11.4-0/engine.go:1281 successfully set publisher answer{"nodeID": "NE_sock2aft7CoQ", 
42:43.709Z DEBUG   sip v2@v2.11.4-0/engine.go:393  local ICE candidate {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_
42:43.711Z DEBUG   sip v2@v2.11.4-0/engine.go:526  ICE connected   {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdr
42:43.888Z DEBUG   Client transaction destroyed{"nodeID": "NE_sock2aft7CoQ", "caller": "transaction.Layer", "tx": "z9hG4bK.NiVfw9x24U0vaQX9__INVITE"}
42:43.888Z INFO sip sip/outbound.go:881 auth requested  {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdrzzVy", "callID": "SCL_GQpq7AjLgN
42:44.095Z DEBUG   Unhandled sip response. UnhandledResponseHandler handler not added  {"nodeID": "NE_sock2aft7CoQ", "caller": "transaction.Layer", "
42:44.674Z DEBUG   sip v2@v2.11.4-0/room.go:1113   ignoring incoming text stream due to no handler for topic {"nodeID
42:47.096Z DEBUG   Unhandled sip response. UnhandledResponseHandler handler not added  {"nodeID": "NE_sock2aft7CoQ", "caller": "transaction.Layer", "
42:47.773Z DEBUG   sip v2@v2.11.4-0/room.go:1113   ignoring incoming text stream due to no handler for topic {"nodeID
42:50.469Z INFO sip sip/outbound.go:454 cannot play dial tone   {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdrzzVy", "callID": "SCL_GQ
42:50.469Z DEBUG   Client transaction destroyed{"nodeID": "NE_sock2aft7CoQ", "caller": "transaction.Layer", "tx": "z9hG4bK.dwcQxZZrnjvc4CzJ__INVITE"}
42:50.469Z INFO sip sip/outbound.go:620 SIP invite failed{"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdrzzVy", "callID": "SCL_GQ
42:50.469Z INFO sip sip/outbound.go:378 SIP call failed {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdrzzVy", "callID": "SCL_GQpq7AjLgN
42:50.469Z WARN sip sip/outbound.go:315 Closing outbound call with error {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdrzzVy", "
42:50.470Z INFO sip sip/media.go:119 call statistics {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdrzzVy", "callID": "SCL_GQpq7AjLgN
42:50.470Z INFO sip sip/media_port.go:439   media timeout loop stopped  {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdrzzVy", "callID":
42:50.470Z WARN sip sip/media_port.go:250   error encountered while reading UDP packets {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdr
42:50.470Z INFO sip sip/media_port.go:256   Stopped discarding packets  {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdrzzVy", "callID":
42:50.470Z WARN sip sip/media_port.go:260   error encountered while clearing read deadline  {"nodeID": "NE_sock2aft7CoQ", "sipTrunk": "ST_XTFzHPdr
42:50.471Z INFO sip service/service.go:265  SIP call ended  {"nodeID": "NE_sock2aft7CoQ", "callID": "SCL_GQpq7AjLgNqs", "reason": "invite-failed"}
t
Thanks — this log helps a lot. I reviewed the snippets you pasted and here’s what they show and what to try next. What the logs show • Repeated sequences where an outbound INVITE is created, then shortly after: “auth requested” and later “SIP invite failed” / “invite-failed.” That matches the behavior you described: multiple rings (retries) then Twilio cancels. • The logs also show ICE connected, room/participant creation, and SDP offer/answer activity on the LiveKit side — media path appears to establish when the call is answered quickly. • The failures happen after Twilio issues an auth challenge (401/407) and the call ultimately times out or is cancelled; you also see “cannot play dial tone” and client transaction destroyed entries. Most likely root causes • Authentication challenge / re-INVITE either not being answered quickly enough or not being answered correctly (credentials, auth header formation). Twilio issues a challenge and if the re-INVITE (with Authorization) isn’t processed in time, Twilio may retry/cancel. • Signaling timing delay (agent/room creation or RPC/Twirp timeout) that causes Twilio to give up before the final response is sent. When you answer quickly, the final 200 OK happens fast enough and the call succeeds. • Less likely but possible: Twilio cancelling due to session timer or early-media policy, or transient network latency/retransmits. Targeted next steps (ordered, so you can try the easiest first) 1. Confirm Twilio trunk auth mode ◦ Tell me whether your Twilio trunk uses IP ACL (no digest auth) or digest credentials. If it’s digest, confirm the credentials configured in LiveKit/agent match Twilio’s credentials. 1. Check timing between INVITE and re-INVITE with Authorization ◦ From your SIP logs or a pcap, note timestamps for: ▪︎ Twilio INVITE, ▪︎ Twilio 401/407 (challenge), ▪︎ Your re-INVITE with Authorization, ▪︎ Any Twilio CANCEL/487. ◦ If the re-INVITE arrives > ~3–5s after the initial INVITE, Twilio may cancel. If it arrives quickly but still fails, the Authorization header or credentials may be malformed or wrong. 1. Look for Twirp / RPC timeouts on agent side ◦ Your earlier message showed a Twirp timeout. Ensure the create_sip_participant call finishes quickly. If room/agent creation is slow, pre-warm an agent or keep a ready worker to respond faster. 1. Capture a short SIP trace for one failing call ◦ Preferred: pcap of the exchange (port 5060). Example: tcpdump -i any -s 0 -w sip_capture.pcap port 5060 • I can analyze INVITE → 401/407 → re-INVITE → CANCEL/487 and the SDP. Share the pcap or paste the raw SIP messages (redact credentials) and I’ll inspect them. 1. Temporarily increase timeouts while debugging ◦ If you can, increase RPC/SIP transaction timeouts slightly so the agent has more time to answer the challenge while you debug. (Do this only as a diagnostic step.) 1. Verify response to the 401/407 challenge ◦ Make sure your agent: ▪︎ Sends the Authorization header with correct realm/nonce/response, ▪︎ Uses the correct username (often the trunk username) and method parameters, ▪︎ Retries the INVITE promptly. 1. Quick workaround to validate timing vs auth ◦ If you temporarily configure the Twilio trunk to accept your server IP (IP ACL) instead of digest auth, test calls. If they succeed reliably, the problem is definitely auth-handling or auth timing. What I can do next • If you paste one full INVITE and the corresponding 401/407 and your re-INVITE (redact secrets) I’ll examine the Authorization formatting and timing. • Or attach a pcap of a single failing call and tell me the approximate timestamp; I’ll analyze the sequence and call out exactly where the delay or malformed header is. • If you prefer, tell me whether your trunk is IP ACL or digest auth and whether you can pre-warm the agent — I can recommend exact config changes. If you share one of: • raw SIP INVITE + 401/407 + re-INVITE (redact Authorization body), or • the pcap file for one failing call (with timestamp), I’ll review line-by-line and tell you the precise change to make. I’ll stay with you until it’s resolved.
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