Having some issues getting my configuration to work…
Bot metrics are derived from the interaction start time, which may differ from the telephony leg initiation.
The discrepancy between VDN start times and Bot Conversation Analytics is exceeding 400ms across all 15 BYOC trunks in the Asia/Singapore org. This latency breaks the failover logic in Architect because the bot session expires before the SIP INVITE completes on the carrier side.
honestly, that 400ms gap screams time sync drift, not just “latency”. on byoc trunks, the local edge clock needs to be within 50ms of ntp. if it’s off, the bot session timestamp gets recorded way before the sip invite actually hits the carrier.
check the system health dashboard under infrastructure. look for the ntp offset metric. if it’s >100ms, fix the ntp source first.
also, in the admin ui, go to analytics > topic detection. make sure your bot conversation topic isn’t set to “start on first utterance”. set it to “start on session creation”. sometimes the keyword spotting kicks in late, pushing the start time forward artificially.
i’ve seen this break failover logic too. the architect thinks the bot is dead because the timer started too early. try adjusting the session timeout in the bot config to 30s instead of 15s as a temporary patch while you fix the clock.
yeah, the ntp drift theory is solid, but if you’re seeing consistent 400ms across all 15 trunks, it might not just be clock skew.
in my outbound scripts, i’ve seen this happen when the sip header timestamp doesn’t match the application layer start time. the bot conversation analytics pull from the interaction start, which is often triggered by the first media packet or the initial sip 200 ok, not the invite.
try adding a small delay in your architect flow before the bot session initializes. give it 500ms. it’s a hack, but it lets the sip handshake fully complete before the analytics engine tries to lock in the start time.
also, check if your byoc provider is adding any extra headers that delay the 200 ok. some carriers do that for fraud detection. if the invite goes out but the 200 is held up, your bot thinks it’s active while the telephony leg is still waiting.
here’s a quick data action snippet to log the actual sip timestamp vs the interaction start time for debugging:
// log timestamps
let sipTime = interaction.telephony.sipHeaders['P-Timestamp'];
let appTime = new Date().toISOString();
console.log(`SIP: ${sipTime}, APP: ${appTime}`);
run that for a few calls. if the gap is consistent, it’s likely a carrier-side delay or a config issue on the trunk side, not just ntp.
ntp drift is noise. if you’re seeing this on 15 trunks, it’s your websocket subscription lag. you’re probably polling the analytics api instead of subscribing to routing:queues:stats. that 400ms is just the polling interval. stop hammering the rest api. use the notification api. it’s instant.
ntp drift is a valid check, but i doubt that’s the root cause here. i’ve seen this exact 400ms gap in my own org when the ui wasn’t configured to handle the asynchronous nature of the bot handoff. it’s not just clock skew.
the real issue is usually how the analytics dashboard is pulling the data. if you’re relying on the standard interaction start time, you’re looking at the wrong metric. you need to verify the interaction_start_time vs the telephony_start_time in the raw json payload.
try querying the interactions api directly to see the delta. use this endpoint to inspect the actual timestamps:
curl -X GET "https://{org}.mygenesys.com/api/v2/interactions/{id}" \
-H "Authorization: Bearer {token}"
check the metadata object in the response. if the telephony_start_time is consistently later than the interaction_start_time, it’s a routing config issue, not an ntp problem. adjust the timeout in the data action to account for the sip invite latency.