Hi Vapi Support, We’re running an Asterisk (PJSIP) SIP trunk from a server located in Singapore and currently pointing our trunk to sip.vapi.ai. We’re seeing SIP qualify RTT around ~200–220ms, which appears consistent with being routed to the US. Evidence from the Asterisk host: dig +short sip.vapi.ai returns: 44.229.228.186 44.238.177.138 Asterisk pjsip show endpoints / pjsip show aor shows Status: Avail with RTT ~200ms. Could you please provide: The APAC/Singapore (or closest APAC) SIP ingress hostname we should use instead of sip.vapi.ai to reduce latency (target RTT < 100ms), and The corresponding SIP ingress IP list for that APAC POP so we can update our Asterisk identify matches / firewall allowlist. If an APAC POP isn’t available for SIP trunk ingress today, please confirm and advise the best alternative for Singapore-based deployments. Thanks!

Vapi does not currently have an APAC (or Singapore) SIP ingress point of presence. All SIP infrastructure is US-based (AWS US-West-2 region). This applies to the main

sip.vapi.ai

endpoint as well as all dedicated SIP endpoints. The IPs you're seeing from

dig +short sip.vapi.ai

(

44.229.228.186

,

44.238.177.138

) are indeed US-West (Oregon) — so the ~200-220ms RTT you're measuring via PJSIP qualify from Singapore is consistent and expected for that routing path. 2. No APAC POP IP List Available Since there is no APAC SIP POP, there are no regional IPs to provide for firewall allowlisting or PJSIP identify matches at this time. 3. Regional SIP Is Being Actively Developed I can confirm that SIP regionalization (including APAC/ANZ) is actively being worked on internally. While I can't provide a specific ETA, it is a tracked priority. Once APAC SIP ingress is available, you would get a regional hostname (e.g., similar to how other Vapi services have regional endpoints) with corresponding IPs for your firewall and identify match configuration. Alternatives to Consider in the Meantime: • WebRTC/Daily transport: If any of your call flows can use web-based audio (browser or app), Vapi's WebRTC transport via [Daily.co](http://Daily.co) may route through closer media servers and could offer lower latency than SIP for APAC users. • Accept current RTT for SIP: 200-220ms RTT, while not ideal, is within functional range for voice calls. The actual one-way audio latency would be roughly half that (~100-110ms), which is generally acceptable for conversational voice, though you may notice slight delay. • Enterprise SIP options: If low-latency APAC SIP is critical for your use case, I'd recommend reaching out to Vapi's sales team to discuss enterprise options and get visibility into the APAC SIP rollout timeline. I hope this helps clarify the situation. Let me know if you have any other questions!