Name the current product and control path
The current Voice API reference documents a static SIP trunk and a Callback API path. A static trunk can contain termination toward Sinch, origination from Sinch, or both. The callback path instead lets an application respond to an Incoming Call Event and use a ConnectSIP action. Record the application, path, direction and owner before diagnosing credentials or routing.
Do not import an older Elastic SIP Trunking endpoint model into this Voice API path without current account evidence. A dashboard label, application key, forwarding URI and callback URL belong to different controls.
Documented service facts
Reviewed 2026-10-19. Compare SIP trunk services →
Outbound authentication
DocumentedCurrent Voice API static-trunk termination authenticates with the application key, application secret and selected regional realm. REST and callback request authentication are separate controls, and the Numbers API uses different credentials.
SIP registration
DocumentedThe current Voice API static-trunk reference documents regional FQDN termination and configured SIP forwarding rather than a customer REGISTER workflow. Do not apply older Elastic SIP Trunking registered-endpoint evidence to this path without current account documentation.
Inbound delivery
DocumentedVoice API origination sends Sinch calls to the customer through SIP forwarding or Callback API ConnectSIP control. Static configuration uses a separate origination application with assigned numbers and a SIP URI; `{number}` can be replaced by the inbound number.
Inbound failover
DocumentedRegional termination FQDNs resolve to several Sinch public edges distributed across Availability Zones. The reviewed public reference does not establish a universal customer-destination retry matrix, callback retry policy, timer, restoration interval or established-call survival result.
Number and regional availability
Not verifiedSinch presents number ordering, porting and broad regional connectivity, but exact country, number type, locality, portability and regulatory eligibility for the intended account and date require direct confirmation.
Audio codecs
DocumentedThe current Voice API trunk reference lists G.711u, G.711a, Opus and G.729. Confirm the ordered region, selected call path, SDP offer and answer, transcoding boundary and observed packets for the actual call.
T.38 fax
Not verifiedT.38 support, direction, fallback behavior and number eligibility were not established for the exact SIP trunk path in this review. Do not infer fax compatibility from voice codec support.
Signaling and media encryption
DocumentedThe current reference recommends TLS 1.2, lists supported cipher suites and Let's Encrypt trust requirements, and says SRTP is offered and accepted when TLS carries signaling. Verify hostname, chain, handshake, SDP crypto and each media direction.
Call-start capacity
Not verifiedProduct material describes elastic or unlimited concurrent capacity without per-session charges, but the reviewed sources do not establish an account call-start CPS limit, burst policy or over-limit response. Obtain concurrency and CPS separately.
Emergency calling
Not verifiedSinch product material mentions E911 support, but this review did not establish the complete address, calling-identity, activation, routing or provider-approved test workflow for the selected account and location.
Pricing model
Not verifiedThe Elastic SIP Trunking page describes pay-as-you-go operation, while exact rates, number charges, emergency fees, taxes, support and commitments remain account- and region-specific. A dated quote is required.
Administration and support
DocumentedVoice API CDRs expose call ID, time, result, reason, duration, identities and application key. Callback events also carry call ID and UTC time. Preserve minimal redacted correlation evidence; a CDR or callback receipt is not customer-edge, PBX or media proof.
SMS and MMS
Not verifiedMessaging behavior, number eligibility, sender registration and pricing were not established as part of this SIP trunk record. Keep voice routing and messaging enablement as separate account decisions.
Configure termination and origination as separate flows
Sinch defines termination as the customer SIP infrastructure sending a call toward Sinch. It requires a regional termination FQDN, application key as authentication name, application secret and a matching regional realm. The called number must be E.164-formatted, and the asserted caller identity must be a Sinch DID on the account or a number verified in the portal.
Origination is Sinch sending a call toward the customer SIP infrastructure. Static configuration uses SIP forwarding and a valid SIP URI; {number} can be replaced by the inbound number. The reference says static trunks use one application for termination and another application for origination, with numbers assigned to the origination application. One successful direction does not prove the other application.
Resolve regional FQDNs without pinning one edge
The current reference publishes separate Europe, North America, South America, Southeast Asia and Australia termination FQDNs and says each resolves to several public edges across Availability Zones. Preserve the configured region, DNS answers, selected address, transport and response. Use the FQDN rather than a proxy address unless Sinch support supplies another design.
The page also publishes current SIP signaling and RTP ranges. Treat those as dated service documentation, not a timeless copied firewall object. Reconcile the live page, ordered region and change record before editing network policy.
Separate callback control from SIP forwarding
With SIP forwarding, the origination application carries the destination URI. With callback control, Sinch sends an Incoming Call Event to the configured backend and expects a valid response; the callback reference says a missing response within its timeout produces an error and disconnects the call. Preserve the call ID, UTC timestamp, application key, inbound number, callback request, application response and resulting SIP leg.
A callback receipt is not a SIP INVITE at the customer edge, and a generated SIP INVITE is not proof that the PBX identified, routed or answered it. Custom call headers can support correlation, but they can contain customer data and have documented size limits; retain only authorized, redacted evidence.
Prove signaling and media independently
Sinch documents UDP and TCP for termination and SIP forwarding URIs that can select UDP, TCP or TLS. Its secure-trunk section recommends TLS 1.2, lists cipher suites, identifies Let's Encrypt certificate trust requirements and says SRTP is offered and accepted when TLS carries signaling. Verify hostname, chain, trust store, handshake, SIP transport, SDP crypto, RTP endpoints and both media directions.
The current page lists G.711u, G.711a, Opus and G.729, SIP ports 5060 and 5061, and regional media ranges with UDP ports 10000 through 60000. Those documented values do not prove one PBX negotiated a codec, accepted SRTP or received media.
Correlate one bounded acceptance trace
For one authorized number and time window, record product, applications, direction, region, DNS answers, source and destination, transport, call ID, CDR result, callback event if used, customer-edge receipt, PBX response, answer, SDP and both media directions. Sinch CDRs expose a call identifier, call time, result, reason, duration, identities, application key and other fields; preserve only the minimum redacted evidence needed.
Confirm current number eligibility, portability, emergency workflow, messaging, fax, DTMF, codec availability, capacity, call-start rate, pricing and support for the exact account. Use the paired diagnostic for a directional failure.
Diagnose a directional Sinch Voice API failure
Use the Sinch directional diagnostic to join application ownership, regional routing, forwarding or callback control, PBX handling and media.
Sources & applicability
Primary references for the technical details above. Operational examples and planning checklists are VoIP.info editorial guidance.
- Sinch Voice API SIP Trunking reference ↗
- Sinch Voice API callback reference ↗
- Sinch Voice API authentication reference ↗
- Sinch Voice API call headers ↗
- Sinch Voice API call detail records ↗
- Sinch Voice API release notes ↗
- Sinch Elastic SIP Trunking product overview ↗
- Sinch BYOC SIP Trunking overview ↗
Examples require adaptation to your topology. No live PBX or hardware testing is claimed.