Freeze the SIP Dashboard object and assigned domain
Vonage documents SIP Trunking alongside Programmable SIP, Voice API and BYOC paths. Record the current trunk type, assigned regional domain and inbound-region preference before applying a configuration written for another object. An example hostname is not a substitute for the value assigned in the authorized dashboard.
Keep outbound authorization separate from inbound delivery. The current trunk flow supports digest credentials and/or an IP ACL for customer-originated calls. The reviewed workflow does not establish a general REGISTER requirement, so a missing registration light is not proof that inbound delivery failed.
Documented service facts
Reviewed 2026-10-18. Compare SIP trunk services →
Outbound authentication
DocumentedThe current SIP Dashboard documentation supports digest authentication and/or an IP Access Control List. Digest uses a trunk user key and a 12-to-64-character password meeting the documented character-class rules; those trunk credentials are separate from the account API key and secret.
SIP registration
Not verifiedThe reviewed trunk workflow uses a Vonage-created regional domain, digest or ACL authorization and configured SIP URIs. It does not establish a REGISTER requirement; confirm the selected trunk and PBX guide before treating registration state as a health signal.
Inbound delivery
DocumentedLink each Vonage number to the intended SIP trunk, then preserve trunk type, inbound region, customer SIP URI, priority, timeout, transport and PBX match as distinct evidence. Linking can replace an earlier application or domain association.
Inbound failover
DocumentedCustomer SIP URIs accept unique priorities from 0 to 100, where 0 is highest, and timeouts from 2,000 to 20,000 ms; an empty timeout uses 5,000 ms. Equal priorities do not guarantee order. The reviewed public pages do not publish one universal SIP-response advancement matrix.
Number and regional availability
Not verifiedThe dashboard supports buying and linking numbers and offers regional SIP processing choices. Exact country, locality, number type, porting, regulatory and inbound-supplier capacity for the intended account and date require direct confirmation.
Audio codecs
DocumentedVonage currently lists PCMA, PCMU, iLBC, G.729 without annex B or D, G.722 and Speex16, and documents RFC 4733 DTMF plus RFC 4028 session timers. Confirm the PBX allow list, negotiated SDP and DTMF on each direction.
T.38 fax
Not verifiedT.38 support, direction, fallback behavior and number eligibility were not established in the reviewed SIP Trunking references. Voice codec support is not evidence of fax compatibility.
Signaling and media encryption
DocumentedVonage documents SIP over TLS 1.2 on port 5061 and RTP or SRTP for media, with AES_CM_128_HMAC_SHA1_80 as the listed SRTP suite. The technical page says Vonage does not validate the user agent’s client certificate, so a successful TLS connection is not proof of mutual certificate authentication. Confirm the protected customer-facing leg.
Call-start capacity
DocumentedA Vonage support article updated May 14, 2026 states a default maximum of 3 outbound calls per second, shared across Voice products under the primary API key, with excess attempts rejected rather than queued. It says concurrency is not limited at that account layer while inbound virtual numbers may have supplier-specific limits. Confirm the account’s current limit and any approved increase.
Emergency calling
Not verifiedThe reviewed SIP Trunking setup and technical pages do not establish the complete emergency-address, calling-identity, activation, routing or approved test procedure for an intended location. Obtain that workflow before deployment.
Pricing model
Not verifiedExact calling rates, number charges, emergency fees, taxes, support and commitments were not established for a specific account, country and date. Use a dated rate deck or quote for the intended traffic mix.
Administration and support
Not verifiedVoice Logs can filter inbound and outbound calls and expose call records for correlation. Retention, contracted support response, per-response retry behavior and recovery commitments remain unverified for a specific account.
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.
Prove the number-to-trunk link before URI order
The dashboard links one or more Vonage numbers to the trunk. Vonage warns that linking a number already associated with another domain or application overrides that earlier association and directs calls to the current domain. Preserve the dialed number, prior association, intended trunk and saved link as separate facts.
A number present in the account is not evidence that it selects this trunk. A saved trunk is not evidence that the number points to it, and a linked number is not evidence that any customer SIP URI is reachable.
Treat priority and timeout as configuration, not a guarantee
The dashboard accepts several customer SIP URIs. Priority uses unique values from 0 through 100, with 0 highest; equal priorities do not guarantee delivery order. Timeout accepts 2,000 through 20,000 milliseconds and defaults to 5,000 milliseconds when left empty. The URI field accepts a domain or IPv4 address with optional user and port, and the guide says the field does not support IPv6.
Record the effective order and timeout for every URI. The reviewed public pages do not provide one response-code matrix promising advancement after every 4xx, 5xx or 6xx. Preserve no response, transport failure, timeout and each final SIP response as different outcomes.
Align transport, protection and PBX identification
Vonage lets the inbound URI select signaling transport; the dashboard documents TLS on port 5061 and non-TLS signaling on port 5060 by default, plus an SRTP choice. Those settings must align with the customer listener, firewall and loaded PBX objects.
For Asterisk or FreePBX, keep transport, endpoint identification, authorization, Request-URI, called-number normalization and the first matching inbound route separate. Vonage’s published examples are starting points, not proof that copied IP addresses, codecs or object syntax remain appropriate for every version or account.
Join provider, network, PBX and media evidence
Use one timestamp, timezone and call identity to join Vonage Voice Logs with SIP signaling, authorized edge capture and PBX logging. If the provider attempted one URI but no packet reached the expected interface, investigate the configured destination, DNS result, transport and firewall before changing the dialplan. If the INVITE arrived, preserve the matched endpoint and PBX response.
A 2xx response proves a completed SIP transaction, not the intended endpoint, announcement, queue or two-way audio. Preserve SDP, selected codec, RTP endpoints, packet counters and decoded audio separately. TLS and SRTP configuration are not evidence of observed negotiation.
Validate a bounded failover and recovery matrix
On an authorized noncritical number, test the normal primary path, a deliberately unreachable primary, the next eligible URI, primary recovery and return to the preferred order. Test a reachable destination that returns a final SIP rejection separately. Record every attempt, elapsed time and caller outcome, then restore the baseline.
Confirm number eligibility, inbound supplier limits, account CPS, emergency workflow, data handling, retention, support entitlement, rates and change approvals through current account evidence. No account, number, trunk, URI, failover event, PBX, packet, protected session, codec, DTMF event, media stream, emergency workflow, price, availability or compatibility was tested for this guide.
Sources & applicability
Primary references for the technical details above. Operational examples and planning checklists are VoIP.info editorial guidance.
- Vonage SIP technical details ↗
- Vonage SIP Dashboard guide ↗
- Vonage codec support article ↗
- Vonage calls-per-second support article ↗
- Vonage SIP Trunking product page ↗
- Vonage Asterisk PJSIP configuration ↗
- Vonage FreePBX configuration ↗
- Vonage Voice Logs guide ↗
Examples require adaptation to your topology. No live PBX or hardware testing is claimed.