WarmTransfer
sbc trunking · published

SIP trunk carrier turn-up and acceptance testing

Verified 2026-09-25 · 60 sources · tier 1–3

SIP trunk carrier turn-up requires validating signalling paths, media parameters, authentication methods, and regulatory routing profiles against service provider specifications 11​13. Standardizing these tests across inbound, outbound, and emergency traffic profiles confirms trunk operational readiness before placing production traffic into service 8​6.

Signalling and Health Monitoring

SIP OPTIONS queries a peer's capabilities outside an existing dialog, returning supported functions in the 200 OK Allow and Accept headers 39. In SIP digest authentication, a user agent server issues a 401 Unauthorized challenge, whereas a proxy issues a 407 Proxy Authentication Required challenge 38.

When configuring Cisco Unified Border Element (CUBE) on IOS XE 17.6 onwards, failing an out-of-dialog SIP OPTIONS ping across all retries causes CUBE to busy out every dial-peer linked to that options-keepalive profile 19. When monitoring members within a server group, CUBE tests the member destinations sequentially rather than in parallel 21.

For Microsoft Teams Direct Routing pairings, validation requires verifying within the SBC management interface that outbound OPTIONS messages receive a 200 OK response and that incoming OPTIONS from Direct Routing receive a 200 OK reply 31. Direct Routing begins issuing its own OPTIONS queries only after receiving an incoming OPTIONS request from the SBC, targeting the FQDN found in the SBC's Contact header 30. Microsoft enforces TLS 1.2 on this interface, requiring one of 4 ECDHE-RSA AES cipher suites: 256-bit or 128-bit GCM, or 256-bit or 128-bit CBC using SHA-384 or SHA-256 35.

Direct Routing considers an SBC healthy if it has transmitted an OPTIONS ping within the preceding 3 minutes 27. If this threshold expires, the SBC is demoted rather than removed, remaining available as a secondary attempt prior to call failure 27. Disabling the SendSIPOptions parameter removes the SBC from the Monitoring and Alert system, though Microsoft recommends retaining the default enabled setting 34. Setting the MaxConcurrentSessions parameter triggers alert notifications when trunk utilization reaches 90 percent or greater, whereas leaving it unset disables session alerting 28. SBC pairing faults and rejected INVITE events, such as trunk FQDN misconfigurations, are not exposed in Teams Call Analytics; Microsoft directs operators to SBC system logs where Direct Routing returns detailed error strings 33.

Routing, Failover, and Registration

Direct Routing balances traffic across healthy, equal-priority SBCs in a route using a Fisher-Yates shuffle algorithm 25. Outbound calls fail over to subsequent SBCs in a voice routing policy by default on receiving 408, 503, or 504 responses, governed by the FailoverResponseCodes parameter 23. Failover occurs only if no non-100 provisional responses have arrived from the SBC, preventing duplicate ringing on the public telephone network 29. FailoverTimeSeconds sets the outbound attempt timeout to a default of 10 seconds before routing to the next trunk or terminating the call 24. If Direct Routing fails to reach an SBC, it attempts connections from alternative datacenters in other regions; access control lists on the SBC that do not include all Direct Routing datacenter IP addresses can cause connection drops 22.

On Twilio Elastic SIP Trunking, termination requires an IP access control list, a credential list, or both; Twilio enforces both mechanisms if configured simultaneously 56​2. Twilio strongly recommends configuring credentials because IP ACLs alone do not mitigate specific attack vectors 59​3. WarmTransfer's reading of the sources is that source-IP trust and cryptographic credentials represent complementary controls with diverging vendor architectures: Cisco highlights IP trust as its primary toll-fraud control, while Twilio positions IP ACLs as insufficient on their own 1.

For Twilio origination, up to 10 SIP URIs can be configured with priority ordering (lower values preferred) and weights for distribution among equal priorities 60. Twilio triggers failover to the next URI after 4 seconds elapse without a SIP response 60. Twilio sizes peak trunk bandwidth by multiplying maximum simultaneous calls by 100 kbps, accounting for G.711 alongside packet overhead 57.

Under RFC 6140, a SIP-PBX registers an entire block of numbers using a single REGISTER transaction rather than individual registrations per address of record 52. RFC 6140 requires the gin option tag in Require and Proxy-Require headers so unsupported intermediaries reject the transaction instead of misrouting it 51. The bnc (bulk number contact) URI parameter identifies the Contact URI as representing the complete number range, prompting the registrar to generate discrete mappings per assigned number 50.

Caller ID and Privacy Assertions

On Twilio Elastic SIP Trunking termination, the outbound Caller ID number must be a Twilio DID assigned to the account or a verified number within the Twilio Console 58.

RFC 3325 provides an Informational specification published in November 2002 describing P-Asserted-Identity for conveying authenticated sending identities across trusted SIP intermediaries 41​40. These intermediaries operate inside a Trust Domain adhering to a formal Spec(T) definition encompassing security, inter-node trust, and privacy controls 43. A user requests identity masking by adding an id token in the Privacy header, requiring nodes to strip the P-Asserted-Identity header before routing outside the Trust Domain boundary 42. In Direct Routing, ForwardPAI defaults to False; enabling it forwards P-Asserted-Identity along with an accompanying Privacy: id header 26.

Media and DTMF Specifications

RFC 4733, published in December 2006, obsoletes RFC 2833 for carrying DTMF inside an RTP telephone-event payload 48. Event codes allocate digits 0-9 to codes 0-9, * to 10, # to 11, and keys A-D to 12-15 45. Standard digits 0-9, *, and # are widely supported, whereas codes A through D occur less frequently 44. The packet terminating a tone event sets the E (end) bit to 1 46. Senders should repeat this final packet 3 times at the standard update interval 47. Supported events are signaled in SDP using an fmtp parameter on the telephone-event payload line, such as specifying range 0-15 49.

On CUBE running IOS XE 17.6 onwards, supported SIP DTMF relay methods include rtp-nte (RFC 2833 NTE), sip-kpml (RFC 4730), sip-notify (Cisco proprietary), sip-info, and the discontinued cisco-rtp; in-band DTMF carried within G.711 requires no explicit configuration 16. When multiple methods are applied to a single CUBE dial-peer, priority executes in the sequence in which the commands were configured 17.

For facsimile, ITU-T T.38 (11/15) represents the current in-force recommendation for real-time Group 3 facsimile communication over IP, superseding revisions from 1998 through 2010 55. Across carrier specifications, the SIP Forum ratified SIPconnect 2.0 on January 24, 2017, introducing criteria for TLS, SRTP, emergency location exchange, early media, and IPv6 beyond SIPconnect 1.1 54.

Acceptance Test Procedures

Cisco's Bring Your Own PSTN (BYoPSTN) acceptance procedure for Webex Meetings requires partner self-certification followed by Webex Meeting PSTN Certification team validation and archiving 12. Prerequisites establish a minimum of 2 data centers reachable over TCP/IP with routing formatted in E.164 11. Test execution includes separate ingress and egress test runs for G.722, G.711 A-law, G.711 mu-law, and G.729, with the G.729 egress case incorporating transcoding 8. Failover validation verifies survivability against an outage of the primary Webex data center as well as an outage of the partner's primary BYoPSTN data center 9.

In the United States, 47 CFR 9.16(b)(1) mandates that multi-line telephone systems (MLTS) support direct dialing to 911 from any station without dialing any additional digit, code, prefix, or post-fix 13. Under 47 CFR 9.16(b)(2), systems capable of doing so without software or hardware upgrades must transmit an un-delayed simultaneous notification to an on-site or off-site monitoring location when 911 is dialed 15. Section 9.16(b)(3) requires dispatchable location delivery to the PSAP, establishing compliance deadlines of January 6, 2021 for fixed endpoints and January 6, 2022 for non-fixed on-premises and off-premises stations 14.

Bandwidth guidance recommends dialing the target PSAP's 10-digit administrative non-emergency line prior to performing test calls to give advance warning 6. Testers should avoid PSAP peak call periods between 6-9 am, at lunchtime, and between 4-7 pm 4. Upon connection, the tester must state immediately that the call is a non-emergency operational test and verify destination PSAP accuracy 5. Tennessee state guidance directs testers to verify the received address and phone number with the telecommunicator, reporting discrepancies directly to the voice carrier 7.

See also

Applicability

The deployment model is not covered by the sources below. Cisco CUBE DTMF relay methods and server monitoring behaviors apply to Cisco IOS XE 17.6 onwards 16​19. Multi-line telephone system regulatory obligations under 47 CFR 9.16 apply within the United States 13​15​14, while specific PSAP address verification guidelines cite emergency protocols within Tennessee 7.

What remains uncertain

  • STIR/SHAKEN attestation verification during outbound caller ID turn-up tests is not covered by the sources below.
  • Hold and transfer interop on carrier trunks (REFER vs re-INVITE and carrier REFER support) is not covered by the sources below.
  • Carrier 933 or non-PSAP 911 test numbers for emergency test calls are not covered by the sources below.
  • The status of the Microsoft SIP Tester client script for Direct Routing is not covered by the sources below.
  • Twilio Elastic SIP Trunking T.38 and DTMF support is not covered by the sources below.
  • Webex Calling local gateway trunk validation steps are not covered by the sources below.
  • CUBE T.38 fax relay with fallback to G.711 pass-through (command syntax and behaviour) is not covered by the sources below.
  • SIPconnect 2.0 clause-level requirements (registration and static modes; TLS; codecs; fax; DTMF) are not covered by the sources below.

See also

Related to

Referenced by

  • SIP scanning and telephony denial of service — Trunk authentication choices (IP ACL vs digest credentials) bear on scanning and toll-fraud exposure; this packet records Twilio's requirement of an IP ACL or credential list and its recommendation to configure credentials; and Cisco's trusted-list guidance for Local Gateway trunks
  • Troubleshooting Genesys Cloud BYOC trunks — The BYOC Cloud configuration checklist and carrier requirements overlap with carrier turn-up: ACLs on both sides; allowlisting of all published IPs; FQDN and TGRP routing.
  • Wrong or missing caller ID on outbound calls — Carrier screening of From/PAI/Diversion and number ownership is usually agreed and tested during trunk turn-up; proposed by this runner and not yet recorded in the corpus

Sources

  1. 1
    Source-IP trust (the CUBE trusted list, Oracle ACLs, Twilio IP ACLs) and digest or TLS credentials are complementary controls, and vendors differ on whether IP trust alone is enough: Cisco documents IP trust as its primary toll-fraud control, while Twilio warns that IP ACLs alone are insufficient.inferred
    Elastic SIP Trunking · Termination Settings > Authentication · Checked 2026-09-25
  2. 2
    Twilio Elastic SIP Trunking requires at least an IP access control list or a credential list for termination, and enforces both when both are set.
    Elastic SIP Trunking · Termination Settings > Authentication · Checked 2026-09-25
  3. 3
    Twilio strongly recommends configuring user credentials on SIP trunks because IP ACLs alone do not protect against certain types of attack.
    Elastic SIP Trunking · Termination Settings > Authentication · Checked 2026-09-25
  4. 4
    Bandwidth advises against placing 911 test calls during the PSAP busy hours of 6-9 am, 4-7 pm and lunchtime.
    Tips for a successful 911 test call · Step 1 · Checked 2026-09-25
  5. 5
    On a 911 test call the caller should say at once that it is not an emergency but a test call, and confirm with the call taker that the right PSAP was reached.
    Tips for a successful 911 test call · Step 3 · Checked 2026-09-25
  6. 6
    Before placing a 911 test call, Bandwidth advises calling the PSAP's 10-digit non-emergency number to tell them a test call is coming.
    Tips for a successful 911 test call · Step 2 · Checked 2026-09-25
  7. 7
    Tennessee's 911 guidance says a test caller should confirm their address and phone number with the telecommunicator, and contact their phone service provider about any location discrepancy.
    How to Test Your Telephone With 911 · Step 4 (verify information) · Checked 2026-09-25
  8. 8
    The Webex BYoPSTN acceptance procedure has separate ingress and egress test cases for each of G.722, G.711 A-law, G.711 mu-law and G.729, and the G.729 egress case involves transcoding.
    Bring your own PSTN acceptance procedure · Ingress test cases; Egress test cases · Checked 2026-09-25
  9. 9
    The Webex BYoPSTN acceptance procedure includes failover test cases for an outage of the primary Webex data center and for an outage of the primary BYoPSTN data center.
    Bring your own PSTN acceptance procedure · Failover test cases · Checked 2026-09-25
  10. 10
    Each test in the Webex BYoPSTN acceptance procedure checks DTMF entry, localized prompts, two-way audio and correct call termination; supplementary cases cover on-hold handling and extended mute periods.
    Bring your own PSTN acceptance procedure · Test case sections; Supplementary functions · Checked 2026-09-25
  11. 11
    Prerequisites in the Webex BYoPSTN acceptance procedure include at least dual data centers reachable over TCP/IP, and routing in E.164 format.
    Bring your own PSTN acceptance procedure · Prerequisites section · Checked 2026-09-25
  12. 12
    Cisco's 'Bring your own PSTN acceptance procedure' covers Webex Meetings BYoPSTN: the partner self-certifies, and the Webex Meeting PSTN Certification team validates and archives the results.
    Bring your own PSTN acceptance procedure · Introduction · Checked 2026-09-25
  13. 13
    47 CFR 9.16(b)(1) requires an MLTS to be configured so users can call 911 directly from any station with dialing facilities, without dialing any additional digit, code, prefix or post-fix.
    47 CFR § 9.16 - General obligations for MLTS (Cornell LII republication) · 47 CFR 9.16(b)(1) · Checked 2026-09-25
  14. 14
    47 CFR 9.16(b)(3) requires an MLTS to convey dispatchable location to the PSAP, with compliance dates of January 6, 2021 for fixed devices and January 6, 2022 for non-fixed on-premises and off-premises devices.
    47 CFR § 9.16 - General obligations for MLTS (Cornell LII republication) · 47 CFR 9.16(b)(3) · Checked 2026-09-25
  15. 15
    47 CFR 9.16(b)(2) requires an MLTS that can do so without an improvement to its hardware or software to notify a central location or another person when a 911 call is made, at the same time as the call and without delaying it.
    47 CFR § 9.16 - General obligations for MLTS (Cornell LII republication) · 47 CFR 9.16(b)(2) · Checked 2026-09-25
  16. 16
    On CUBE (IOS XE 17.6 onwards) the documented DTMF relay methods for SIP are rtp-nte (RFC 2833 NTE), sip-kpml (RFC 4730), sip-notify (Cisco proprietary), sip-info and cisco-rtp (discontinued); in-band DTMF in G.711 audio needs no configuration.
    Cisco Unified Border Element Configuration Guide - Cisco IOS XE 17.6 Onwards - DTMF Relay · DTMF Relay chapter, DTMF relay methods overview · Checked 2026-09-25
  17. 17
    When more than one DTMF relay method is configured on a CUBE dial-peer, preference runs from highest to lowest in the order the methods were configured.
    Cisco Unified Border Element Configuration Guide - Cisco IOS XE 17.6 Onwards - DTMF Relay · DTMF Relay chapter, multiple DTMF relay methods · Checked 2026-09-25
  18. 18
    In the CUBE SIP registration feature's end-to-end mode, CUBE ignores its local registrar and authentication configuration and passes the Authorization headers through to the registrar.
    Cisco Unified Border Element Configuration Guide - Cisco IOS XE 17.6 Onwards - Configure SIP Registration · Configure SIP Registration chapter, end-to-end mode description · Checked 2026-09-25
  19. 19
    When SIP out-of-dialog OPTIONS pings to a monitored destination fail after all retries, CUBE busies out every dial-peer associated with that options-keepalive profile.
    Cisco Unified Border Element Configuration Guide - Cisco IOS XE 17.6 Onwards - SIP Trunk Monitoring · SIP Trunk Monitoring chapter, feature description · Checked 2026-09-25
  20. 20
    The CUBE options-keepalive profile defaults are a 60-second up-interval, a 30-second down-interval and 5 retries.
    Cisco Unified Border Element Configuration Guide - Cisco IOS XE 17.6 Onwards - SIP Trunk Monitoring · SIP Trunk Monitoring chapter, options-keepalive profile configuration · Checked 2026-09-25
  21. 21
    CUBE tests the members of a server group one at a time, not in parallel, when monitoring them with OPTIONS.
    Cisco Unified Border Element Configuration Guide - Cisco IOS XE 17.6 Onwards - SIP Trunk Monitoring · SIP Trunk Monitoring chapter, server group monitoring · Checked 2026-09-25
  22. 22
    If Direct Routing cannot connect to a trunk, it retries from a different Microsoft datacenter, possibly in another region, so an SBC ACL that omits some Direct Routing datacenter IP addresses can cause connection failures.
    Trunk failover on outbound calls - Microsoft Teams · Failover on network errors · Checked 2026-09-25
  23. 23
    By default Direct Routing retries an outbound call on another SBC in the voice routing policy when the first SBC returns 408, 503 or 504; the list is set with the FailoverResponseCodes parameter.
    Trunk failover on outbound calls - Microsoft Teams · Failover of specific SIP codes received from the SBC (Default values line) · Checked 2026-09-25
  24. 24
    The Direct Routing FailoverTimeSeconds setting defaults to 10 seconds: an outbound call the gateway has not answered within that time moves to the next available trunk, or is dropped if there is none.
    Connect your Session Border Controller (SBC) to Direct Routing · SBC settings table, FailoverTimeSeconds row · Checked 2026-09-25
  25. 25
    When two or more SBCs in one Direct Routing route are healthy and of equal priority, calls are spread across them with a Fisher-Yates shuffle.
    Monitor Direct Routing · Monitoring availability of SBCs using SIP options messages, last paragraph · Checked 2026-09-25
  26. 26
    The Direct Routing ForwardPAI setting defaults to False; when it is on, the P-Asserted-Identity header is forwarded with the call and a Privacy:ID header is also sent.
    Connect your Session Border Controller (SBC) to Direct Routing · SBC settings table, ForwardPAI row · Checked 2026-09-25
  27. 27
    When routing a call, Direct Routing treats an SBC as healthy if it has sent OPTIONS within the last three minutes; otherwise the SBC is demoted, not removed, and is tried again before the call fails.
    Monitor Direct Routing · Monitoring availability of SBCs using SIP options messages · Checked 2026-09-25
  28. 28
    When MaxConcurrentSessions is set on a Direct Routing SBC, the alerting system notifies at 90 percent or more of the value; if it is unset, no alerts are generated.
    Connect your Session Border Controller (SBC) to Direct Routing · SBC settings table, MaxConcurrentSessions row · Checked 2026-09-25
  29. 29
    Direct Routing fails over on a configured response code only if the SBC has sent no non-100 provisional response, which avoids double ringing on the telephone network.
    Trunk failover on outbound calls - Microsoft Teams · Failover of specific SIP codes received from the SBC, first paragraph · Checked 2026-09-25
  30. 30
    Direct Routing starts sending its own OPTIONS only after it receives OPTIONS from the SBC, and sends them to the SBC FQDN in the Contact header of those incoming OPTIONS.
    Connect your Session Border Controller (SBC) to Direct Routing · Verify the SBC connection > Validate SIP options · Checked 2026-09-25
  31. 31
    Microsoft's step for validating a new Direct Routing SBC pairing is to confirm in the SBC management interface that its outgoing SIP OPTIONS get 200 OK and that it answers Direct Routing's incoming OPTIONS with 200 OK.
    Connect your Session Border Controller (SBC) to Direct Routing · Use PowerShell > Verify the SBC connection > Validate SIP options · Checked 2026-09-25
  32. 32
    The Direct Routing connect article says SBC OPTIONS pings must not exceed one transaction every 60 seconds and must not be more or less frequent than one every 180 seconds, per trunk per endpoint; read with the monitor article, the working interval is between 60 and 180 seconds.inferred
    Connect your Session Border Controller (SBC) to Direct Routing · Connect the SBC to the tenant > Considerations (SIP OPTIONS bullet) · Checked 2026-09-25
  33. 33
    Teams Call Analytics does not help with SBC pairing problems or rejected INVITEs such as a misconfigured trunk FQDN; for those Microsoft points to the SBC logs, where Direct Routing sends detailed error descriptions.
    Monitor Direct Routing · Monitor Call Quality Analytics dashboard and SBC logs, final paragraph · Checked 2026-09-25
  34. 34
    Turning off SendSIPOptions on a Direct Routing SBC excludes it from the Monitoring and Alert system; Microsoft highly recommends leaving it on, which is the default.
    Connect your Session Border Controller (SBC) to Direct Routing · SBC settings table, SendSIPOptions row · Checked 2026-09-25
  35. 35
    Microsoft forces TLS 1.2 on the Direct Routing SIP interface, and the SBC must connect with one of four listed ECDHE-RSA AES cipher suites (256/128-bit GCM, or 256/128-bit CBC with SHA-384/SHA-256).
    Connect your Session Border Controller (SBC) to Direct Routing · Connect the SBC to the tenant > Considerations · Checked 2026-09-25
  36. 36
    Because RFC 4733 DTMF support is negotiated per call in SDP, a DTMF acceptance test is only meaningful if it covers both inbound and outbound calls and every codec the trunk may negotiate.inferred
    RFC 4733: RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals · Sections 2.4.1 and 2.5.1.1 (negotiation); reasoning by the researcher · Checked 2026-09-25
  37. 37
    A turn-up failover test is only valid if it reproduces the trigger the platform actually acts on: missing OPTIONS for Direct Routing SBC health or CUBE dial-peer busyout, a configured final response code before any 18x for Direct Routing SIP-code failover, and no SIP response within 4 seconds for Twilio origination.inferred
    Trunk failover on outbound calls - Microsoft Teams · Synthesis across the failover sections of the cited sources · Checked 2026-09-25
  38. 38
    In SIP digest authentication a user agent server challenges with 401 Unauthorized, while a proxy challenges with 407 Proxy Authentication Required.
    RFC 3261 — SIP: Session Initiation Protocol · Section 22 (Usage of HTTP Authentication) · Checked 2026-09-25
  39. 39
    SIP OPTIONS (RFC 3261 section 11) queries a peer's capabilities and may be sent outside a dialog; the capabilities come back in the 200 OK's Allow and Accept headers.
    RFC 3261 — SIP: Session Initiation Protocol · Section 11 (Querying for Capabilities) · Checked 2026-09-25
  40. 40
    RFC 3325 is an Informational RFC published in November 2002, not a Standards Track specification.
    RFC 3325: Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks · RFC header (Category: Informational) · Checked 2026-09-25
  41. 41
    RFC 3325 defines P-Asserted-Identity for use among trusted SIP entities, typically intermediaries, to carry the identity of the sending user as verified by authentication.
    RFC 3325: Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks · Section 9.1 · Checked 2026-09-25
  42. 42
    Under RFC 3325 a user requests identity privacy by including the 'id' token in the Privacy header, asking that P-Asserted-Identity be removed before the request leaves the Trust Domain.
    RFC 3325: Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks · Sections 7 and 9.3 · Checked 2026-09-25
  43. 43
    RFC 3325 relies on a Trust Domain whose members follow a documented specification, Spec(T), covering authentication, inter-node security, membership and default privacy handling.
    RFC 3325: Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks · Sections 1 and 11 (Trust Domain, Spec(T)) · Checked 2026-09-25
  44. 44
    RFC 4733 says DTMF digits 0-9, * and # are commonly supported and that digits A through D are less frequently encountered.
    RFC 4733: RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals · Section 3.2 · Checked 2026-09-25
  45. 45
    In RFC 4733 the DTMF digits 0-9 are event codes 0-9, * is event code 10, # is event code 11 and A-D are event codes 12-15.
    RFC 4733: RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals · Section 3.2, Table 3 · Checked 2026-09-25
  46. 46
    In the RFC 4733 payload, the E (end) bit set to one marks the packet that contains the end of the event.
    RFC 4733: RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals · Section 2.3.2 · Checked 2026-09-25
  47. 47
    RFC 4733 says the sender SHOULD send the final packet of each event three times, at the interval it uses for updates.
    RFC 4733: RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals · Section 2.5.1.4 · Checked 2026-09-25
  48. 48
    RFC 4733, published December 2006, obsoletes RFC 2833 as the specification for the RTP telephone-event payload used to carry DTMF.
    RFC 4733: RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals · RFC header (Obsoletes: 2833) · Checked 2026-09-25
  49. 49
    Under RFC 4733, the events an endpoint supports are declared in SDP with an fmtp line on the telephone-event payload type, for example a range 0-15 covering the DTMF events.
    RFC 4733: RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals · Sections 2.4.1 and 2.5.1.1 (SDP examples) · Checked 2026-09-25
  50. 50
    The RFC 6140 'bnc' (bulk number contact) URI parameter marks a Contact URI as standing for all the phone numbers assigned to the SIP-PBX; the registrar expands it into one mapping per number.
    RFC 6140: Registration for Multiple Phone Numbers in the Session Initiation Protocol (SIP) · Sections 5.1 and 9.2.1 · Checked 2026-09-25
  51. 51
    An RFC 6140 registration carries the 'gin' option tag in both the Require and Proxy-Require header fields, so a proxy or registrar that lacks the capability rejects the request instead of misinterpreting it.
    RFC 6140: Registration for Multiple Phone Numbers in the Session Initiation Protocol (SIP) · Sections 5.1 and 9.1 · Checked 2026-09-25
  52. 52
    RFC 6140 lets a SIP-PBX register a whole block of phone numbers with a service provider in a single REGISTER transaction, instead of one registration per address of record.
    RFC 6140: Registration for Multiple Phone Numbers in the Session Initiation Protocol (SIP) · Section 1 (Introduction) · Checked 2026-09-25
  53. 53
    CUBE's rtp-nte method, which Cisco labels RFC 2833, is the same named-telephone-event mechanism now specified by RFC 4733, so a carrier that asks for RFC 4733 DTMF is normally met by rtp-nte.inferred
    Cisco Unified Border Element Configuration Guide - Cisco IOS XE 17.6 Onwards - DTMF Relay · DTMF Relay chapter, rtp-nte description; combined with RFC 4733 header · Checked 2026-09-25
  54. 54
    The SIP Forum announced ratification of SIPconnect 2.0 on January 24, 2017; beyond SIPconnect 1.1 it covers security (TLS and SRTP), emergency-calling location information, early media and IPv6.
    SIP Forum Announces Ratification of Version 2.0 of the SIPconnect Technical Recommendation · Press release body, coverage paragraph · Checked 2026-09-25
  55. 55
    The ITU-T T.38 recommendation, 'Procedures for real-time Group 3 facsimile communication over IP networks', has T.38 (11/15) as its in-force version; the 1998-2010 editions are superseded.
    T.38 : Procedures for real-time Group 3 facsimile communication over IP networks · Recommendation page, versions table · Checked 2026-09-25
  56. 56
    Twilio Elastic SIP Trunking termination requires at least an IP access control list or a credential list, and enforces both when both are configured.
    Elastic SIP Trunking · Termination section, authentication · Checked 2026-09-25
  57. 57
    Twilio sizes peak SIP trunk bandwidth as maximum simultaneous calls times 100 kbps, a figure covering G.711 plus overhead.
    Elastic SIP Trunking · Bandwidth requirements section · Checked 2026-09-25
  58. 58
    On a Twilio Elastic SIP Trunking termination call, the Caller ID number must be a Twilio DID on the account or a number verified in the Twilio Console.
    Elastic SIP Trunking · Termination section, caller ID · Checked 2026-09-25
  59. 60
    Twilio Elastic SIP Trunking accepts up to ten origination SIP URIs, ordered by priority (lower number is preferred) with weight splitting load among equal priorities, and fails over to the next URI after 4 seconds without a SIP response.
    Elastic SIP Trunking · Origination section, origination URIs, priority and weight · Checked 2026-09-25

Documents

tier 1 standards and regulators

47 CFR § 9.16 - General obligations for MLTS (Cornell LII republication)

Legal Information Institute, Cornell Law School (republishing eCFR) · 2020-12-03 · accessed 2026-09-24

tier 1 standards and regulators

How to Test Your Telephone With 911

Tennessee Department of Commerce & Insurance (Tennessee Emergency Communications Board) · accessed 2026-09-25

tier 1 standards and regulators

RFC 3261 — SIP: Session Initiation Protocol

IETF / RFC Editor · 2002-06 · accessed 2026-09-04

tier 1 standards and regulators

RFC 3325: Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks

IETF / RFC Editor · 2002-11-01 · accessed 2026-09-25

tier 1 standards and regulators

RFC 4733: RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals

IETF / RFC Editor · 2006-12-01 · accessed 2026-09-24

tier 1 standards and regulators

RFC 6140: Registration for Multiple Phone Numbers in the Session Initiation Protocol (SIP)

IETF / RFC Editor · 2011-03-01 · accessed 2026-09-25

tier 1 standards and regulators

T.38 : Procedures for real-time Group 3 facsimile communication over IP networks

ITU-T · 2015-11-29 · accessed 2026-09-24

tier 2 current vendor documentation

Bring your own PSTN acceptance procedure

Cisco Webex · 2025-02-11 · accessed 2026-09-25

tier 2 current vendor documentation

Cisco Unified Border Element Configuration Guide - Cisco IOS XE 17.6 Onwards - Configure SIP Registration

Cisco Systems · 2026-04-25 · accessed 2026-09-25

tier 2 current vendor documentation

Cisco Unified Border Element Configuration Guide - Cisco IOS XE 17.6 Onwards - DTMF Relay

Cisco Systems · 2026-04-25 · accessed 2026-09-24

tier 2 current vendor documentation

Cisco Unified Border Element Configuration Guide - Cisco IOS XE 17.6 Onwards - SIP Trunk Monitoring

Cisco · 2026-04-25 · accessed 2026-09-24

tier 2 current vendor documentation

Connect your Session Border Controller (SBC) to Direct Routing

Microsoft · 2026-04-26 · accessed 2026-09-21

tier 2 current vendor documentation

Elastic SIP Trunking

Twilio · 2026-07-21 · accessed 2026-09-24

tier 2 current vendor documentation

Monitor Direct Routing

Microsoft · 2026-09-10 · accessed 2026-09-21

tier 2 current vendor documentation

Tips for a successful 911 test call

Bandwidth · 2025-11-17 · accessed 2026-09-25

tier 2 current vendor documentation

Trunk failover on outbound calls - Microsoft Teams

Microsoft · 2025-05-06 · accessed 2026-09-25

tier 3 vendor-maintained repositories

SIP Forum Announces Ratification of Version 2.0 of the SIPconnect Technical Recommendation

SIP Forum (via PRWeb) · 2017-01-24 · accessed 2026-09-25

Cite this page

APA

WarmTransfer. (2026, September 25). SIP trunk carrier turn-up and acceptance testing. WarmTransfer. https://warmtransfer.net/knowledge/sip-trunk-carrier-turn-up

BibTeX

@misc{warmtransfer-sip-trunk-carrier-turn-up,
  title  = {SIP trunk carrier turn-up and acceptance testing},
  author = {{WarmTransfer}},
  year   = {2026},
  url    = {https://warmtransfer.net/knowledge/sip-trunk-carrier-turn-up},
  note   = {Verified 2026-09-25}
}