# SS7 and carrier interworking concepts Canonical: https://warmtransfer.net/knowledge/ss7-carrier-interworking Last verified: 2026-09-24 Gateways translate signalling between traditional carrier SS7 networks and SIP-based networks using defined mapping standards[^48]. These interworking mechanisms handle call setup, call teardown, cause code translation, and number portability parameters across network boundaries[^48][^38]. ## Call setup and teardown mapping RFC 3398 (December 2002) is a Standards Track IETF document describing how a gateway maps ISUP signalling to SIP and SIP to ISUP[^48]. Under RFC 3398, when an ISUP ACM carries a Backward Call Indicators value of subscriber free, the gateway SHOULD send 180 Ringing toward SIP[^2]. Under RFC 3398, on receipt of an ISUP ANM or CON the gateway SHOULD send 200 OK toward SIP, which marks answer[^3]. ITU-T Q.1912.5, on interworking between SIP and BICC or ISUP, is in force in its January 2018 edition with Corrigendum 1 (07/18), and the March 2004 edition is superseded[^43]. ## Cause code and Reason header mapping RFC 3398 describes how a gateway maps ISUP signalling to SIP and SIP to ISUP:[^48] | ISUP cause value | SIP status code | Citation | | --- | --- | --- | | 1 (unallocated number) | 404 Not Found |[^6] | | 17 (user busy) | 486 Busy Here |[^8] | | 18 (no user responding) | 408 Request Timeout |[^9] | | 19 (no answer from user) | 480 Temporarily Unavailable |[^10] | | 21 (call rejected) | 403 Forbidden |[^11] | | 28 (address incomplete) | 484 Address Incomplete |[^12] | | 34 (no circuit available), 38 (network out of order) and 41 (temporary failure) | 503 Service Unavailable |[^13] | | 102 (recovery on timer expiry) | 504 Gateway Timeout |[^7] | Under RFC 3398, an ISUP cause value not listed in the mapping table SHOULD produce the default SIP response 500 Server Internal Error[^33]. RFC 3398 maps a SIP 404 Not Found final response to ISUP cause value 1 (unallocated number) toward the PSTN[^55]. RFC 3398 maps both SIP 486 Busy Here and SIP 600 Busy Everywhere to ISUP cause value 17 (user busy)[^58]. RFC 3398 maps SIP 603 Decline to ISUP cause value 21 (call rejected)[^60]. RFC 3398 maps SIP 484 Address Incomplete to ISUP cause value 28 (address incomplete)[^57]. Under RFC 3398, a SIP status code not listed in the SIP-to-ISUP table SHOULD be mapped to the default ISUP cause 16 (normal clearing)[^50]. Cisco IOS SIP gateways let administrators override default cause mapping under sip-ua with set pstn-cause sip-status and set sip-status pstn-cause [^19]. RFC 3326 defines the SIP Reason header with registered protocol values SIP and Q.850, where Q.850 carries an ITU-T Q.850 cause value in the cause parameter[^47]. RFC 3326 allows the Reason header in any request within a dialog and in any CANCEL request, which lets a gateway pass the ISUP release cause onward in BYE or CANCEL[^44]. RFC 6432 permits any SIP response except 100 Trying to carry a Reason header with a Q.850 cause code; it permits but does not mandate this[^45]. RFC 6432 recommends integrity protection (for example TLS or IPsec) for Reason headers carrying Q.850 codes because a wrong code could trigger the wrong announcement[^46]. ## Overlap dialling and encapsulation RFC 3398 places overlap dialling (ISUP Subsequent Address Messages) outside its scope and assumes gateways collect all digits with timers before sending an INVITE[^49]. RFC 3578 states that the preferred approach for a gateway handling PSTN overlap signalling is to convert it into SIP en-bloc signalling[^40]. When overlap is carried into SIP under RFC 3578, each arriving SAM refreshes timer T10 and triggers a new INVITE with the accumulated digits, and 4xx-6xx responses do not trigger an ISUP REL until T10 expires with no more SAMs[^39]. RFC 3372 (September 2002, BCP 63) defines SIP-T as a set of mechanisms for interfacing traditional telephone signalling with SIP, not a new protocol[^51]. RFC 3204 registers the MIME media types application/ISUP and application/QSIG for carrying ISUP and QSIG objects in SIP message bodies[^4]. The application/ISUP media type requires a version parameter identifying the ISUP variant and allows an optional base parameter naming a fallback specification[^34]. RFC 4666 (M3UA, September 2006, obsoleting RFC 3332) transports SS7 MTP3-User signalling such as ISUP and SCCP over IP using SCTP, between a Signalling Gateway and IP-based application server processes[^37]. ## Number portability parameters When converting ISUP to SIP for a ported geographical number, RFC 4694 requires the routing number found in the ISUP Called Party Number to be carried in the rn parameter of the routing tel URI[^14]. RFC 4694 requires the number marked as ported number in the ISUP Generic Address Parameter to become the telephone number of the routing tel URI[^31]. RFC 4694 requires npdi on the routing tel URI when the Ported Number Translation Indicator bit in the ISUP Forward Call Indicators is set to 1[^41]. Under RFC 4694, a network node MUST NOT retrieve number portability information for a geographical number if the tel URI already carries the npdi parameter[^38]. RFC 4694 requires the cic parameter on the routing tel URI when the ISUP Carrier Identification Parameter is present[^18]. RFC 4694 says a node MAY put rn on the caller-information tel URI when the ISUP Jurisdiction Information Parameter is present; JIP does not populate the routing URI[^35]. ## Non-IP regulatory transitions and interconnection 47 CFR 64.6303(a) required voice service providers by June 30 2021 either to upgrade their whole network to SIP and fully implement STIR/SHAKEN or to keep documented participation in developing a non-IP caller ID authentication solution[^17]. 47 CFR 64.6303(b) set June 30 2023 for gateway providers to either upgrade to SIP with full STIR/SHAKEN or keep proof of participation in non-IP authentication standards work[^15]. 47 CFR 64.6303(c) set December 31 2023 for non-gateway intermediate providers to either upgrade to SIP with STIR/SHAKEN or provide the Commission on request documented proof of work on non-IP authentication[^16]. ATIS lists ATIS-1000095 as a method for carrying STIR/SHAKEN attestation across TDM networks using SS7 parameters, among the deliverables of its Non-IP Call Authentication Task Force[^5]. FCC 25-25 (adopted 2025-04-28, released 2025-04-29, WC Docket 17-97) is a notice of proposed rulemaking that proposes requiring providers still relying on non-IP networks to implement a non-IP caller ID authentication framework[^23]. FCC 25-25 proposes a compliance deadline 2 years after the rule's effective date and proposes to repeal the continuing extension that non-IP providers have held[^26]. FCC 25-25 proposes that providers certify in their Robocall Mitigation Database filings whether they have implemented a non-IP caller ID authentication framework[^24]. FCC 25-73 names SS7 signalling points with local switches, tandems and 911 selective routers as elements of the typical TDM network topology in use[^29]. FCC 25-73 (Advancing IP Interconnection NPRM, released 2025-10-29) proposes forbearing from sections 251(c)(2) and 251(c)(6) and removing 47 CFR 51.305, 51.321 and 51.323 effective December 31 2028, ending incumbent LEC obligations that keep TDM interconnection in place[^27]. FCC 25-73 says sunsetting incumbent LEC interconnection obligations would not prohibit TDM interconnection; incumbents could continue to interconnect in TDM[^30]. ## Applicability Applies to: IETF SIP-ISUP gateways, IETF tel URI, ITU-T SIP-ISUP, Cisco IOS voice gateway, IETF SIP, IETF SIP-T, IETF SIGTRAN M3UA, FCC caller ID authentication rules, ATIS non-IP call authentication standards, and FCC interconnection rules. Deployments: on-premises, multi-tenant, and carrier. Sources checked 2026-09-24. The 47 CFR 64.6303 requirements, FCC 25-25 proposed rules, ATIS-1000095 deliverables, and FCC 25-73 proposals apply to carrier networks in the United States[^17][^15][^16][^23][^5][^27]. Configurable PSTN cause mapping under sip-ua applies to Cisco IOS Release 15M&T gateways[^19]. ## What remains uncertain A final FCC order on non-IP caller ID authentication following FCC 25-25 and any adopted deadline are not covered by the sources below. Default Q.850 to SIP mappings used by major carriers are not covered by the sources below. The outcome of FCC 25-73 Advancing IP Interconnection after reply comments closed is not covered by the sources below. Calling number presentation and privacy mapping between ISUP and SIP (RFC 3323 and RFC 3325 and P-Asserted-Identity) are not covered by the sources below. Current Cisco IOS XE CUBE cause code mapping defaults (m_cause-code-mapping PDF) are not covered by the sources below. FCC number portability rules and NPAC LRN operation behind rn and npdi are not covered by the sources below. 3GPP IMS to CS network interworking (current release) is not covered by the sources below. Raw-text verification of RFC 3398 table rows 480 and 503 and of section 7.2.5 heading is not covered by the sources below. ## Sources [^1]: The cause a PSTN caller receives for a SIP 503 depends on the gateway: RFC 3398 recommends cause 38 but the Cisco IOS 15M&T default is cause 63, so gateway configuration should be checked before interpreting a release cause (inferred). Source: [SIP Configuration Guide, Cisco IOS Release 15M&T - Configuring SIP Message Timer and Response Features](https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/sip/configuration/15-mt/sip-config-15-mt-book/voi-sip-timer.html), Section SIP - Configurable PSTN Cause Code Mapping compared with RFC 3398 Section 8.2.6.1. Checked 2026-09-24. [^2]: Under RFC 3398, when an ISUP ACM carries a Backward Call Indicators value of subscriber free, the gateway SHOULD send 180 Ringing toward SIP. Source: [RFC 3398: Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping](https://www.rfc-editor.org/rfc/rfc3398.html), Section 7.2.6 ACM received. Checked 2026-09-24. [^3]: Under RFC 3398, on receipt of an ISUP ANM or CON the gateway SHOULD send 200 OK toward SIP, which marks answer. Source: [RFC 3398: Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping](https://www.rfc-editor.org/rfc/rfc3398.html), Section 7.2.7 CON or ANM Received. Checked 2026-09-24. [^4]: RFC 3204 registers the MIME media types application/ISUP and application/QSIG for carrying ISUP and QSIG objects in SIP message bodies. Source: [RFC 3204: MIME media types for ISUP and QSIG Objects](https://www.rfc-editor.org/rfc/rfc3204.html), Abstract; Sections 3.1 and 3.2. Checked 2026-09-24. [^5]: ATIS lists ATIS-1000095 as a method for carrying STIR/SHAKEN attestation across TDM networks using SS7 parameters, among the deliverables of its Non-IP Call Authentication Task Force. Source: [Non-IP Call Authentication Task Force (NIPCA-TF)](https://atis.org/committees-forums/ptsc/non-ip-call-authentication-task-force/), NIPCA-TF page, deliverables list. Checked 2026-09-24. [^6]: RFC 3398 maps ISUP cause value 1 (unallocated number) received from the PSTN to the SIP response 404 Not Found. Source: [RFC 3398: Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping](https://www.rfc-editor.org/rfc/rfc3398.html), Section 7.2.4.1 ISDN Cause Code to Status Code Mapping, table. Checked 2026-09-24. [^7]: RFC 3398 maps ISUP cause value 102 (recovery on timer expiry) to the SIP response 504 Gateway Timeout. Source: [RFC 3398: Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping](https://www.rfc-editor.org/rfc/rfc3398.html), Section 7.2.4.1 ISDN Cause Code to Status Code Mapping, table. Checked 2026-09-24. [^8]: RFC 3398 maps ISUP cause value 17 (user busy) to the SIP response 486 Busy Here. Source: [RFC 3398: Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping](https://www.rfc-editor.org/rfc/rfc3398.html), Section 7.2.4.1 ISDN Cause Code to Status Code Mapping, table. Checked 2026-09-24. [^9]: RFC 3398 maps ISUP cause value 18 (no user responding) to the SIP response 408 Request Timeout. Source: [RFC 3398: Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping](https://www.rfc-editor.org/rfc/rfc3398.html), Section 7.2.4.1 ISDN Cause Code to Status Code Mapping, table. Checked 2026-09-24. [^10]: RFC 3398 maps ISUP cause value 19 (no answer from user) to the SIP response 480 Temporarily Unavailable. Source: [RFC 3398: Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping](https://www.rfc-editor.org/rfc/rfc3398.html), Section 7.2.4.1 ISDN Cause Code to Status Code Mapping, table. Checked 2026-09-24. [^11]: RFC 3398 maps ISUP cause value 21 (call rejected) to the SIP response 403 Forbidden. Source: [RFC 3398: Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping](https://www.rfc-editor.org/rfc/rfc3398.html), Section 7.2.4.1 ISDN Cause Code to Status Code Mapping, table. Checked 2026-09-24. [^12]: RFC 3398 maps ISUP cause value 28 (address incomplete) to the SIP response 484 Address Incomplete. Source: [RFC 3398: Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping](https://www.rfc-editor.org/rfc/rfc3398.html), Section 7.2.4.1 ISDN Cause Code to Status Code Mapping, table. Checked 2026-09-24. [^13]: RFC 3398 maps ISUP cause values 34 (no circuit available), 38 (network out of order) and 41 (temporary failure) to the SIP response 503 Service Unavailable. Source: [RFC 3398: Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping](https://www.rfc-editor.org/rfc/rfc3398.html), Section 7.2.4.1 ISDN Cause Code to Status Code Mapping, table. Checked 2026-09-24. [^14]: When converting ISUP to SIP for a ported geographical number, RFC 4694 requires the routing number found in the ISUP Called Party Number to be carried in the rn parameter of the routing tel URI. Source: [RFC 4694: Number Portability Parameters for the "tel" URI](https://www.rfc-editor.org/rfc/rfc4694.html), Section 5.2.4. Checked 2026-09-24. [^15]: 47 CFR 64.6303(b) set June 30 2023 for gateway providers to either upgrade to SIP with full STIR/SHAKEN or keep proof of participation in non-IP authentication standards work. Source: [47 CFR § 64.6303 - Caller ID authentication in non-IP networks](https://www.law.cornell.edu/cfr/text/47/64.6303), 47 CFR 64.6303(b). Checked 2026-09-24. [^16]: 47 CFR 64.6303(c) set December 31 2023 for non-gateway intermediate providers to either upgrade to SIP with STIR/SHAKEN or provide the Commission on request documented proof of work on non-IP authentication. Source: [47 CFR § 64.6303 - Caller ID authentication in non-IP networks](https://www.law.cornell.edu/cfr/text/47/64.6303), 47 CFR 64.6303(c). Checked 2026-09-24. [^17]: 47 CFR 64.6303(a) required voice service providers by June 30 2021 either to upgrade their whole network to SIP and fully implement STIR/SHAKEN or to keep documented participation in developing a non-IP caller ID authentication solution. Source: [47 CFR § 64.6303 - Caller ID authentication in non-IP networks](https://www.law.cornell.edu/cfr/text/47/64.6303), 47 CFR 64.6303(a). Checked 2026-09-24. [^18]: RFC 4694 requires the cic parameter on the routing tel URI when the ISUP Carrier Identification Parameter is present. Source: [RFC 4694: Number Portability Parameters for the "tel" URI](https://www.rfc-editor.org/rfc/rfc4694.html), Section 5.2.4. Checked 2026-09-24. [^19]: Cisco IOS SIP gateways let administrators override default cause mapping under sip-ua with set pstn-cause sip-status and set sip-status pstn-cause . Source: [SIP Configuration Guide, Cisco IOS Release 15M&T - Configuring SIP Message Timer and Response Features](https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/sip/configuration/15-mt/sip-config-15-mt-book/voi-sip-timer.html), Section SIP - Configurable PSTN Cause Code Mapping. Checked 2026-09-24. [^20]: The Cisco IOS 15M&T SIP guide describes adding the Reason header with the Q.850 cause to outgoing BYE and CANCEL requests and 4xx, 5xx and 6xx responses, so the original ISDN disconnect cause survives the SIP leg. Source: [SIP Configuration Guide, Cisco IOS Release 15M&T - Configuring SIP Message Timer and Response Features](https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/sip/configuration/15-mt/sip-config-15-mt-book/voi-sip-timer.html), Reason header section of the Configuring SIP Message Timer and Response Features chapter. Checked 2026-09-24. [^21]: In the Cisco IOS 15M&T default SIP-to-PSTN mapping, SIP 503 Service Unavailable maps to cause 63 (service or option not available). Source: [SIP Configuration Guide, Cisco IOS Release 15M&T - Configuring SIP Message Timer and Response Features](https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/sip/configuration/15-mt/sip-config-15-mt-book/voi-sip-timer.html), Section SIP - Configurable PSTN Cause Code Mapping, default SIP-to-PSTN table. Checked 2026-09-24. [^22]: Under RFC 3398, when an early ACM (called party status no indication) is received, the gateway SHOULD send 183 Session Progress toward SIP. Source: [RFC 3398: Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping](https://www.rfc-editor.org/rfc/rfc3398.html), Section 7.2.5 (early ACM). Checked 2026-09-24. [^23]: FCC 25-25 (adopted 2025-04-28, released 2025-04-29, WC Docket 17-97) is a notice of proposed rulemaking that proposes requiring providers still relying on non-IP networks to implement a non-IP caller ID authentication framework. Source: [Call Authentication Trust Anchor (proposed rule; FCC 25-25; WC Docket No. 17-97)](https://www.govinfo.gov/content/pkg/FR-2025-06-16/html/2025-10998.htm), 90 FR 25186-25187 (summary and dates). Checked 2026-09-24. [^24]: FCC 25-25 proposes that providers certify in their Robocall Mitigation Database filings whether they have implemented a non-IP caller ID authentication framework. Source: [Call Authentication Trust Anchor (proposed rule; FCC 25-25; WC Docket No. 17-97)](https://www.govinfo.gov/content/pkg/FR-2025-06-16/html/2025-10998.htm), 90 FR 25192. Checked 2026-09-24. [^25]: FCC 25-25 proposes that ATIS-1000095.v002 (in-band authentication over TDM) and ATIS-1000096 (out-of-band multiple STI-CPS) are developed, reasonably available and effective, and seeks comment on ATIS-1000105 (out-of-band agreed STI-CPS). Source: [Call Authentication Trust Anchor (proposed rule; FCC 25-25; WC Docket No. 17-97)](https://www.govinfo.gov/content/pkg/FR-2025-06-16/html/2025-10998.htm), 90 FR 25186-25192 (discussion of non-IP standards). Checked 2026-09-24. [^26]: FCC 25-25 proposes a compliance deadline two years after the rule's effective date and proposes to repeal the continuing extension that non-IP providers have held. Source: [Call Authentication Trust Anchor (proposed rule; FCC 25-25; WC Docket No. 17-97)](https://www.govinfo.gov/content/pkg/FR-2025-06-16/html/2025-10998.htm), 90 FR 25192 and proposed 47 CFR 64.6303 at 90 FR 25199. Checked 2026-09-24. [^27]: FCC 25-73 (Advancing IP Interconnection NPRM, released 2025-10-29) proposes forbearing from sections 251(c)(2) and 251(c)(6) and removing 47 CFR 51.305, 51.321 and 51.323 effective December 31 2028, ending incumbent LEC obligations that keep TDM interconnection in place. Source: [Advancing IP Interconnection (proposed rule; FCC 25-73; WC Docket Nos. 25-304 25-208 17-97)](https://www.govinfo.gov/content/pkg/FR-2025-11-26/html/2025-21324.htm), 90 FR 54266-54272. Checked 2026-09-24. [^28]: FCC 25-73 asks whether the Commission should mandate direct IP-to-IP interconnection but does not propose such a mandate. Source: [Advancing IP Interconnection (proposed rule; FCC 25-73; WC Docket Nos. 25-304 25-208 17-97)](https://www.govinfo.gov/content/pkg/FR-2025-11-26/html/2025-21324.htm), 90 FR 54274-54275. Checked 2026-09-24. [^29]: FCC 25-73 names SS7 signalling points with local switches, tandems and 911 selective routers as elements of the typical TDM network topology in use. Source: [Advancing IP Interconnection (proposed rule; FCC 25-73; WC Docket Nos. 25-304 25-208 17-97)](https://www.govinfo.gov/content/pkg/FR-2025-11-26/html/2025-21324.htm), 90 FR 54269. Checked 2026-09-24. [^30]: FCC 25-73 says sunsetting incumbent LEC interconnection obligations would not prohibit TDM interconnection; incumbents could continue to interconnect in TDM. Source: [Advancing IP Interconnection (proposed rule; FCC 25-73; WC Docket Nos. 25-304 25-208 17-97)](https://www.govinfo.gov/content/pkg/FR-2025-11-26/html/2025-21324.htm), 90 FR 54272. Checked 2026-09-24. [^31]: RFC 4694 requires the number marked as ported number in the ISUP Generic Address Parameter to become the telephone number of the routing tel URI. Source: [RFC 4694: Number Portability Parameters for the "tel" URI](https://www.rfc-editor.org/rfc/rfc4694.html), Section 5.2.4. Checked 2026-09-24. [^32]: RFC 3204 says that when the Content-Disposition handling parameter is absent on an ISUP or QSIG body, handling=required is assumed, so a UAS that does not understand the body returns 415 Unsupported Media Type; with handling=optional the UAS ignores it. Source: [RFC 3204: MIME media types for ISUP and QSIG Objects](https://www.rfc-editor.org/rfc/rfc3204.html), Section 6. Checked 2026-09-24. [^33]: Under RFC 3398, an ISUP cause value not listed in the mapping table SHOULD produce the default SIP response 500 Server Internal Error. Source: [RFC 3398: Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping](https://www.rfc-editor.org/rfc/rfc3398.html), Section 7.2.4.1, sentence preceding the table. Checked 2026-09-24. [^34]: The application/ISUP media type requires a version parameter identifying the ISUP variant and allows an optional base parameter naming a fallback specification. Source: [RFC 3204: MIME media types for ISUP and QSIG Objects](https://www.rfc-editor.org/rfc/rfc3204.html), Section 3.1. Checked 2026-09-24. [^35]: RFC 4694 says a node MAY put rn on the caller-information tel URI when the ISUP Jurisdiction Information Parameter is present; JIP does not populate the routing URI. Source: [RFC 4694: Number Portability Parameters for the "tel" URI](https://www.rfc-editor.org/rfc/rfc4694.html), Section 5.2.4. Checked 2026-09-24. [^36]: RFC 4666 registers SCTP port 2905 for M3UA. Source: [RFC 4666: Signaling System 7 (SS7) Message Transfer Part 3 (MTP3) - User Adaptation Layer (M3UA)](https://www.rfc-editor.org/rfc/rfc4666.html), Section 1.4.8. Checked 2026-09-24. [^37]: RFC 4666 (M3UA, September 2006, obsoleting RFC 3332) transports SS7 MTP3-User signalling such as ISUP and SCCP over IP using SCTP, between a Signalling Gateway and IP-based application server processes. Source: [RFC 4666: Signaling System 7 (SS7) Message Transfer Part 3 (MTP3) - User Adaptation Layer (M3UA)](https://www.rfc-editor.org/rfc/rfc4666.html), Abstract; Sections 1.2 and 1.3.1. Checked 2026-09-24. [^38]: Under RFC 4694, a network node MUST NOT retrieve number portability information for a geographical number if the tel URI already carries the npdi parameter. Source: [RFC 4694: Number Portability Parameters for the "tel" URI](https://www.rfc-editor.org/rfc/rfc4694.html), Section 5.1. Checked 2026-09-24. [^39]: When overlap is carried into SIP under RFC 3578, each arriving SAM refreshes timer T10 and triggers a new INVITE with the accumulated digits, and 4xx-6xx responses do not trigger an ISUP REL until T10 expires with no more SAMs. Source: [RFC 3578: Mapping of Integrated Services Digital Network (ISDN) User Part (ISUP) Overlap Signalling to the Session Initiation Protocol (SIP)](https://www.rfc-editor.org/rfc/rfc3578.html), Section 3.2. Checked 2026-09-24. [^40]: RFC 3578 states that the preferred approach for a gateway handling PSTN overlap signalling is to convert it into SIP en-bloc signalling. Source: [RFC 3578: Mapping of Integrated Services Digital Network (ISDN) User Part (ISUP) Overlap Signalling to the Session Initiation Protocol (SIP)](https://www.rfc-editor.org/rfc/rfc3578.html), Section 1 Introduction. Checked 2026-09-24. [^41]: RFC 4694 requires npdi on the routing tel URI when the Ported Number Translation Indicator bit in the ISUP Forward Call Indicators is set to 1. Source: [RFC 4694: Number Portability Parameters for the "tel" URI](https://www.rfc-editor.org/rfc/rfc4694.html), Section 5.2.4. Checked 2026-09-24. [^42]: RFC 3398 notes that ISUP has a presentation-restriction concept for the calling number and says gateways MAY support more sophisticated SIP privacy systems, without specifying a detailed mapping. Source: [RFC 3398: Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping](https://www.rfc-editor.org/rfc/rfc3398.html), Section 5.7 (privacy). Checked 2026-09-24. [^43]: ITU-T Q.1912.5, on interworking between SIP and BICC or ISUP, is in force in its January 2018 edition with Corrigendum 1 (07/18), and the March 2004 edition is superseded. Source: [ITU-T Q.1912.5: Interworking between session initiation protocol (SIP) and bearer independent call control protocol or ISDN user part](https://www.itu.int/rec/T-REC-Q.1912.5), Recommendation landing page, version table. Checked 2026-09-24. [^44]: RFC 3326 allows the Reason header in any request within a dialog and in any CANCEL request, which lets a gateway pass the ISUP release cause onward in BYE or CANCEL. Source: [RFC 3326: The Reason Header Field for the Session Initiation Protocol (SIP)](https://www.rfc-editor.org/rfc/rfc3326.html), Section 2; Section 3.4 example of a gateway receiving ISUP REL. Checked 2026-09-24. [^45]: RFC 6432 permits any SIP response except 100 Trying to carry a Reason header with a Q.850 cause code; it permits but does not mandate this. Source: [RFC 6432: Carrying Q.850 Codes in Reason Header Fields in SIP (Session Initiation Protocol) Responses](https://www.rfc-editor.org/rfc/rfc6432.html), Section 3. Checked 2026-09-24. [^46]: RFC 6432 recommends integrity protection (for example TLS or IPsec) for Reason headers carrying Q.850 codes because a wrong code could trigger the wrong announcement. Source: [RFC 6432: Carrying Q.850 Codes in Reason Header Fields in SIP (Session Initiation Protocol) Responses](https://www.rfc-editor.org/rfc/rfc6432.html), Section 4 Security Considerations. Checked 2026-09-24. [^47]: RFC 3326 defines the SIP Reason header with registered protocol values SIP and Q.850, where Q.850 carries an ITU-T Q.850 cause value in the cause parameter. Source: [RFC 3326: The Reason Header Field for the Session Initiation Protocol (SIP)](https://www.rfc-editor.org/rfc/rfc3326.html), Section 2. Checked 2026-09-24. [^48]: RFC 3398 (December 2002) is a Standards Track IETF document describing how a gateway maps ISUP signalling to SIP and SIP to ISUP. Source: [RFC 3398: Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping](https://www.rfc-editor.org/rfc/rfc3398.html), Header (Category line) and Abstract. Checked 2026-09-24. [^49]: RFC 3398 places overlap dialling (ISUP Subsequent Address Messages) outside its scope and assumes gateways collect all digits with timers before sending an INVITE. Source: [RFC 3398: Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping](https://www.rfc-editor.org/rfc/rfc3398.html), Section 4 (scope). Checked 2026-09-24. [^50]: Under RFC 3398, a SIP status code not listed in the SIP-to-ISUP table SHOULD be mapped to the default ISUP cause 16 (normal clearing). Source: [RFC 3398: Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping](https://www.rfc-editor.org/rfc/rfc3398.html), Section 8.2.6.1, sentence preceding the table. Checked 2026-09-24. [^51]: RFC 3372 (September 2002, BCP 63) defines SIP-T as a set of mechanisms for interfacing traditional telephone signalling with SIP, not a new protocol. Source: [RFC 3372: Session Initiation Protocol for Telephones (SIP-T): Context and Architectures](https://www.rfc-editor.org/rfc/rfc3372.html), Header (BCP 63) and Introduction. Checked 2026-09-24. [^52]: In SIP-T, full ISUP messages are carried end to end by encapsulating them in multipart MIME bodies of SIP messages, while key ISUP parameters are also translated into SIP headers for routing. Source: [RFC 3372: Session Initiation Protocol for Telephones (SIP-T): Context and Architectures](https://www.rfc-editor.org/rfc/rfc3372.html), Sections on encapsulation and translation. Checked 2026-09-24. [^53]: RFC 3372 recommends the SIP INFO method for carrying mid-call ISUP signalling that does not change SIP call state. Source: [RFC 3372: Session Initiation Protocol for Telephones (SIP-T): Context and Architectures](https://www.rfc-editor.org/rfc/rfc3372.html), Section on transfer of mid-call ISUP signalling messages. Checked 2026-09-24. [^54]: RFC 3372 sets three SIP-T requirements: transparency of ISUP information across a SIP segment, routability of SIP requests using header information, and transfer of mid-call signalling. Source: [RFC 3372: Session Initiation Protocol for Telephones (SIP-T): Context and Architectures](https://www.rfc-editor.org/rfc/rfc3372.html), Section on SIP-T requirements (transparency, routability, mid-call signalling). Checked 2026-09-24. [^55]: RFC 3398 maps a SIP 404 Not Found final response to ISUP cause value 1 (unallocated number) toward the PSTN. Source: [RFC 3398: Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping](https://www.rfc-editor.org/rfc/rfc3398.html), Section 8.2.6.1 SIP Status Code to ISDN Cause Code Mapping, table. Checked 2026-09-24. [^56]: RFC 3398 maps SIP 480 Temporarily Unavailable to ISUP cause value 19 (no answer from user). Source: [RFC 3398: Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping](https://www.rfc-editor.org/rfc/rfc3398.html), Section 8.2.6.1 SIP Status Code to ISDN Cause Code Mapping, table. Checked 2026-09-24. [^57]: RFC 3398 maps SIP 484 Address Incomplete to ISUP cause value 28 (address incomplete). Source: [RFC 3398: Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping](https://www.rfc-editor.org/rfc/rfc3398.html), Section 8.2.6.1 SIP Status Code to ISDN Cause Code Mapping, table. Checked 2026-09-24. [^58]: RFC 3398 maps both SIP 486 Busy Here and SIP 600 Busy Everywhere to ISUP cause value 17 (user busy). Source: [RFC 3398: Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping](https://www.rfc-editor.org/rfc/rfc3398.html), Section 8.2.6.1 SIP Status Code to ISDN Cause Code Mapping, table. Checked 2026-09-24. [^59]: RFC 3398 maps SIP 503 Service Unavailable to ISUP cause value 38 (network out of order). Source: [RFC 3398: Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping](https://www.rfc-editor.org/rfc/rfc3398.html), Section 8.2.6.1 SIP Status Code to ISDN Cause Code Mapping, table. Checked 2026-09-24. [^60]: RFC 3398 maps SIP 603 Decline to ISUP cause value 21 (call rejected). Source: [RFC 3398: Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping](https://www.rfc-editor.org/rfc/rfc3398.html), Section 8.2.6.1 SIP Status Code to ISDN Cause Code Mapping, table. Checked 2026-09-24.