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<rfc ipr="trust200902" docName="draft-wendt-stir-secure-sip-rtp-00" category="info" consensus="true" submissionType="IETF" tocInclude="true" sortRefs="true" symRefs="true">
  <front>
    <title abbrev="Securing SIP and RTP with STIR">Securing SIP and RTP with STIR Certificates</title>

    <author fullname="Chris Wendt">
      <organization>Somos, Inc.</organization>
      <address>
        <postal>
          <country>US</country>
        </postal>
        <email>chris@appliedbits.com</email>
      </address>
    </author>

    <date year="2026" month="November" day="01"/>

    <area>Applications and Real-Time</area>
    <workgroup>Secure Telephone Identity Revisited</workgroup>
    <keyword>SIP</keyword> <keyword>TLS</keyword> <keyword>mutual TLS</keyword> <keyword>certificates</keyword> <keyword>interconnect</keyword> <keyword>post-quantum</keyword>

    <abstract>


<?line 90?>

<t>Certificates issued within the Secure Telephone Identity Revisited (STIR) framework establish, through a governed issuance process, that a party holds authority over telephone numbers or a service provider code, and where a domain identifier is bound to that authority, that the same party controls a domain. This document defines a certificate profile for TLS built on the TNAuthList extension and on the binding of telephone number authority to a domain identifier, so that the provider operating an interface is identified by a service provider code and by a validated domain rather than by prior arrangement between the parties, and it specifies the use of TLS, mutual TLS, and media protection across that interface. It applies to the network to network interface between providers. It brings the use of TLS with SIP up to date with TLS 1.3 and current certificate identity practice, and places the issuance of the credentials on which interconnection depends within a defined governance and issuance framework.</t>



    </abstract>



  </front>

  <middle>


<?line 94?>

<section anchor="introduction"><name>Introduction</name>

<t>Interconnection between telephone service providers is moving onto the public internet, and increasingly onto arrangements in which any provider may exchange traffic with any other without a bilateral agreement having been established in advance. A provider operating such an interface needs to determine, at the moment a connection is presented to it, which provider is at the other end and whether it is prepared to exchange traffic with that provider. Where no agreement precedes the connection, the receiving provider holds no configuration describing the peer, and both determinations have to be made from what the peer presents.</t>

<t>A certificate carrying the TNAuthList extension <xref target="RFC8226"/> asserts authority over telephone numbers or a service provider code, and is issued only after an authority responsible for those numbers has validated that assertion. Where the certificate additionally carries a domain identifier bound to that authority <xref target="I-D.wendt-stir-tn-domain-binding"/>, the issuer has validated control of the domain in the same act, so the certificate identifies a provider by a service provider code and by a domain that a relying party can verify. The governance and issuance policy under which such certificates are granted is what determines who may obtain one, and therefore who may participate. Participation is established by holding a credential rather than by agreement between the parties.</t>

<t>What has been missing is a specification of how that credential is used at the interface itself: what a certificate presented for TLS contains, how a peer authenticates it, what a peer is entitled to conclude from it, and how the signaling and media it protects are secured. This document supplies that.</t>

<t>The specifications defining the use of TLS with SIP are a further obstacle. SIP <xref target="RFC3261"/> has supported TLS since publication, and the documents describing how that support works were largely complete by 2010. TLS 1.3 <xref target="RFC9846"/> has since replaced the handshake, removed renegotiation, and encrypted the certificate exchange; guidance for representing and verifying service identity in certificates was revised and republished as <xref target="RFC9525"/>, narrowing the identifier types in scope and removing practices the SIP documents still permit; recommendations for the secure use of TLS were consolidated in <xref target="RFC9325"/>; earlier TLS versions were deprecated <xref target="RFC8996"/>; and post-quantum key agreement has been introduced. None of these changes were made with SIP in mind, and none of the SIP documents have been revisited in light of them.</t>

<t>Implementations have diverged accordingly. Deployments make locally reasonable choices about which certificate identifier to populate and which to check, whether and how to require a client certificate, and how to decide that a presented certificate belongs to a peer whose traffic they will carry. Interconnection is then arranged bilaterally, and commonly falls back to address based access control, with TLS providing confidentiality but not a usable peer identity.</t>

<t>A further reason for this document is the transition to post-quantum cryptography. <xref target="NIST.IR.8547"/> deprecates the elliptic curve and RSA algorithms in current use for both key establishment and signatures after 2030 and disallows them after 2035. The SIP and STIR specifications predate that timeline, and interconnection equipment deployed now will be in service through it. This document identifies where the transition touches SIP over TLS, SRTP, and STIR, and what each step depends on, so that the pieces can be taken up in the forums that own them.</t>

<t>This document therefore does three things. It sets out the specifications that define the use of TLS and of media protection with SIP, and where each has been overtaken. It defines a profile of current practice for the interface between providers, including the certificate profile described above and the protection of both signaling and media across that interface. And it identifies the points at which existing specifications require change rather than profiling, which it does not undertake here.</t>

<section anchor="scope"><name>Scope</name>

<t>This document is about the certificates used for TLS between providers and what a peer concludes from them: what a certificate contains, how its subject is validated, how a peer authenticates and authorizes the party presenting it, and how the signaling and media it protects are keyed. The use of TLS as a SIP transport, including the TLS version, cipher suites, and how a certificate identifier is matched against a SIP domain, is specified by <xref target="I-D.wendt-sipcore-tls-updates"/>; this document narrows those rules for the interface it covers and does not restate them. It does not define new TLS mechanisms, new certificate extensions, or new cryptographic algorithms. Where this document recommends a behavior that differs from an existing specification, it identifies that specification and the nature of the difference; it does not formally update those documents.</t>

<t>This document applies to the network to network interface, that is, to the signaling and media exchanged between the border elements of two interconnected providers. It covers the protection of both signaling and media across that interface, since the trust in each derives from the same credential and the protection of one bounds what the other achieves.</t>

<t>The user to network interface, between a provider and its own subscribers, is out of scope. So is the treatment of signaling and media within a provider's network. This scoping is deliberate and distinguishes this profile from work that sought protection between user agents: at the network to network interface the two border elements are the endpoints of the segment being protected, and no intermediary lies between them to terminate a security association.</t>

<t>Protection described here applies to a single interface between two adjacent providers. Where a communication traverses more than one such interface, each is protected independently. This document does not provide, and does not claim, confidentiality between the calling and called parties.</t>

<t>This document does not define SRTP transforms or RTP payload formats, which are specified elsewhere and referenced here.</t>

</section>
</section>
<section anchor="conventions"><name>Conventions</name>

<t>The key words "<bcp14>MUST</bcp14>", "<bcp14>MUST NOT</bcp14>", "<bcp14>REQUIRED</bcp14>", "<bcp14>SHALL</bcp14>", "<bcp14>SHALL
NOT</bcp14>", "<bcp14>SHOULD</bcp14>", "<bcp14>SHOULD NOT</bcp14>", "<bcp14>RECOMMENDED</bcp14>", "<bcp14>NOT RECOMMENDED</bcp14>",
"<bcp14>MAY</bcp14>", and "<bcp14>OPTIONAL</bcp14>" in this document are to be interpreted as
described in BCP 14 <xref target="RFC2119"/> <xref target="RFC8174"/> when, and only when, they
appear in all capitals, as shown here.</t>

<?line -18?>

</section>
<section anchor="landscape"><name>The Existing Specification Landscape</name>

<t>This section surveys the documents that together define the use of TLS with SIP, and notes for each what it establishes and what has changed since.</t>

<section anchor="core-sip-documents"><name>Core SIP Documents</name>

<t><xref target="RFC3261"/> defines TLS as a SIP transport and the "sips" URI scheme that requests its use. Section 26 sets out the security architecture, including the expectation that peers inspect each other certificates on interdomain connections. Its transport requirements were written against TLS 1.0 <xref target="RFC5246"/> and its mandatory cipher suite, TLS_RSA_WITH_AES_128_CBC_SHA, uses static RSA key exchange, which does not exist in TLS 1.3. The base specification therefore states a mandatory floor that a conforming modern implementation cannot meet.</t>

<t><xref target="RFC3263"/> defines how a SIP URI is resolved to a transport, address, and port using NAPTR and SRV records, and is the mechanism by which a TLS connection target is selected. Because the name used to reach a host can differ from the name in the URI that was resolved, the resolution procedure and the certificate matching procedure interact, and that interaction is a recurring source of implementation disagreement.</t>

<t><xref target="RFC5630"/> clarifies the semantics of the "sips" URI scheme, which <xref target="RFC3261"/> left open to more than one reading. It establishes that "sips" requests TLS on each hop toward the target domain rather than end to end.</t>

</section>
<section anchor="certificate-identity-and-connection-management"><name>Certificate Identity and Connection Management</name>

<t><xref target="RFC5922"/> is the central document for certificate identity in SIP. It specifies which identifiers in a certificate may represent a SIP domain and how they are compared against the domain being reached. It establishes that a SIP domain identity <bcp14>SHOULD</bcp14> be conveyed as a subjectAltName of type uniformResourceIdentifier containing a "sip" URI with no user part, that a dNSName entry is accepted only if no matching "sip" URI entry is present, and that the Subject common name may be examined only if no subjectAltName appears at all. It also specifies the behavior of a peer that accepts a connection and authenticates the sending domain, which is the basis for mutual authentication in SIP as it exists today.</t>

<t>Two of those rules are now in conflict with <xref target="RFC9525"/>, which obsoletes the general service identity guidance that <xref target="RFC5922"/> was written against. <xref target="RFC9525"/> establishes that service identity can be expressed only in the subjectAltName extension and that the common name is no longer valid for this purpose, and it scopes itself to identifiers expressed as domain names, IP addresses, and SRV service names. A "sip" URI entry is outside that scope. The result is that the identifier type <xref target="RFC5922"/> prefers is not covered by current service identity guidance, while the identifier type it treats as a fallback, dNSName, is the one that general purpose TLS implementations actually verify.</t>

<t><xref target="RFC5923"/> defines connection reuse, allowing a TLS connection established in one direction to carry requests in the other. It requires that both peers be authenticated for such reuse, and is therefore the principal reason mutual TLS is used in SIP. It was written against a TLS version in which a server could request a client certificate through renegotiation, an option TLS 1.3 does not provide.</t>

<t><xref target="RFC5924"/>, an Experimental specification, defines an extended key usage, id-kp-sipDomain, marking a certificate as issued for SIP domain use and restricting the names in that certificate to use by a SIP service. Its deployment is limited, and the consequences of its presence or absence in a certificate used for SIP are not treated consistently.</t>

</section>
<section anchor="tls-and-service-identity"><name>TLS and Service Identity</name>

<t><xref target="RFC9846"/> defines TLS 1.3, which changes several properties the SIP documents assume: the handshake is restructured, renegotiation is removed in favor of post-handshake authentication, the certificate is encrypted, and session resumption and early data are redefined. <xref target="RFC9325"/> provides current recommendations for the secure use of TLS, and <xref target="RFC8996"/> deprecates TLS 1.0 and 1.1. <xref target="RFC6066"/> defines the server name indication extension, which is relevant wherever a single host serves more than one SIP domain, a case <xref target="RFC5922"/> anticipates through multiple certificate identities rather than through a negotiated server name.</t>

</section>
<section anchor="telephone-number-authority-in-certificates"><name>Telephone Number Authority in Certificates</name>

<t><xref target="RFC8226"/> defines the TNAuthList certificate extension, which conveys the telephone numbers, number ranges, or service provider codes for which a certificate holder is authorized. Although developed for signing PASSporTs, the extension is non critical and carries no restriction on the protocols in which the certificate may be used. <xref target="RFC9060"/> defines delegation of that authority to a subordinate certificate.</t>

<t><xref target="I-D.wendt-stir-tn-domain-binding"/> defines a binding between telephone number authority and a domain identifier, established by an issuing certification authority that validates control of the domain and a telephone number authority token within a single issuance, so that the resulting certificate attests that the same entity holds both.</t>

</section>
<section anchor="media-protection"><name>Media Protection</name>

<t>The protection of media is defined separately from the protection of signaling, and the relationship between them determines what protecting the signaling achieves. <xref target="RFC3711"/> defines SRTP. Where keys for SRTP are carried in the session description <xref target="RFC4568"/>, the confidentiality of the media depends entirely on the protection of the signaling that carries them. Where DTLS-SRTP <xref target="RFC5763"/> is used, the signaling carries only a certificate fingerprint <xref target="RFC8122"/> and the media keys are established in the media plane. A substantial amount of interconnected traffic uses neither, and carries media with no protection at all. <xref target="RFC8643"/> defines an opportunistic approach for negotiating SRTP where support is not known in advance, intended as a transitional mechanism for deployments moving toward encrypted media.</t>

<t><xref target="RFC8862"/> is the current best practice for securing media signaled with SIP. It requires DTLS-SRTP, and defines a profile of STIR in which a verification service confirms that the fingerprints carried in the session description match the keys exchanged in the DTLS-SRTP handshake, using the PASSporT <spanx style="verb">mky</spanx> claim to protect those fingerprints and an <spanx style="verb">msec</spanx> PASSporT type to signal compliance. It was written for an endpoint to endpoint model in which user agents implement both STIR roles, and it predates DTLS 1.3 <xref target="RFC9147"/> and the introduction of post-quantum key agreement. It also observes that media relays and back to back user agents frequently terminate media security associations, so that only hop by hop confidentiality is achievable in many deployments <xref target="RFC7879"/>.</t>

</section>
</section>
<section anchor="certificates"><name>Certificates for SIP over TLS</name>

<section anchor="domain-identity"><name>Domain Identity</name>

<t>A certificate used under this profile carries the domain of the interface in a subjectAltName dNSName entry, and a peer verifying the certificate verifies that entry, following the representation and matching rules of <xref target="I-D.wendt-sipcore-tls-updates"/>. The subjectAltName entry of type uniformResourceIdentifier containing a "sip" URI, which <xref target="RFC5922"/> prefers, is not used at this interface.</t>

<t>The identities carried in signaling across this interface are telephone numbers, expressed as a "tel" URI or as a "sip" URI with the user=phone parameter, in which the host portion identifies the domain to which the request is routed rather than forming part of the identity. A telephone number carries no domain of its own, and no association between a telephone number and a domain can be derived from the number itself. That association is established at issuance by the binding described in <xref target="I-D.wendt-stir-tn-domain-binding"/>, which is why a separate binding is required here and is not required for an identity of the form user@domain, where the domain is part of the identity.</t>

<t>The identity a peer verifies at this interface is therefore a domain name, and it is the same identifier the binding associates with telephone number authority. The domain that authenticates the transport peer and the domain that corroborates the entity holding the telephone numbers are the same value in the same certificate, verified by the same path validation.</t>

</section>
<section anchor="names"><name>Names in the Certificate</name>

<t>A certificate used under this profile can carry two kinds of name: a domain, and telephone number authority. Both appear in the certificate and both are examined by a peer, and this section says what each establishes.</t>

<t>The domain is the identifier a peer verifies. A provider publishes a domain at which its interface is reached, and a connecting peer verifies that domain against the presented certificate. Separately, a provider holds a certificate binding a domain to the telephone numbers or service provider code it is authorized for. A provider that operates its interface under a domain it also has bound presents one certificate that serves both, and a peer that validates it learns both which party it is connected to and what that party is authorized for.</t>

<t>Where the two domains differ, a peer learns less. If the interface is operated under a domain that carries no bound telephone number authority, validating the certificate establishes which party operates the interface and nothing about its authority over telephone numbers. A provider whose interface is operated on its behalf by another party, under that party's domain, is in this position, as is a provider that has bound one domain to its numbers and publishes its interface under another.</t>

<t>An element accepting a connection under this profile <bcp14>MUST</bcp14> therefore determine which of these cases applies rather than assume the first. Where the certificate presented for the connection is a bound certificate and the domain matched for the connection is the bound domain, the authorization determination of <xref target="mutual-tls"/> applies to the party at the other end of the connection. Where it is not, the authenticated identity is that of the party operating the interface only, and any determination about telephone number authority is made from the credential presented with each communication rather than from the transport connection.</t>

<t>Deployments <bcp14>SHOULD</bcp14> operate an interface under a domain bound to the telephone number authority of the provider on whose behalf traffic is carried, so that the transport connection and the traffic carried over it are attributable to the same party. Where an interface is operated by one party on behalf of another, that arrangement is visible to a peer as the difference described here, and this profile does not treat it as an error.</t>

<t>Telephone number authority is carried in the TNAuthList extension <xref target="RFC8226"/>, conveying the telephone numbers, number ranges, or service provider codes for which the subject is authorized. TNAuthList is a non critical extension, so a TLS implementation that does not recognize it ignores it and path validation is unaffected. <xref target="RFC8226"/> places no restriction on the protocols in which such a certificate may be used, and anticipates its use by protocols other than the ones for which it was defined. In an interconnect context the useful content is typically a service provider code identifying the provider operating the SIP endpoint. Where a certificate is scoped to specific telephone numbers, it is issued as a delegate certificate <xref target="RFC9060"/> whose scope is encompassed by that of its parent.</t>

<t>Of the identifiers a certificate can carry, this profile uses one for peer verification: the subjectAltName dNSName entry. The subjectAltName uniformResourceIdentifier entry carrying a "sip" URI, which <xref target="RFC5922"/> prefers, and the Subject common name are not used. Telephone number authority is read from the TNAuthList extension and is never used as a name to match.</t>

</section>
<section anchor="binding-domain-identity-to-telephone-number-authority"><name>Binding Domain Identity to Telephone Number Authority</name>

<t>A certificate carrying both a domain identifier and telephone number authority asserts more than either alone only if the two were validated together. <xref target="I-D.wendt-stir-tn-domain-binding"/> defines that binding: the issuing certification authority validates control of the domain and validates an authority token covering the telephone numbers or service provider codes, and treats the two as a single condition for issuance. The resulting certificate attests that the same entity holds both.</t>

<t>This is the property that makes mutual TLS useful between interconnected networks. Verifying a peer certificate establishes which domain the peer is, and <xref target="RFC5922"/> already specifies how to match it. What it does not establish is whether that domain is entitled to originate traffic for a given provider or set of numbers, which is why interconnect arrangements are made bilaterally and enforced by address. A bound certificate carries that authorization in the credential itself, validated by an issuer against the numbering authority.</t>

</section>
<section anchor="certificate-contents"><name>Certificate Contents</name>

<t>A certificate used under this profile carries the following:</t>

<t><list style="symbols">
  <t>A subjectAltName dNSName entry containing the SIP domain, as described above.</t>
  <t>Where telephone number authority is asserted, a TNAuthList extension <xref target="RFC8226"/>, bound to the domain identifier as described in <xref target="I-D.wendt-stir-tn-domain-binding"/>.</t>
  <t>An extended key usage extension <xref target="RFC5280"/> containing id-kp-serverAuth where the certificate is presented when accepting a connection, id-kp-clientAuth where it is presented when initiating one, or both where one certificate is used for both roles.</t>
  <t>A key usage extension containing digitalSignature.</t>
</list></t>

<t><xref target="RFC5924"/> defines id-kp-sipDomain to mark a certificate as issued for SIP domain use. Where it is present, it is present in addition to id-kp-serverAuth and id-kp-clientAuth rather than instead of them, since a TLS implementation that finds an extended key usage extension without the key purpose it requires will reject the certificate.</t>

<t>A peer that both accepts and initiates connections, as is normal where connection reuse is in use, <bcp14>MAY</bcp14> present one certificate carrying both key purposes or <bcp14>MAY</bcp14> hold two certificates, one for each role, with the same domain and the same telephone number authority in each. Both arrangements conform to this profile. The CA/Browser Forum has moved against issuing publicly trusted certificates carrying both id-kp-serverAuth and id-kp-clientAuth, and a certification authority that follows those requirements can issue for this profile only as two certificates. A certification authority issuing under <xref target="RFC8226"/> is not bound by those requirements, but permitting both arrangements lets either kind of issuer support the profile.</t>

<t>Separate certificates, with separate keying material, <bcp14>MUST</bcp14> be used for TLS session establishment and for PASSporT signing.</t>

<t>Both certificates may be issued by the same certification authority, on the basis of the same validated domain control and telephone number authority, so the separation adds an issuance rather than a validation. The two keys are held in different places and their compromise has different consequences, as discussed in <xref target="security"/>.</t>

<t>The two certificates also differ in profile. A certificate used for TLS carries the extended key usage values described above, and a certificate used to sign PASSporTs carries the claim constraints and validity period appropriate to that use. A single certificate satisfying both sets of requirements would be constrained by neither.</t>

</section>
</section>
<section anchor="algorithms"><name>Algorithms and TLS 1.3</name>

<section anchor="protocol-versions"><name>Protocol Versions</name>

<t><xref target="I-D.wendt-sipcore-tls-updates"/> requires TLS 1.3 <xref target="RFC9846"/> for SIP and permits TLS 1.2 for interoperability. This profile narrows that for the interface it covers: TLS 1.3 <bcp14>MUST</bcp14> be used, and earlier versions <bcp14>MUST NOT</bcp14> be, including TLS 1.2, and including TLS 1.0 and 1.1 which are deprecated in any case <xref target="RFC8996"/>. Cipher suite selection is determined by TLS 1.3 itself; the recommendations of <xref target="RFC9325"/> apply where they concern configuration outside the cipher suite.</t>

<t>This profile describes an interface that is newly deployed rather than an existing one being retrofitted, and requiring a single version removes the negotiation and downgrade considerations that supporting more than one entails. Two of the properties this profile depends on are available only in TLS 1.3. Ephemeral key agreement is unconditional, so the forward secrecy that <xref target="keying-choice"/> relies on is a property of the protocol rather than of the cipher suite selected. And the hybrid post-quantum key agreement groups referenced in <xref target="key-agreement"/> are defined for TLS 1.3, so a deployment negotiating an earlier version cannot meet the key agreement requirements of this profile.</t>

<t>Existing interconnection interfaces commonly specify TLS 1.2, and equipment deployed for them may require updating before it can participate in an interface conforming to this profile.</t>

<t>The version and cipher suite requirements of <xref target="RFC3261"/> predate TLS 1.3 and cannot be satisfied by a conforming TLS 1.3 implementation, as described in <xref target="landscape"/>. Updating them is outside the scope of this profile and is specified in <xref target="I-D.wendt-sipcore-tls-updates"/>.</t>

</section>
<section anchor="key-agreement"><name>Key Agreement</name>

<t>Key agreement determines whether traffic recorded today can be decrypted later. An adversary that records a TLS session and its handshake, and that later acquires the ability to solve the underlying problem the key agreement rests on, recovers the traffic keys. Ephemeral key exchange does not prevent this, because the property it provides is protection against later compromise of a long term key, not against the key agreement itself being broken.</t>

<t>This matters more for SIP than for many applications, because the confidentiality lifetime of the traffic is long. Call detail, calling and called numbers, and call patterns retain sensitivity for many years, and where the media keys are carried in the session description <xref target="RFC4568"/> the confidentiality of the conversation itself depends entirely on the protection of the signaling that carried them.</t>

<t>Implementations <bcp14>SHOULD</bcp14> therefore support a post-quantum key agreement group and <bcp14>SHOULD</bcp14> negotiate one where both peers support it. Either a hybrid group, which combines a post-quantum key encapsulation mechanism with an elliptic curve exchange so that the resulting secret is no weaker than the classical exchange alone, such as those defined in <xref target="I-D.ietf-tls-ecdhe-mlkem"/>, or a group using a post-quantum mechanism alone, such as those defined in <xref target="I-D.ietf-tls-mlkem"/>, satisfies this requirement. Which to prefer is a deployment choice; TLS 1.3 negotiates the group, so peers that prefer differently still agree on one both support. General guidance on adopting post-quantum cryptography in TLS based applications is given in <xref target="I-D.ietf-uta-pqc-app"/>.</t>

<t>Where DTLS-SRTP <xref target="RFC5763"/> is used, the media keys are established in the media plane and the signaling carries only a fingerprint <xref target="RFC8122"/>, from which nothing is recoverable. In that case post-quantum key agreement on the SIP connection protects signaling but not media, and equivalent protection of the media requires the same consideration in the media plane.</t>

</section>
<section anchor="signature-algorithms-and-certificates"><name>Signature Algorithms and Certificates</name>

<t>Certificates under this profile continue to use classical signature algorithms. Forging a signature at some future point permits impersonation from that point forward; it does not permit decryption of traffic recorded earlier. The retroactive risk lies in key agreement, not in authentication, and the two can therefore be addressed on different timescales. This also avoids the substantially larger certificates and handshakes that post-quantum signature algorithms currently entail.</t>

<t>Different timescales do not mean authentication can be left as it is. A signature verified today on a certificate or a PASSporT is relied on later, in records of which party connected and which party signed a communication, and an adversary that can later forge the signature can later dispute or fabricate those records. <xref target="pq-transition"/> sets out the order in which the transition applies to this interface and what the signature step depends on.</t>

</section>
<section anchor="interop"><name>Interoperability Considerations</name>

<t>Post-quantum and hybrid key agreement produce larger key shares than elliptic curve groups, which increases the size of the ClientHello and can cause it to be split across more than one packet. Implementations and intermediaries that assume a ClientHello arrives in a single record or a single datagram may fail in ways that are difficult to diagnose. SIP interconnection commonly traverses session border controllers and firewalls, so this population warrants testing rather than assumption.</t>

<t>An implementation that encounters a key agreement group it does not recognize is expected to select one it does support rather than to fail the handshake. Implementations that reject unknown groups outright prevent peers from offering new ones and should be identified during interoperability testing.</t>

<t>Early data, or 0-RTT, <bcp14>MUST NOT</bcp14> be used for SIP. Early data is replayable by design, and SIP requests are not safe to replay.</t>

</section>
<section anchor="pq-transition"><name>The Post-Quantum Transition</name>

<t><xref target="NIST.IR.8547"/> sets a timeline for moving off the quantum-vulnerable algorithms in current use: ECDSA, EdDSA, and RSA for signatures, and ECDH, finite field DH, and RSA for key establishment are deprecated after 2030 and disallowed after 2035. The equipment deployed under this profile will be in service across those dates.</t>

<t>Neither the SIPCORE nor the STIR working group is the forum for selecting post-quantum algorithms or their parameters. That work belongs to the TLS, LAMPS, and JOSE working groups, to NIST, and to the CA/Browser Forum for the Web PKI, and it is still settling. This document therefore does not name post-quantum algorithms beyond the key agreement groups already referenced in <xref target="key-agreement"/>, which are the part of the picture that has settled enough to deploy. What it does is state where the transition touches this interface, in what order, and what each step depends on, so that SIP and STIR follow the Web PKI as it settles rather than diverging from it.</t>

<t>Key agreement comes first, because it is the retroactive risk. Signaling and media recorded today can be decrypted later by an adversary that later obtains the ability to break the key agreement, and the confidentiality lifetime of call records is long. <xref target="key-agreement"/> and <xref target="sdes-requirements"/> already require post-quantum key agreement where both peers support it. That requirement depends on nothing outside TLS 1.3 and DTLS 1.3 and can be met now.</t>

<t>Signatures come second. Forging a signature at some future point permits impersonation and the fabrication of records from that point forward; it does not permit decryption of traffic already recorded. The transition can therefore follow the Web PKI's, and it has more dependencies: a post-quantum signature algorithm registered for X.509 by LAMPS, a certificate profile for STI certificates that permits it, a JOSE algorithm registration so that PASSporTs can be signed with it, and TLS signature scheme registrations for the handshake. The larger certificates and signatures that post-quantum algorithms entail also bear on SIP in a way they do not bear on the web: a PASSporT travels in a SIP header field, and the certificate chain in a TLS handshake traverses session border controllers with their own size limits. Those effects need measurement before a requirement is written.</t>

<t>Certificate transparency is part of the signature transition. STI certificate transparency <xref target="I-D.ietf-stir-certificate-transparency"/> is built on <xref target="RFC6962"/>, whose log signatures and Merkle tree hashes are classical. The IETF's work on post-quantum transparency logs, in the PLANTS working group, is where that transition is being worked out, and STI certificate transparency is expected to follow it rather than define its own.</t>

<t>Implementations under this profile <bcp14>SHOULD</bcp14> be built so that the signature algorithm, the key agreement group, and the certificate chain size they accept are configurable rather than fixed, and <bcp14>SHOULD</bcp14> support the negotiation mechanisms that let a peer offer a new algorithm without failing the handshake with a peer that lacks it, as described in <xref target="interop"/>. That is the property that lets the transition proceed by configuration when the dependencies above are met.</t>

</section>
</section>
<section anchor="securing-the-media"><name>Securing the Media</name>

<section anchor="media-baseline"><name>Media Security Baseline</name>

<t>Media across the interface <bcp14>MUST</bcp14> be protected using SRTP <xref target="RFC3711"/>. Two mechanisms for establishing the SRTP keys are in use, and both are supported by this profile.</t>

<t>Signaling anchored keying, using security descriptions carried in the session description <xref target="RFC4568"/>, places the SRTP master keys in the SIP signaling. It is the mechanism in general use at interconnection interfaces today. Where it is used, every element on the signaling path holds the media keys, and the confidentiality of the media is therefore no stronger than the protection of the signaling on every hop that carries it.</t>

<t>Media anchored keying, using DTLS-SRTP <xref target="RFC5763"/> <xref target="RFC5764"/>, establishes the SRTP keys through a handshake on the media path. The signaling carries only a fingerprint of the endpoint credential <xref target="RFC8122"/>, from which the keys cannot be derived.</t>

<t>Either mechanism <bcp14>MAY</bcp14> be used. Both provide equivalent protection of the media against an attacker on the interface itself when the requirements of this profile are met. They differ in how far the keying material propagates beyond that interface, and in operational cost. <xref target="keying-choice"/> sets out both so that a deployment can select on the basis of what it is exposed to. Where signaling anchored keying is used, the requirements of <xref target="sdes-requirements"/> apply.</t>

<t>Where a peer is not known to support media protection at all, <xref target="RFC8643"/> <bcp14>MAY</bcp14> be used to negotiate it opportunistically. This is a transitional measure for moving deployments away from unprotected media, not a durable configuration, and it provides no protection against an attacker able to modify the signaling.</t>

<t>This document does not define SRTP transforms. A transform is the pair of algorithms SRTP applies to each packet, one to encrypt the payload and one to authenticate the packet, and the transforms available are defined separately from the keying mechanisms that establish the master key. <xref target="RFC3711"/> defines AES in counter mode with HMAC-SHA1 authentication, <xref target="RFC6188"/> adds AES with 192 and 256 bit keys in counter mode, and <xref target="RFC7714"/> defines AES-GCM, in which encryption and authentication are performed as a single authenticated encryption operation. Further transforms are under development in the IETF.</t>

<t>A transform is not negotiated directly. It is selected through a named set that also fixes the key length, the salt length, and the length of the authentication tag: a crypto suite where security descriptions are used <xref target="RFC4568"/>, or a protection profile where DTLS-SRTP is used <xref target="RFC5764"/>. Both are registered with IANA.</t>

<t>The transform in most common use at interconnection interfaces today, AES_CM_128_HMAC_SHA1_80, authenticates packets with HMAC-SHA1. SHA-1 has been withdrawn by NIST and is no longer considered secure. Implementations <bcp14>MUST</bcp14> support an AES-GCM transform <xref target="RFC7714"/> and <bcp14>SHOULD</bcp14> negotiate it where both peers support it. Where an HMAC-SHA1 transform is negotiated for interoperability with a peer that supports nothing else, the deployment accepts that packet authentication rests on a withdrawn algorithm. A transform that does not rest on SHA-1 for either function is expected from the IETF, and this profile will reference it when it is available.</t>

</section>
<section anchor="keying-choice"><name>Properties of the Two Mechanisms</name>

<t>The two mechanisms are equivalent in most respects and differ in one that is a security property and two that are operational. A deployment selects between them on the basis of what it is exposed to, and both are permitted by this profile.</t>

<t>The difference is smaller at this interface than the design of media anchored keying suggests. Media anchored keying was specified for use with SIP, but its mandatory use and its deployment at scale followed from browser based communications, in which the signaling is carried by a party that is not a party to the media <xref target="RFC8827"/>. In that arrangement, keys placed in the signaling are disclosed to the signaling party by construction, and withholding them from the signaling is the property the mechanism exists to provide. At the interface described here the two parties to the signaling are the two parties to the media, and no third party is present in the signaling path from which keys need to be withheld. That property is therefore already provided by the topology, and the differences that remain are the ones described below.</t>

<t>Arrangements in which media is exchanged among more than two elements are a further reason media anchored keying is preferred in SIP deployments generally. Conference mixing, announcement and interactive response functions, recording, and transcoding introduce media paths that differ from the signaling path and involve elements that are not parties to the signaling. Those functions are performed within a provider's network, on legs other than the interconnection leg. The media association for a communication crossing the interface described here is between two elements. Where an intermediary is present at the interface itself, as in an arrangement where one provider receives traffic on behalf of another, that element terminates the association and re-originates it, which produces a sequence of two party legs rather than a media path shared among several elements.</t>

<section anchor="protection-across-the-interface"><name>Protection Across the Interface</name>

<t>Where the signaling is protected as required by this profile, with TLS 1.3, mutual authentication, and ephemeral key agreement on every hop, the two mechanisms protect the media across the interface equally.</t>

<t>In both cases the keys are held by the two border elements, which are the endpoints of the interface and hold the media in the clear regardless, so neither protects a provider against its peer. In both cases the channel that establishes or carries the keys provides forward secrecy, so an adversary that later obtains a long term private key recovers neither. In both cases the residual exposure is to a break of the key agreement itself, which is addressed for both by the requirements of <xref target="algorithms"/>.</t>

<t>Neither mechanism therefore offers better protection than the other against an attacker observing the interface.</t>

</section>
<section anchor="propagation-of-keying-material-beyond-the-interface"><name>Propagation of Keying Material Beyond the Interface</name>

<t>At this interface the signaling and the media are exchanged between the same two border elements, and commonly over the same path. An attacker positioned on that path observes both under either mechanism, and what it observes of the signaling is protected by TLS in both cases. The mechanisms therefore do not differ in what they expose to an attacker on the path.</t>

<t>They differ in where the keying material exists after the exchange. Under media anchored keying the signaling carries only a fingerprint of the endpoint credential, from which no key can be derived, and the keys exist only in the elements that terminate the media. Under signaling anchored keying the keys are carried in the session description, and are therefore present wherever that signaling is present. Signaling is captured for tracing, retained for operational and regulatory purposes, and exported to monitoring, analytics, and accounting systems as a matter of ordinary operation. The keys consequently exist in data whose lifetime is unrelated to the duration of the communication and in systems that have no role in handling media.</t>

<t>The exposure this creates is compound rather than positional. An attacker that captures media from the path, and that at any later time obtains the corresponding signaling from a system that retained or received it, recovers the media. Capture from the path alone is not sufficient, and access to the retained signaling alone is not sufficient. Under media anchored keying the second does not assist the first, and media captured from the path remains unrecoverable regardless of what signaling is subsequently obtained.</t>

<t>Signaling anchored keying also couples the two planes. Any failure to protect the signaling on any hop that carries the keys is a failure of media confidentiality, whereas under media anchored keying a compromise of the signaling discloses the fact and the parties of a communication but not its content.</t>

<t>The requirements of <xref target="sdes-requirements"/> apply to the interface, but the handling of signaling within each provider's own network is outside the scope of this profile, and signaling is commonly carried there without protection. The condition on which signaling anchored keying depends is therefore one a deployment can satisfy at the interface and cannot verify beyond it. Media anchored keying does not depend on that condition.</t>

</section>
<section anchor="operational-cost"><name>Operational Cost</name>

<t>Media anchored keying requires a handshake on the media path before media can flow. This adds to the delay before a communication is established and must be performed for every communication, which at interconnection volumes is a material load on the border element. Signaling anchored keying adds nothing beyond the signaling already exchanged.</t>

<t>Authenticated encryption transforms bound the amount of data that may be protected under a single key, and long lived communications may approach that bound. Media anchored keying can rekey within the media plane. Signaling anchored keying requires a new offer and answer, which deployments are correspondingly less likely to perform.</t>

</section>
<section anchor="selecting-a-mechanism"><name>Selecting a Mechanism</name>

<t>A deployment whose interface traverses the public internet, or whose signaling is retained, exported, or handled beyond the interface in ways it cannot constrain, is exposed to the propagation described above and should weigh it accordingly. A deployment for which call setup delay or handshake load at the border is the dominant constraint, or which interconnects with peers that do not support DTLS-SRTP, has a corresponding reason to use signaling anchored keying.</t>

<t>Both mechanisms are in use and both are permitted here. What this profile requires in either case is that the protection on which the chosen mechanism depends is actually present: for signaling anchored keying, that every hop carrying the keys is protected as required by <xref target="sdes-requirements"/>, and for both, that the key agreement protecting the communication meets the requirements of <xref target="algorithms"/>.</t>

</section>
</section>
<section anchor="sdes-requirements"><name>Requirements for Signaling Anchored Keying</name>

<t>Where security descriptions <xref target="RFC4568"/> are used to establish SRTP keys, every signaling hop that carries the keys <bcp14>MUST</bcp14> be protected as described in <xref target="algorithms"/>. A deployment <bcp14>MUST NOT</bcp14> carry security descriptions over an unprotected signaling hop, and <bcp14>MUST NOT</bcp14> accept them on one.</t>

<t>Because the media keys are present in the signaling, they are available to every element on the signaling path. A deployment using signaling anchored keying accepts that the confidentiality of the media extends only to parties outside that set of elements.</t>

</section>
<section anchor="relationship-to-the-existing-best-practice"><name>Relationship to the Existing Best Practice</name>

<t><xref target="RFC8862"/> defines the current best practice for securing media signaled with SIP, and this profile does not replace it. Where its conditions can be met, the binding it defines, in which a PASSporT carrying the <spanx style="verb">mky</spanx> claim protects the fingerprints in the session description, remains the strongest available assurance that the keys exchanged in the media plane are those the authenticated originator intended.</t>

<t>Two of its assumptions do not hold in telephone network interconnection. It places the authentication and verification service roles in the user agents, whereas in interconnection those roles are commonly performed by provider infrastructure. And it observes that few certification authorities issue certificates for telephone numbers, which is no longer the case: the credential described in <xref target="certificates"/> is issued for exactly that purpose and is chained to an authority for the numbers concerned.</t>

<t>The consequence is that the fingerprint binding of <xref target="RFC8862"/> can be performed using the same credential that authenticates the signaling peer and that carries telephone number authority. Where the media security association is not terminated between the signing entity and the verifying entity, this yields an assurance about the media that is anchored in the same certificate as the assurance about the identity.</t>

</section>
<section anchor="key-agreement-for-media"><name>Key Agreement for Media</name>

<t>The DTLS handshake that establishes SRTP keys is subject to the same consideration described in <xref target="key-agreement"/> for signaling. An adversary that records the handshake and the media, and that later acquires the ability to break the key agreement, recovers the media keys and therefore the conversation.</t>

<t>Implementations <bcp14>SHOULD</bcp14> support DTLS 1.3 <xref target="RFC9147"/> for DTLS-SRTP, and <bcp14>SHOULD</bcp14> negotiate a hybrid post-quantum key agreement group where both peers support it.</t>

<t>Which key agreement protects the conversation depends on the keying mechanism. Under media anchored keying it is the one in the media handshake, and protecting the signaling alone does not protect the media. Under signaling anchored keying it is the one on every signaling hop that carries the keys, and a recorded communication is recovered in full if that key agreement is broken, since the media keys were carried within it. This is a difference in the number of operations an adversary performs rather than in the capability it requires: an adversary able to break the key agreement protecting the signaling is able to break the one protecting the media handshake, and where both are recorded from the same path, media anchored keying requires it to break the second as well rather than presenting a different obstacle. The requirement in this section applies to both handshakes for that reason.</t>

</section>
<section anchor="termination-of-media-security-associations"><name>Termination of Media Security Associations</name>

<t>An assurance about media that is anchored at an originating endpoint is lost wherever an element terminates the media security association and re-originates the media, as described in <xref target="RFC7879"/>. This is the principal reason that end to end media security has not been achieved in telephone networks, and it is why <xref target="RFC8862"/> observes that only hop by hop confidentiality is available in many deployments.</t>

<t>Within the scope of this profile the difficulty does not arise. The two border elements are the endpoints of the interface being protected, and no element lies between them to terminate the association. The assurance this profile provides is an assurance about one interface, established between the two providers that operate its endpoints.</t>

<t>An assurance that survives termination by an intermediary, and therefore extends across more than one interface or into a provider's own network, is a different problem. It is identified in <xref target="gaps"/> and is not addressed here.</t>

</section>
<section anchor="keying-and-equipment"><name>Keying Mechanisms and Deployed Equipment</name>

<t>The two keying mechanisms are affected differently by the move to TLS 1.3 and by the post-quantum transition, and the difference reverses the usual ordering of their strength for the time being.</t>

<t>Under signaling anchored keying, the media keys travel in the signaling, so whatever protects the SIP connection protects them. A border element that negotiates TLS 1.3 with a post-quantum key agreement group on its SIP connection has, by that act alone, protected the SRTP keys against later recovery. Nothing changes on the media path: the SRTP transform, the packet format, and the media gateway behind the border element are untouched. The exposure that remains is the one <xref target="sdes-requirements"/> describes, that every element on the signaling path holds the keys.</t>

<t>Under media anchored keying, the media keys come from the DTLS handshake on the media path, and the protection of the SIP connection does not reach them. Until that handshake runs DTLS 1.3 with a post-quantum key agreement group, the media is protected by DTLS 1.2 key agreement, which is exactly the recorded-now-recovered-later exposure that <xref target="key-agreement"/> addresses for signaling. For the present, therefore, signaling anchored keying over a TLS 1.3 connection with post-quantum key agreement gives stronger protection against that attack than DTLS-SRTP as deployed, while DTLS-SRTP retains its advantage of keeping the keys off the signaling path. A deployment choosing between them should weigh which of the two it is more exposed to.</t>

<t>What has to change in deployed equipment follows from this. For signaling anchored keying, the border element's TLS stack is the only thing that changes: TLS 1.3, the post-quantum key agreement groups, and the larger ClientHello they produce (<xref target="interop"/>). Media gateways and session border controllers that terminate SRTP need, at most, support for an authenticated encryption transform in place of HMAC-SHA1 (<xref target="media-baseline"/>), which is a transform change rather than a keying change. For media anchored keying, the DTLS-SRTP implementation in the media path, which in telephone networks is a session border controller or media gateway rather than an endpoint, has to move to DTLS 1.3 and carry the same post-quantum groups, and the specification gap identified in <xref target="gaps"/> has to be closed first so that implementers converge on one behavior. That is a larger change, to equipment with longer replacement cycles than the border element's TLS stack, and it is why this profile treats post-quantum key agreement on the media handshake as a <bcp14>SHOULD</bcp14> that follows the signaling rather than a requirement that can be met now.</t>

</section>
</section>
<section anchor="media-resilience"><name>Media Resilience</name>

<t>Telephone network interconnection increasingly traverses paths that are not engineered for loss, while the RTP profile in common use for interconnection predates the mechanisms developed for such paths. <xref target="RFC8834"/> profiles the use of RTP for browser based communications, and requires feedback and resilience mechanisms that interconnection deployments generally do not negotiate.</t>

<t>The following are applicable to interconnected voice and are not precluded by anything in this profile:</t>

<t><list style="symbols">
  <t>Codec level redundancy, where the codec provides it. The in band forward error correction of Opus <xref target="RFC6716"/> recovers isolated losses at low cost, and is effective for speech where the codec is available.</t>
  <t>Redundant encoding <xref target="RFC2198"/>, which carries earlier payloads alongside current ones and is generally more efficient for speech than generic forward error correction.</t>
  <t>Generic forward error correction <xref target="RFC8627"/>, which is applicable but carries higher overhead for the packet sizes typical of voice.</t>
  <t>RTCP feedback <xref target="RFC4585"/>, and the corresponding profile for use with SRTP <xref target="RFC5124"/>, without which a sender has no timely indication of loss and no basis for adaptation.</t>
</list></t>

<t>Two constraints govern whether these are useful in practice. The first is codec availability: interconnection continues to carry a large volume of traffic using codecs that provide no in band redundancy, and the resilience available to such traffic is limited to mechanisms applied above the codec. The second is intermediary behavior: elements that transcode, that do not pass unrecognized payload types, or that do not forward RTCP feedback will defeat these mechanisms regardless of what the endpoints negotiate.</t>

<t>Negotiation is where these mechanisms most often fail in practice rather than in their operation. This is the same pattern described for certificate identity in <xref target="landscape"/>, and it is identified in <xref target="gaps"/> accordingly.</t>

</section>
<section anchor="gaps"><name>Identified Protocol Gaps</name>

<t>The preceding sections describe what can be adopted as it stands and what this profile constrains. This section identifies where existing specifications require change, which this document does not undertake.</t>

<t>The TLS version and cipher suite requirements of <xref target="RFC3261"/>, and the certificate identity rules of <xref target="RFC5922"/>, cannot be reconciled with current TLS practice by profiling alone. Updates to both are specified in <xref target="I-D.wendt-sipcore-tls-updates"/>.</t>

<t>The authorization of an authenticated peer, as distinct from its authentication, has no specification for SIP. <xref target="RFC5922"/> defines how a peer domain is matched but not how a receiving network determines that the domain is entitled to originate for a given provider or set of telephone numbers. This profile addresses that through the credential described in <xref target="certificates"/>; whether a corresponding SIP level mechanism is required is left open.</t>

<t>An identity assurance about media that survives termination of the media security association by an intermediary is not available. Whether one can be defined, and at what layer, is a question this profile identifies rather than answers.</t>

<t>The use of DTLS-SRTP with DTLS 1.3 is not specified. <xref target="RFC5764"/> and the fingerprint handling of <xref target="RFC8842"/> were written against DTLS 1.2 <xref target="RFC6347"/>, and the SRTP protection profiles and the <spanx style="verb">use_srtp</spanx> extension have not been restated for DTLS 1.3 <xref target="RFC9147"/>, whose record layer and key schedule differ. The key exporter that <xref target="RFC5764"/> relies on survives in DTLS 1.3, so the mechanism is expected to carry over, but an implementer following the published specifications arrives at DTLS 1.2, and that is what deployed implementations run. The consequence for this profile is stated in <xref target="keying-and-equipment"/>.</t>

<t>Several choices in this profile are about TLS and DTLS rather than about SIP, and are best judged by the working group responsible for those protocols. The TLS working group's review is sought on: requiring TLS 1.3 alone at a newly deployed interface; the treatment of post-quantum key agreement in <xref target="key-agreement"/>, <xref target="pq-transition"/>, and <xref target="keying-and-equipment"/>, including whether hybrid and pure groups should both be permitted; the client authentication and connection management choices of <xref target="connection-management"/>, which follow from the removal of renegotiation; the prohibition of early data; and the DTLS 1.3 gap above. Review by the working groups responsible for SRTP and for the DTLS-SRTP specifications is sought on the transform and keying questions of <xref target="media-baseline"/> and <xref target="keying-and-equipment"/>. The STIR working group is where the certificate and algorithm dependencies identified in <xref target="pq-transition"/> are coordinated, since the certificates are STIR certificates; the SIP transport changes that this profile narrows are the subject of <xref target="I-D.wendt-sipcore-tls-updates"/> in SIPCORE.</t>

<t>The negotiation of media resilience mechanisms for interconnection, and the behavior of intermediaries with respect to them, may require a profile or protocol work in the working group responsible for RTP. This document identifies the requirement and does not specify it.</t>

</section>
<section anchor="mutual-tls"><name>Mutual TLS and Peer Authorization</name>

<t>At the interface described by this profile both peers are providers, and each has the same interest in knowing which provider it is exchanging traffic with. Connections between border elements <bcp14>MUST</bcp14> therefore be mutually authenticated. The element initiating the connection <bcp14>MUST</bcp14> present a certificate conforming to <xref target="certificates"/>, and the element accepting the connection <bcp14>MUST</bcp14> request one and <bcp14>MUST</bcp14> verify it.</t>

<section anchor="authentication"><name>Authentication</name>

<t>A peer verifying a presented certificate <bcp14>MUST</bcp14> validate the certification path to a trust anchor it accepts, and <bcp14>MUST</bcp14> verify that the domain of the peer it expected to reach matches a dNSName entry in the certificate, applying the rules referenced in <xref target="domain-identity"/>.</t>

<t>Both steps are required. Validating the path alone establishes only that a certification authority the verifier trusts issued the certificate to some party, and for a widely trusted authority that is a large set of parties. A deployment that requests a client certificate but does not verify the name within it will accept any peer holding a certificate from any accepted authority. This configuration is known to occur.</t>

<t>An accepting element <bcp14>MAY</bcp14> use the certificate authorities extension of the TLS CertificateRequest message <xref target="RFC9846"/> to indicate the trust anchors it accepts, which allows an initiating element that holds more than one certificate to present the appropriate one.</t>

</section>
<section anchor="authorization"><name>Authorization</name>

<t>Authentication corroborates the domain of the peer. It does not establish that the peer is entitled to send the traffic it is sending. That is a separate determination, and one for which SIP provides no mechanism.</t>

<t>Where the presented certificate carries a TNAuthList extension <xref target="RFC8226"/> bound to the domain identifier as described in <xref target="I-D.wendt-stir-tn-domain-binding"/>, the service provider code it contains identifies the provider operating the peer, as validated by the issuer rather than as asserted by the peer. An accepting element <bcp14>MAY</bcp14> use that value to determine whether the peer is one with which it has an interconnection relationship, and to apply the policy associated with that relationship.</t>

<t>This is a determination about the provider operating the interface. It is not a determination about the telephone numbers of any individual communication, which remains the function of the STIR credential presented with that communication. An accepting element that applies both <bcp14>MAY</bcp14> require consistency between them, treating a communication whose signing credential is unrelated to the authenticated peer as a matter for local policy.</t>

</section>
<section anchor="no-prior-arrangement"><name>Interconnection Without Prior Arrangement</name>

<t>This profile supports interconnection in which two providers exchange traffic without a bilateral arrangement established in advance. Where interconnection is bilateral, a provider authorizes a peer from local configuration: an address range, a pinned certificate, or a trust anchor provisioned for that peer. A provider receiving a connection from a participant with which it has not previously exchanged traffic has no such configuration to consult.</t>

<t>Authorization therefore derives from the credential. A provider presenting a certificate that conforms to <xref target="certificates"/> and validates to a trust anchor the receiving provider accepts has demonstrated to its issuer control of the domain it presents and authority over the telephone numbers or service provider code the certificate asserts <xref target="I-D.wendt-stir-tn-domain-binding"/>. The set of trust anchors the participants accept in common determines the set of providers able to interconnect. The criteria under which certificates are issued and revoked are determined by the authority operating the certification hierarchy and are outside the scope of this document.</t>

<t>Two consequences follow for an element implementing this profile. A participant is excluded by revocation of its certificate, or by the issuer declining to reissue it where short lived certificates are used, rather than by a configuration change at every peer. And validating a presented certificate does not oblige a provider to carry the traffic; local policy continues to apply, and a provider may decline traffic from an authenticated participant on any grounds.</t>

</section>
<section anchor="trust-anchors"><name>Trust Anchors</name>

<t>Certificates presented at this interface <bcp14>MUST</bcp14> be issued by a certification authority operating under <xref target="RFC8226"/>, and an element <bcp14>MUST NOT</bcp14> accept a certificate for this interface on the basis of a trust anchor from a certification hierarchy that does not issue under <xref target="RFC8226"/>.</t>

<t>A certificate used here carries telephone number authority in a TNAuthList extension, co-validated with a domain identifier <xref target="I-D.wendt-stir-tn-domain-binding"/>. A certification authority that validates only control of a domain does not validate authority over telephone numbers and cannot issue a certificate carrying that assertion. Accepting a trust anchor from such a hierarchy would admit any party that has demonstrated control of any domain.</t>

<t>An element <bcp14>SHOULD</bcp14> be able to add a trust anchor in advance of its use, rather than in response to a handshake failure.</t>

</section>
<section anchor="caa"><name>Limiting Issuance for an Interface Domain</name>

<t>A provider that operates its interface under a domain can limit which certification authorities may issue certificates for that domain, and through which ACME account, by publishing a CAA record <xref target="RFC8659"/>. <xref target="I-D.wendt-stir-tn-domain-binding"/> defines the <spanx style="verb">issuesti</spanx> Property Tag for bound certificates, with the <spanx style="verb">accounturi</spanx> and <spanx style="verb">validationmethods</spanx> parameters of <xref target="RFC8657"/>, and a certification authority issuing a bound certificate for use under this profile processes it as that document specifies. The <spanx style="verb">issue</spanx> and <spanx style="verb">issuewild</spanx> Properties of <xref target="RFC8659"/> do not govern these certificates, following the approach of <spanx style="verb">issuemail</spanx> <xref target="RFC9495"/>, so a provider's existing CAA records for its Web PKI certificates neither block nor authorize them.</t>

<t>Where domain control is validated with a standing record, as with the dns-persist-01 challenge <xref target="I-D.ietf-acme-dns-persist"/> that <xref target="I-D.wendt-stir-tn-domain-binding"/> lists, the CAA record and the validation record together name the certification authority and account the provider has authorized, in two records at the provider's domain.</t>

</section>
</section>
<section anchor="connection-management"><name>Connection Management</name>

<t>The items in this section are consequences of using the certificate for mutual authentication rather than properties of SIP over TLS generally. Connection reuse requires both peers to be authenticated, which requires the client certificate, which TLS 1.3 requires in the initial handshake.</t>

<section anchor="connection-reuse"><name>Connection Reuse</name>

<t>Connection reuse <xref target="RFC5923"/> allows a connection established in one direction to carry requests in the other, and requires that both peers be authenticated. That condition is met by <xref target="mutual-tls"/>, so reuse is applicable across this interface and <bcp14>SHOULD</bcp14> be used, since establishing a connection per direction doubles the handshake load and the number of connections to be maintained without adding any property.</t>

<t>An element that reuses a connection applies the authorization determination of <xref target="mutual-tls"/> to requests received on it in either direction. The authenticated identity of the peer does not change with the direction of a request.</t>

</section>
<section anchor="client-authentication-and-the-handshake"><name>Client Authentication and the Handshake</name>

<t>In TLS 1.3 a server cannot obtain a client certificate by renegotiating an established connection, as renegotiation does not exist. An accepting element that requires client authentication <bcp14>MUST</bcp14> therefore request the certificate in the initial handshake. Post-handshake authentication is defined by <xref target="RFC9846"/> but is not widely implemented, and this profile does not rely on it.</t>

<t>Early data <bcp14>MUST NOT</bcp14> be used, as stated in <xref target="interop"/>.</t>

</section>
<section anchor="long-lived-connections"><name>Long Lived Connections</name>

<t>Connections between border elements are typically long lived, while the certificates used to establish them are not, particularly where the practices described in <xref target="operational"/> are followed. Expiry of a certificate does not invalidate a connection already established under it, and an element <bcp14>SHOULD NOT</bcp14> tear down an established connection solely because the certificate that authenticated it has expired. The renewed certificate is presented at the next handshake.</t>

<t>An element <bcp14>SHOULD</bcp14> perform a TLS key update on a long lived connection in accordance with the limits applicable to the negotiated cipher suite.</t>

</section>
<section anchor="server-name-indication"><name>Server Name Indication</name>

<t>Where a border element serves more than one domain on a single address, an initiating element <bcp14>SHOULD</bcp14> include the server name indication extension <xref target="RFC6066"/> carrying the domain it intends to reach, so that the accepting element can present the corresponding certificate. An accepting element that serves a single domain does not depend on it, and <bcp14>MUST NOT</bcp14> reject a connection solely because it is absent.</t>

</section>
</section>
<section anchor="operational"><name>Operational Considerations</name>

<t>Certificates used for TLS under this profile are expected to be short-lived, as described in <xref target="I-D.ietf-stir-certificates-shortlived"/>, and a border element renews its TLS certificate on the same cadence as its signing certificate. Renewal does not interrupt established connections (<xref target="connection-management"/>).</t>

<t>Further operational considerations will be added as the profile is reviewed.</t>

</section>
<section anchor="security"><name>Security Considerations</name>

<t>This profile relies on the security properties of TLS 1.3 <xref target="RFC9846"/>, SRTP <xref target="RFC3711"/>, DTLS-SRTP <xref target="RFC5763"/>, and STIR certificates <xref target="RFC8226"/>, and on the binding of domain identity to telephone number authority <xref target="I-D.wendt-stir-tn-domain-binding"/>. The considerations of those documents, and of <xref target="RFC9325"/> and <xref target="RFC9525"/>, apply.</t>

<t>Separate keys for TLS and for PASSporT signing are required because the two are held in different places and their compromise has different consequences. A TLS key is present on every border element, the most exposed and widely distributed part of a deployment, and its compromise permits impersonation at the transport layer until the certificate expires. A signing key asserts authority over telephone numbers, and its compromise permits an attacker to originate signed communications from any number within the certificate's scope. Using one key for both places the second consequence on every element that terminates TLS.</t>

<t>The transform in common use for SRTP at interconnection interfaces authenticates packets with HMAC-SHA1, which NIST has withdrawn. <xref target="media-baseline"/> requires support for an authenticated encryption transform for that reason, and a replacement for the HMAC-SHA1 transforms is needed from the IETF.</t>

<t>Further security considerations will be added as the profile is reviewed.</t>

</section>
<section anchor="iana-considerations"><name>IANA Considerations</name>

<t>This document has no IANA actions.</t>

</section>
<section anchor="acknowledgments"><name>Acknowledgments</name>

<t>Thanks to Russ Housley for his review of this document.</t>

</section>


  </middle>

  <back>


<references title='References' anchor="sec-combined-references">

    <references title='Normative References' anchor="sec-normative-references">



<reference anchor="RFC2119">
  <front>
    <title>Key words for use in RFCs to Indicate Requirement Levels</title>
    <author fullname="S. Bradner" initials="S." surname="Bradner"/>
    <date month="March" year="1997"/>
    <abstract>
      <t>In many standards track documents several words are used to signify the requirements in the specification. These words are often capitalized. This document defines these words as they should be interpreted in IETF documents. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements.</t>
    </abstract>
  </front>
  <seriesInfo name="BCP" value="14"/>
  <seriesInfo name="RFC" value="2119"/>
  <seriesInfo name="DOI" value="10.17487/RFC2119"/>
</reference>
<reference anchor="RFC8174">
  <front>
    <title>Ambiguity of Uppercase vs Lowercase in RFC 2119 Key Words</title>
    <author fullname="B. Leiba" initials="B." surname="Leiba"/>
    <date month="May" year="2017"/>
    <abstract>
      <t>RFC 2119 specifies common key words that may be used in protocol specifications. This document aims to reduce the ambiguity by clarifying that only UPPERCASE usage of the key words have the defined special meanings.</t>
    </abstract>
  </front>
  <seriesInfo name="BCP" value="14"/>
  <seriesInfo name="RFC" value="8174"/>
  <seriesInfo name="DOI" value="10.17487/RFC8174"/>
</reference>
<reference anchor="RFC3261">
  <front>
    <title>SIP: Session Initiation Protocol</title>
    <author fullname="J. Rosenberg" initials="J." surname="Rosenberg"/>
    <author fullname="H. Schulzrinne" initials="H." surname="Schulzrinne"/>
    <author fullname="G. Camarillo" initials="G." surname="Camarillo"/>
    <author fullname="A. Johnston" initials="A." surname="Johnston"/>
    <author fullname="J. Peterson" initials="J." surname="Peterson"/>
    <author fullname="R. Sparks" initials="R." surname="Sparks"/>
    <author fullname="M. Handley" initials="M." surname="Handley"/>
    <author fullname="E. Schooler" initials="E." surname="Schooler"/>
    <date month="July" year="2002"/>
    <abstract>
      <t>This document describes Session Initiation Protocol (SIP), an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants. These sessions include Internet telephone calls, multimedia distribution, and multimedia conferences. [STANDARDS-TRACK]</t>
    </abstract>
  </front>
  <seriesInfo name="RFC" value="3261"/>
  <seriesInfo name="DOI" value="10.17487/RFC3261"/>
</reference>
<reference anchor="RFC5280">
  <front>
    <title>Internet X.509 Public Key Infrastructure Certificate and Certificate Revocation List (CRL) Profile</title>
    <author fullname="D. Cooper" initials="D." surname="Cooper"/>
    <author fullname="S. Santesson" initials="S." surname="Santesson"/>
    <author fullname="S. Farrell" initials="S." surname="Farrell"/>
    <author fullname="S. Boeyen" initials="S." surname="Boeyen"/>
    <author fullname="R. Housley" initials="R." surname="Housley"/>
    <author fullname="W. Polk" initials="W." surname="Polk"/>
    <date month="May" year="2008"/>
    <abstract>
      <t>This memo profiles the X.509 v3 certificate and X.509 v2 certificate revocation list (CRL) for use in the Internet. An overview of this approach and model is provided as an introduction. The X.509 v3 certificate format is described in detail, with additional information regarding the format and semantics of Internet name forms. Standard certificate extensions are described and two Internet-specific extensions are defined. A set of required certificate extensions is specified. The X.509 v2 CRL format is described in detail along with standard and Internet-specific extensions. An algorithm for X.509 certification path validation is described. An ASN.1 module and examples are provided in the appendices. [STANDARDS-TRACK]</t>
    </abstract>
  </front>
  <seriesInfo name="RFC" value="5280"/>
  <seriesInfo name="DOI" value="10.17487/RFC5280"/>
</reference>
<reference anchor="RFC5922">
  <front>
    <title>Domain Certificates in the Session Initiation Protocol (SIP)</title>
    <author fullname="V. Gurbani" initials="V." surname="Gurbani"/>
    <author fullname="S. Lawrence" initials="S." surname="Lawrence"/>
    <author fullname="A. Jeffrey" initials="A." surname="Jeffrey"/>
    <date month="June" year="2010"/>
    <abstract>
      <t>This document describes how to construct and interpret certain information in a PKIX-compliant (Public Key Infrastructure using X.509) certificate for use in a Session Initiation Protocol (SIP) over Transport Layer Security (TLS) connection. More specifically, this document describes how to encode and extract the identity of a SIP domain in a certificate and how to use that identity for SIP domain authentication. As such, this document is relevant both to implementors of SIP and to issuers of certificates. [STANDARDS-TRACK]</t>
    </abstract>
  </front>
  <seriesInfo name="RFC" value="5922"/>
  <seriesInfo name="DOI" value="10.17487/RFC5922"/>
</reference>
<reference anchor="RFC8226">
  <front>
    <title>Secure Telephone Identity Credentials: Certificates</title>
    <author fullname="J. Peterson" initials="J." surname="Peterson"/>
    <author fullname="S. Turner" initials="S." surname="Turner"/>
    <date month="February" year="2018"/>
    <abstract>
      <t>In order to prevent the impersonation of telephone numbers on the Internet, some kind of credential system needs to exist that cryptographically asserts authority over telephone numbers. This document describes the use of certificates in establishing authority over telephone numbers, as a component of a broader architecture for managing telephone numbers as identities in protocols like SIP.</t>
    </abstract>
  </front>
  <seriesInfo name="RFC" value="8226"/>
  <seriesInfo name="DOI" value="10.17487/RFC8226"/>
</reference>
<reference anchor="RFC9846">
  <front>
    <title>The Transport Layer Security (TLS) Protocol Version 1.3</title>
    <author fullname="E. Rescorla" initials="E." surname="Rescorla"/>
    <date month="July" year="2026"/>
    <abstract>
      <t>This document specifies version 1.3 of the Transport Layer Security (TLS) protocol. TLS allows client/server applications to communicate over the Internet in a way that is designed to prevent eavesdropping, tampering, and message forgery.</t>
      <t>This document obsoletes RFC 8446, which specified TLS 1.3. This document obsoletes RFC 5246 (specifying TLS 1.2) and RFCs 5077, 6961, 7627, and 8422, all of which pertain to TLS 1.2 or earlier, and updates RFCs 5705 and 6066. This document also specifies new requirements for TLS 1.2 implementations.</t>
    </abstract>
  </front>
  <seriesInfo name="RFC" value="9846"/>
  <seriesInfo name="DOI" value="10.17487/RFC9846"/>
</reference>
<reference anchor="RFC9060">
  <front>
    <title>Secure Telephone Identity Revisited (STIR) Certificate Delegation</title>
    <author fullname="J. Peterson" initials="J." surname="Peterson"/>
    <date month="September" year="2021"/>
    <abstract>
      <t>The Secure Telephone Identity Revisited (STIR) certificate profile provides a way to attest authority over telephone numbers and related identifiers for the purpose of preventing telephone number spoofing. This specification details how that authority can be delegated from a parent certificate to a subordinate certificate. This supports a number of use cases, including those where service providers grant credentials to enterprises or other customers capable of signing calls with STIR.</t>
    </abstract>
  </front>
  <seriesInfo name="RFC" value="9060"/>
  <seriesInfo name="DOI" value="10.17487/RFC9060"/>
</reference>
<reference anchor="RFC9325">
  <front>
    <title>Recommendations for Secure Use of Transport Layer Security (TLS) and Datagram Transport Layer Security (DTLS)</title>
    <author fullname="Y. Sheffer" initials="Y." surname="Sheffer"/>
    <author fullname="P. Saint-Andre" initials="P." surname="Saint-Andre"/>
    <author fullname="T. Fossati" initials="T." surname="Fossati"/>
    <date month="November" year="2022"/>
    <abstract>
      <t>Transport Layer Security (TLS) and Datagram Transport Layer Security (DTLS) are used to protect data exchanged over a wide range of application protocols and can also form the basis for secure transport protocols. Over the years, the industry has witnessed several serious attacks on TLS and DTLS, including attacks on the most commonly used cipher suites and their modes of operation. This document provides the latest recommendations for ensuring the security of deployed services that use TLS and DTLS. These recommendations are applicable to the majority of use cases.</t>
      <t>RFC 7525, an earlier version of the TLS recommendations, was published when the industry was transitioning to TLS 1.2. Years later, this transition is largely complete, and TLS 1.3 is widely available. This document updates the guidance given the new environment and obsoletes RFC 7525. In addition, this document updates RFCs 5288 and 6066 in view of recent attacks.</t>
    </abstract>
  </front>
  <seriesInfo name="BCP" value="195"/>
  <seriesInfo name="RFC" value="9325"/>
  <seriesInfo name="DOI" value="10.17487/RFC9325"/>
</reference>
<reference anchor="RFC9525">
  <front>
    <title>Service Identity in TLS</title>
    <author fullname="P. Saint-Andre" initials="P." surname="Saint-Andre"/>
    <author fullname="R. Salz" initials="R." surname="Salz"/>
    <date month="November" year="2023"/>
    <abstract>
      <t>Many application technologies enable secure communication between two entities by means of Transport Layer Security (TLS) with Internet Public Key Infrastructure using X.509 (PKIX) certificates. This document specifies procedures for representing and verifying the identity of application services in such interactions.</t>
      <t>This document obsoletes RFC 6125.</t>
    </abstract>
  </front>
  <seriesInfo name="RFC" value="9525"/>
  <seriesInfo name="DOI" value="10.17487/RFC9525"/>
</reference>

<reference anchor="I-D.wendt-stir-tn-domain-binding">
   <front>
      <title>Binding a Domain Identifier to Telephone Number Authorization in STIR Certificates</title>
      <author fullname="Chris Wendt" initials="C." surname="Wendt">
         <organization>Somos, Inc.</organization>
      </author>
      <date day="1" month="November" year="2026"/>
      <abstract>
	 <t>   This document defines a mechanism for binding a domain identifier to
   telephone number authorization within a STIR certificate.  A
   certificate produced under this mechanism carries, as a co-validated
   pair, the telephone numbers or service provider codes a subject is
   authorized for in a TNAuthList extension and a domain identifying the
   subject in a SubjectAltName dNSName entry.  The binding is
   established at issuance by requiring proof of domain control and
   validation of a TNAuthList authority token within a single
   certificate issuance, such that the resulting certificate attests
   that the responsible authority for the TNAuthList entries and the
   party controlling the domain both authorized the issuance.  The
   mechanism applies to STIR certificates whose TNAuthList contains
   telephone number entries, service provider code entries, or both,
   allowing a domain to be bound to the right-to-use holder for a set of
   numbers or to the provider identified by a service provider code.
   This document defines the issuance conformance requirements and the
   relying party verification rule that together make the binding
   meaningful.  It does not define telephone number or service provider
   code authorization or domain validation, both of which are specified
   elsewhere and referenced here.  It also defines the STI declaration
   record, a DNS record published at the bound domain in which the
   holder of the telephone numbers declares how legitimate calls and
   messages from them are originated and signed, so that a relying party
   can recognize ones that are not, and the requirements that any
   mechanism for discovering the domain bound to a telephone number must
   meet.

	 </t>
      </abstract>
   </front>
   <seriesInfo name="Internet-Draft" value="draft-wendt-stir-tn-domain-binding-02"/>
   
</reference>

<reference anchor="I-D.wendt-sipcore-tls-updates">
   <front>
      <title>TLS 1.3 and Certificate Identity for the Session Initiation Protocol</title>
      <author fullname="Chris Wendt" initials="C." surname="Wendt">
         <organization>Somos, Inc.</organization>
      </author>
      <date day="1" month="November" year="2026"/>
      <abstract>
	 <t>   The specifications that define the use of TLS with the Session
   Initiation Protocol (SIP) predate TLS 1.3 and the current guidance
   for representing and verifying the identity of a service in a
   certificate.  As a result, [RFC3261] states a mandatory cipher suite
   that a TLS 1.3 implementation cannot offer, and the certificate
   identity rules of [RFC5922] are no longer consistent with [RFC9525].
   This document updates both, specifying a TLS version and cipher suite
   baseline for SIP and revising the representation and matching of SIP
   domain identities in certificates.

	 </t>
      </abstract>
   </front>
   <seriesInfo name="Internet-Draft" value="draft-wendt-sipcore-tls-updates-00"/>
   
</reference>



    </references>

    <references title='Informative References' anchor="sec-informative-references">



<reference anchor="RFC3263">
  <front>
    <title>Session Initiation Protocol (SIP): Locating SIP Servers</title>
    <author fullname="J. Rosenberg" initials="J." surname="Rosenberg"/>
    <author fullname="H. Schulzrinne" initials="H." surname="Schulzrinne"/>
    <date month="July" year="2002"/>
    <abstract>
      <t>The Session Initiation Protocol (SIP) uses DNS procedures to allow a client to resolve a SIP Uniform Resource Identifier (URI) into the IP address, port, and transport protocol of the next hop to contact. It also uses DNS to allow a server to send a response to a backup client if the primary client has failed. This document describes those DNS procedures in detail. [STANDARDS-TRACK]</t>
    </abstract>
  </front>
  <seriesInfo name="RFC" value="3263"/>
  <seriesInfo name="DOI" value="10.17487/RFC3263"/>
</reference>
<reference anchor="RFC3711">
  <front>
    <title>The Secure Real-time Transport Protocol (SRTP)</title>
    <author fullname="M. Baugher" initials="M." surname="Baugher"/>
    <author fullname="D. McGrew" initials="D." surname="McGrew"/>
    <author fullname="M. Naslund" initials="M." surname="Naslund"/>
    <author fullname="E. Carrara" initials="E." surname="Carrara"/>
    <author fullname="K. Norrman" initials="K." surname="Norrman"/>
    <date month="March" year="2004"/>
    <abstract>
      <t>This document describes the Secure Real-time Transport Protocol (SRTP), a profile of the Real-time Transport Protocol (RTP), which can provide confidentiality, message authentication, and replay protection to the RTP traffic and to the control traffic for RTP, the Real-time Transport Control Protocol (RTCP). [STANDARDS-TRACK]</t>
    </abstract>
  </front>
  <seriesInfo name="RFC" value="3711"/>
  <seriesInfo name="DOI" value="10.17487/RFC3711"/>
</reference>
<reference anchor="RFC4568">
  <front>
    <title>Session Description Protocol (SDP) Security Descriptions for Media Streams</title>
    <author fullname="F. Andreasen" initials="F." surname="Andreasen"/>
    <author fullname="M. Baugher" initials="M." surname="Baugher"/>
    <author fullname="D. Wing" initials="D." surname="Wing"/>
    <date month="July" year="2006"/>
    <abstract>
      <t>This document defines a Session Description Protocol (SDP) cryptographic attribute for unicast media streams. The attribute describes a cryptographic key and other parameters that serve to configure security for a unicast media stream in either a single message or a roundtrip exchange. The attribute can be used with a variety of SDP media transports, and this document defines how to use it for the Secure Real-time Transport Protocol (SRTP) unicast media streams. The SDP crypto attribute requires the services of a data security protocol to secure the SDP message. [STANDARDS-TRACK]</t>
    </abstract>
  </front>
  <seriesInfo name="RFC" value="4568"/>
  <seriesInfo name="DOI" value="10.17487/RFC4568"/>
</reference>
<reference anchor="RFC5246">
  <front>
    <title>The Transport Layer Security (TLS) Protocol Version 1.2</title>
    <author fullname="T. Dierks" initials="T." surname="Dierks"/>
    <author fullname="E. Rescorla" initials="E." surname="Rescorla"/>
    <date month="August" year="2008"/>
    <abstract>
      <t>This document specifies Version 1.2 of the Transport Layer Security (TLS) protocol. The TLS protocol provides communications security over the Internet. The protocol allows client/server applications to communicate in a way that is designed to prevent eavesdropping, tampering, or message forgery. [STANDARDS-TRACK]</t>
    </abstract>
  </front>
  <seriesInfo name="RFC" value="5246"/>
  <seriesInfo name="DOI" value="10.17487/RFC5246"/>
</reference>
<reference anchor="RFC5630">
  <front>
    <title>The Use of the SIPS URI Scheme in the Session Initiation Protocol (SIP)</title>
    <author fullname="F. Audet" initials="F." surname="Audet"/>
    <date month="October" year="2009"/>
    <abstract>
      <t>This document provides clarifications and guidelines concerning the use of the SIPS URI scheme in the Session Initiation Protocol (SIP). It also makes normative changes to SIP. [STANDARDS-TRACK]</t>
    </abstract>
  </front>
  <seriesInfo name="RFC" value="5630"/>
  <seriesInfo name="DOI" value="10.17487/RFC5630"/>
</reference>
<reference anchor="RFC5763">
  <front>
    <title>Framework for Establishing a Secure Real-time Transport Protocol (SRTP) Security Context Using Datagram Transport Layer Security (DTLS)</title>
    <author fullname="J. Fischl" initials="J." surname="Fischl"/>
    <author fullname="H. Tschofenig" initials="H." surname="Tschofenig"/>
    <author fullname="E. Rescorla" initials="E." surname="Rescorla"/>
    <date month="May" year="2010"/>
    <abstract>
      <t>This document specifies how to use the Session Initiation Protocol (SIP) to establish a Secure Real-time Transport Protocol (SRTP) security context using the Datagram Transport Layer Security (DTLS) protocol. It describes a mechanism of transporting a fingerprint attribute in the Session Description Protocol (SDP) that identifies the key that will be presented during the DTLS handshake. The key exchange travels along the media path as opposed to the signaling path. The SIP Identity mechanism can be used to protect the integrity of the fingerprint attribute from modification by intermediate proxies. [STANDARDS-TRACK]</t>
    </abstract>
  </front>
  <seriesInfo name="RFC" value="5763"/>
  <seriesInfo name="DOI" value="10.17487/RFC5763"/>
</reference>
<reference anchor="RFC5923">
  <front>
    <title>Connection Reuse in the Session Initiation Protocol (SIP)</title>
    <author fullname="V. Gurbani" initials="V." role="editor" surname="Gurbani"/>
    <author fullname="R. Mahy" initials="R." surname="Mahy"/>
    <author fullname="B. Tate" initials="B." surname="Tate"/>
    <date month="June" year="2010"/>
    <abstract>
      <t>This document enables a pair of communicating proxies to reuse a congestion-controlled connection between themselves for sending requests in the forwards and backwards direction. Because the connection is essentially aliased for requests going in the backwards direction, reuse is predicated upon both the communicating endpoints authenticating themselves using X.509 certificates through Transport Layer Security (TLS). For this reason, we only consider connection reuse for TLS over TCP and TLS over Stream Control Transmission Protocol (SCTP). This document also provides guidelines on connection reuse and virtual SIP servers and the interaction of connection reuse and DNS SRV lookups in SIP. [STANDARDS-TRACK]</t>
    </abstract>
  </front>
  <seriesInfo name="RFC" value="5923"/>
  <seriesInfo name="DOI" value="10.17487/RFC5923"/>
</reference>
<reference anchor="RFC5924">
  <front>
    <title>Extended Key Usage (EKU) for Session Initiation Protocol (SIP) X.509 Certificates</title>
    <author fullname="S. Lawrence" initials="S." surname="Lawrence"/>
    <author fullname="V. Gurbani" initials="V." surname="Gurbani"/>
    <date month="June" year="2010"/>
    <abstract>
      <t>This memo documents an extended key usage (EKU) X.509 certificate extension for restricting the applicability of a certificate to use with a Session Initiation Protocol (SIP) service. As such, in addition to providing rules for SIP implementations, this memo also provides guidance to issuers of certificates for use with SIP. This document defines an Experimental Protocol for the Internet community.</t>
    </abstract>
  </front>
  <seriesInfo name="RFC" value="5924"/>
  <seriesInfo name="DOI" value="10.17487/RFC5924"/>
</reference>
<reference anchor="RFC6066">
  <front>
    <title>Transport Layer Security (TLS) Extensions: Extension Definitions</title>
    <author fullname="D. Eastlake 3rd" initials="D." surname="Eastlake 3rd"/>
    <date month="January" year="2011"/>
    <abstract>
      <t>This document provides specifications for existing TLS extensions. It is a companion document for RFC 5246, "The Transport Layer Security (TLS) Protocol Version 1.2". The extensions specified are server_name, max_fragment_length, client_certificate_url, trusted_ca_keys, truncated_hmac, and status_request. [STANDARDS-TRACK]</t>
    </abstract>
  </front>
  <seriesInfo name="RFC" value="6066"/>
  <seriesInfo name="DOI" value="10.17487/RFC6066"/>
</reference>
<reference anchor="RFC2198">
  <front>
    <title>RTP Payload for Redundant Audio Data</title>
    <author fullname="C. Perkins" initials="C." surname="Perkins"/>
    <author fullname="I. Kouvelas" initials="I." surname="Kouvelas"/>
    <author fullname="O. Hodson" initials="O." surname="Hodson"/>
    <author fullname="V. Hardman" initials="V." surname="Hardman"/>
    <author fullname="M. Handley" initials="M." surname="Handley"/>
    <author fullname="J.C. Bolot" initials="J.C." surname="Bolot"/>
    <author fullname="A. Vega-Garcia" initials="A." surname="Vega-Garcia"/>
    <author fullname="S. Fosse-Parisis" initials="S." surname="Fosse-Parisis"/>
    <date month="September" year="1997"/>
    <abstract>
      <t>This document describes a payload format for use with the real-time transport protocol (RTP), version 2, for encoding redundant audio data. [STANDARDS-TRACK]</t>
    </abstract>
  </front>
  <seriesInfo name="RFC" value="2198"/>
  <seriesInfo name="DOI" value="10.17487/RFC2198"/>
</reference>
<reference anchor="RFC4585">
  <front>
    <title>Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)</title>
    <author fullname="J. Ott" initials="J." surname="Ott"/>
    <author fullname="S. Wenger" initials="S." surname="Wenger"/>
    <author fullname="N. Sato" initials="N." surname="Sato"/>
    <author fullname="C. Burmeister" initials="C." surname="Burmeister"/>
    <author fullname="J. Rey" initials="J." surname="Rey"/>
    <date month="July" year="2006"/>
    <abstract>
      <t>Real-time media streams that use RTP are, to some degree, resilient against packet losses. Receivers may use the base mechanisms of the Real-time Transport Control Protocol (RTCP) to report packet reception statistics and thus allow a sender to adapt its transmission behavior in the mid-term. This is the sole means for feedback and feedback-based error repair (besides a few codec-specific mechanisms). This document defines an extension to the Audio-visual Profile (AVP) that enables receivers to provide, statistically, more immediate feedback to the senders and thus allows for short-term adaptation and efficient feedback-based repair mechanisms to be implemented. This early feedback profile (AVPF) maintains the AVP bandwidth constraints for RTCP and preserves scalability to large groups. [STANDARDS-TRACK]</t>
    </abstract>
  </front>
  <seriesInfo name="RFC" value="4585"/>
  <seriesInfo name="DOI" value="10.17487/RFC4585"/>
</reference>
<reference anchor="RFC5124">
  <front>
    <title>Extended Secure RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/SAVPF)</title>
    <author fullname="J. Ott" initials="J." surname="Ott"/>
    <author fullname="E. Carrara" initials="E." surname="Carrara"/>
    <date month="February" year="2008"/>
    <abstract>
      <t>An RTP profile (SAVP) for secure real-time communications and another profile (AVPF) to provide timely feedback from the receivers to a sender are defined in RFC 3711 and RFC 4585, respectively. This memo specifies the combination of both profiles to enable secure RTP communications with feedback. [STANDARDS-TRACK]</t>
    </abstract>
  </front>
  <seriesInfo name="RFC" value="5124"/>
  <seriesInfo name="DOI" value="10.17487/RFC5124"/>
</reference>
<reference anchor="RFC5764">
  <front>
    <title>Datagram Transport Layer Security (DTLS) Extension to Establish Keys for the Secure Real-time Transport Protocol (SRTP)</title>
    <author fullname="D. McGrew" initials="D." surname="McGrew"/>
    <author fullname="E. Rescorla" initials="E." surname="Rescorla"/>
    <date month="May" year="2010"/>
    <abstract>
      <t>This document describes a Datagram Transport Layer Security (DTLS) extension to establish keys for Secure RTP (SRTP) and Secure RTP Control Protocol (SRTCP) flows. DTLS keying happens on the media path, independent of any out-of-band signalling channel present. [STANDARDS-TRACK]</t>
    </abstract>
  </front>
  <seriesInfo name="RFC" value="5764"/>
  <seriesInfo name="DOI" value="10.17487/RFC5764"/>
</reference>
<reference anchor="RFC6188">
  <front>
    <title>The Use of AES-192 and AES-256 in Secure RTP</title>
    <author fullname="D. McGrew" initials="D." surname="McGrew"/>
    <date month="March" year="2011"/>
    <abstract>
      <t>This memo describes the use of the Advanced Encryption Standard (AES) with 192- and 256-bit keys within the Secure RTP (SRTP) protocol. It details counter mode encryption for SRTP and Secure Realtime Transport Control Protocol (SRTCP) and a new SRTP Key Derivation Function (KDF) for AES-192 and AES-256. [STANDARDS-TRACK]</t>
    </abstract>
  </front>
  <seriesInfo name="RFC" value="6188"/>
  <seriesInfo name="DOI" value="10.17487/RFC6188"/>
</reference>
<reference anchor="RFC6716">
  <front>
    <title>Definition of the Opus Audio Codec</title>
    <author fullname="JM. Valin" initials="JM." surname="Valin"/>
    <author fullname="K. Vos" initials="K." surname="Vos"/>
    <author fullname="T. Terriberry" initials="T." surname="Terriberry"/>
    <date month="September" year="2012"/>
    <abstract>
      <t>This document defines the Opus interactive speech and audio codec. Opus is designed to handle a wide range of interactive audio applications, including Voice over IP, videoconferencing, in-game chat, and even live, distributed music performances. It scales from low bitrate narrowband speech at 6 kbit/s to very high quality stereo music at 510 kbit/s. Opus uses both Linear Prediction (LP) and the Modified Discrete Cosine Transform (MDCT) to achieve good compression of both speech and music. [STANDARDS-TRACK]</t>
    </abstract>
  </front>
  <seriesInfo name="RFC" value="6716"/>
  <seriesInfo name="DOI" value="10.17487/RFC6716"/>
</reference>
<reference anchor="RFC7714">
  <front>
    <title>AES-GCM Authenticated Encryption in the Secure Real-time Transport Protocol (SRTP)</title>
    <author fullname="D. McGrew" initials="D." surname="McGrew"/>
    <author fullname="K. Igoe" initials="K." surname="Igoe"/>
    <date month="December" year="2015"/>
    <abstract>
      <t>This document defines how the AES-GCM Authenticated Encryption with Associated Data family of algorithms can be used to provide confidentiality and data authentication in the Secure Real-time Transport Protocol (SRTP).</t>
    </abstract>
  </front>
  <seriesInfo name="RFC" value="7714"/>
  <seriesInfo name="DOI" value="10.17487/RFC7714"/>
</reference>
<reference anchor="RFC7879">
  <front>
    <title>DTLS-SRTP Handling in SIP Back-to-Back User Agents</title>
    <author fullname="R. Ravindranath" initials="R." surname="Ravindranath"/>
    <author fullname="T. Reddy" initials="T." surname="Reddy"/>
    <author fullname="G. Salgueiro" initials="G." surname="Salgueiro"/>
    <author fullname="V. Pascual" initials="V." surname="Pascual"/>
    <author fullname="P. Ravindran" initials="P." surname="Ravindran"/>
    <date month="May" year="2016"/>
    <abstract>
      <t>Session Initiation Protocol (SIP) Back-to-Back User Agents (B2BUAs) exist on the signaling and media paths between the endpoints. This document describes the behavior of B2BUAs when Secure Real-time Transport (SRTP) security context is set up with the Datagram Transport Layer Security (DTLS) protocol.</t>
    </abstract>
  </front>
  <seriesInfo name="RFC" value="7879"/>
  <seriesInfo name="DOI" value="10.17487/RFC7879"/>
</reference>
<reference anchor="RFC8122">
  <front>
    <title>Connection-Oriented Media Transport over the Transport Layer Security (TLS) Protocol in the Session Description Protocol (SDP)</title>
    <author fullname="J. Lennox" initials="J." surname="Lennox"/>
    <author fullname="C. Holmberg" initials="C." surname="Holmberg"/>
    <date month="March" year="2017"/>
    <abstract>
      <t>This document specifies how to establish secure connection-oriented media transport sessions over the Transport Layer Security (TLS) protocol using the Session Description Protocol (SDP). It defines the SDP protocol identifier, 'TCP/TLS'. It also defines the syntax and semantics for an SDP 'fingerprint' attribute that identifies the certificate that will be presented for the TLS session. This mechanism allows media transport over TLS connections to be established securely, so long as the integrity of session descriptions is assured.</t>
      <t>This document obsoletes RFC 4572 by clarifying the usage of multiple fingerprints.</t>
    </abstract>
  </front>
  <seriesInfo name="RFC" value="8122"/>
  <seriesInfo name="DOI" value="10.17487/RFC8122"/>
</reference>
<reference anchor="RFC8627">
  <front>
    <title>RTP Payload Format for Flexible Forward Error Correction (FEC)</title>
    <author fullname="M. Zanaty" initials="M." surname="Zanaty"/>
    <author fullname="V. Singh" initials="V." surname="Singh"/>
    <author fullname="A. Begen" initials="A." surname="Begen"/>
    <author fullname="G. Mandyam" initials="G." surname="Mandyam"/>
    <date month="July" year="2019"/>
    <abstract>
      <t>This document defines new RTP payload formats for the Forward Error Correction (FEC) packets that are generated by the non-interleaved and interleaved parity codes from source media encapsulated in RTP. These parity codes are systematic codes (Flexible FEC, or "FLEX FEC"), where a number of FEC repair packets are generated from a set of source packets from one or more source RTP streams. These FEC repair packets are sent in a redundancy RTP stream separate from the source RTP stream(s) that carries the source packets. RTP source packets that were lost in transmission can be reconstructed using the source and repair packets that were received. The non-interleaved and interleaved parity codes that are defined in this specification offer a good protection against random and bursty packet losses, respectively, at a cost of complexity. The RTP payload formats that are defined in this document address scalability issues experienced with the earlier specifications and offer several improvements. Due to these changes, the new payload formats are not backward compatible with earlier specifications; however, endpoints that do not implement this specification can still work by simply ignoring the FEC repair packets.</t>
    </abstract>
  </front>
  <seriesInfo name="RFC" value="8627"/>
  <seriesInfo name="DOI" value="10.17487/RFC8627"/>
</reference>
<reference anchor="RFC8643">
  <front>
    <title>An Opportunistic Approach for Secure Real-time Transport Protocol (OSRTP)</title>
    <author fullname="A. Johnston" initials="A." surname="Johnston"/>
    <author fullname="B. Aboba" initials="B." surname="Aboba"/>
    <author fullname="A. Hutton" initials="A." surname="Hutton"/>
    <author fullname="R. Jesske" initials="R." surname="Jesske"/>
    <author fullname="T. Stach" initials="T." surname="Stach"/>
    <date month="August" year="2019"/>
    <abstract>
      <t>Opportunistic Secure Real-time Transport Protocol (OSRTP) is an implementation of the Opportunistic Security mechanism, as defined in RFC 7435, applied to the Real-time Transport Protocol (RTP). OSRTP allows encrypted media to be used in environments where support for encryption is not known in advance and is not required. OSRTP does not require Session Description Protocol (SDP) extensions or features and is fully backwards compatible with existing implementations using encrypted and authenticated media and implementations that do not encrypt or authenticate media packets. OSRTP is not specific to any key management technique for Secure RTP (SRTP). OSRTP is a transitional approach useful for migrating existing deployments of real-time communications to a fully encrypted and authenticated state.</t>
    </abstract>
  </front>
  <seriesInfo name="RFC" value="8643"/>
  <seriesInfo name="DOI" value="10.17487/RFC8643"/>
</reference>
<reference anchor="RFC6962">
  <front>
    <title>Certificate Transparency</title>
    <author fullname="B. Laurie" initials="B." surname="Laurie"/>
    <author fullname="A. Langley" initials="A." surname="Langley"/>
    <author fullname="E. Kasper" initials="E." surname="Kasper"/>
    <date month="June" year="2013"/>
    <abstract>
      <t>This document describes an experimental protocol for publicly logging the existence of Transport Layer Security (TLS) certificates as they are issued or observed, in a manner that allows anyone to audit certificate authority (CA) activity and notice the issuance of suspect certificates as well as to audit the certificate logs themselves. The intent is that eventually clients would refuse to honor certificates that do not appear in a log, effectively forcing CAs to add all issued certificates to the logs.</t>
      <t>Logs are network services that implement the protocol operations for submissions and queries that are defined in this document.</t>
    </abstract>
  </front>
  <seriesInfo name="RFC" value="6962"/>
  <seriesInfo name="DOI" value="10.17487/RFC6962"/>
</reference>
<reference anchor="RFC8827">
  <front>
    <title>WebRTC Security Architecture</title>
    <author fullname="E. Rescorla" initials="E." surname="Rescorla"/>
    <date month="January" year="2021"/>
    <abstract>
      <t>This document defines the security architecture for WebRTC, a protocol suite intended for use with real-time applications that can be deployed in browsers -- "real-time communication on the Web".</t>
    </abstract>
  </front>
  <seriesInfo name="RFC" value="8827"/>
  <seriesInfo name="DOI" value="10.17487/RFC8827"/>
</reference>
<reference anchor="RFC8834">
  <front>
    <title>Media Transport and Use of RTP in WebRTC</title>
    <author fullname="C. Perkins" initials="C." surname="Perkins"/>
    <author fullname="M. Westerlund" initials="M." surname="Westerlund"/>
    <author fullname="J. Ott" initials="J." surname="Ott"/>
    <date month="January" year="2021"/>
    <abstract>
      <t>The framework for Web Real-Time Communication (WebRTC) provides support for direct interactive rich communication using audio, video, text, collaboration, games, etc. between two peers' web browsers. This memo describes the media transport aspects of the WebRTC framework. It specifies how the Real-time Transport Protocol (RTP) is used in the WebRTC context and gives requirements for which RTP features, profiles, and extensions need to be supported.</t>
    </abstract>
  </front>
  <seriesInfo name="RFC" value="8834"/>
  <seriesInfo name="DOI" value="10.17487/RFC8834"/>
</reference>
<reference anchor="RFC8862">
  <front>
    <title>Best Practices for Securing RTP Media Signaled with SIP</title>
    <author fullname="J. Peterson" initials="J." surname="Peterson"/>
    <author fullname="R. Barnes" initials="R." surname="Barnes"/>
    <author fullname="R. Housley" initials="R." surname="Housley"/>
    <date month="January" year="2021"/>
    <abstract>
      <t>Although the Session Initiation Protocol (SIP) includes a suite of security services that has been expanded by numerous specifications over the years, there is no single place that explains how to use SIP to establish confidential media sessions. Additionally, existing mechanisms have some feature gaps that need to be identified and resolved in order for them to address the pervasive monitoring threat model. This specification describes best practices for negotiating confidential media with SIP, including a comprehensive protection solution that binds the media layer to SIP layer identities.</t>
    </abstract>
  </front>
  <seriesInfo name="BCP" value="228"/>
  <seriesInfo name="RFC" value="8862"/>
  <seriesInfo name="DOI" value="10.17487/RFC8862"/>
</reference>
<reference anchor="RFC9147">
  <front>
    <title>The Datagram Transport Layer Security (DTLS) Protocol Version 1.3</title>
    <author fullname="E. Rescorla" initials="E." surname="Rescorla"/>
    <author fullname="H. Tschofenig" initials="H." surname="Tschofenig"/>
    <author fullname="N. Modadugu" initials="N." surname="Modadugu"/>
    <date month="April" year="2022"/>
    <abstract>
      <t>This document specifies version 1.3 of the Datagram Transport Layer Security (DTLS) protocol. DTLS 1.3 allows client/server applications to communicate over the Internet in a way that is designed to prevent eavesdropping, tampering, and message forgery.</t>
      <t>The DTLS 1.3 protocol is based on the Transport Layer Security (TLS) 1.3 protocol and provides equivalent security guarantees with the exception of order protection / non-replayability. Datagram semantics of the underlying transport are preserved by the DTLS protocol.</t>
      <t>This document obsoletes RFC 6347.</t>
    </abstract>
  </front>
  <seriesInfo name="RFC" value="9147"/>
  <seriesInfo name="DOI" value="10.17487/RFC9147"/>
</reference>
<reference anchor="RFC6347">
  <front>
    <title>Datagram Transport Layer Security Version 1.2</title>
    <author fullname="E. Rescorla" initials="E." surname="Rescorla"/>
    <author fullname="N. Modadugu" initials="N." surname="Modadugu"/>
    <date month="January" year="2012"/>
    <abstract>
      <t>This document specifies version 1.2 of the Datagram Transport Layer Security (DTLS) protocol. The DTLS protocol provides communications privacy for datagram protocols. The protocol allows client/server applications to communicate in a way that is designed to prevent eavesdropping, tampering, or message forgery. The DTLS protocol is based on the Transport Layer Security (TLS) protocol and provides equivalent security guarantees. Datagram semantics of the underlying transport are preserved by the DTLS protocol. This document updates DTLS 1.0 to work with TLS version 1.2. [STANDARDS-TRACK]</t>
    </abstract>
  </front>
  <seriesInfo name="RFC" value="6347"/>
  <seriesInfo name="DOI" value="10.17487/RFC6347"/>
</reference>
<reference anchor="RFC8842">
  <front>
    <title>Session Description Protocol (SDP) Offer/Answer Considerations for Datagram Transport Layer Security (DTLS) and Transport Layer Security (TLS)</title>
    <author fullname="C. Holmberg" initials="C." surname="Holmberg"/>
    <author fullname="R. Shpount" initials="R." surname="Shpount"/>
    <date month="January" year="2021"/>
    <abstract>
      <t>This document defines the Session Description Protocol (SDP) offer/answer procedures for negotiating and establishing a Datagram Transport Layer Security (DTLS) association. The document also defines the criteria for when a new DTLS association must be established. The document updates RFCs 5763 and 7345 by replacing common SDP offer/answer procedures with a reference to this specification.</t>
      <t>This document defines a new SDP media-level attribute, "tls-id".</t>
      <t>This document also defines how the "tls-id" attribute can be used for negotiating and establishing a Transport Layer Security (TLS) connection, in conjunction with the procedures in RFCs 4145 and 8122.</t>
    </abstract>
  </front>
  <seriesInfo name="RFC" value="8842"/>
  <seriesInfo name="DOI" value="10.17487/RFC8842"/>
</reference>
<reference anchor="RFC8996">
  <front>
    <title>Deprecating TLS 1.0 and TLS 1.1</title>
    <author fullname="K. Moriarty" initials="K." surname="Moriarty"/>
    <author fullname="S. Farrell" initials="S." surname="Farrell"/>
    <date month="March" year="2021"/>
    <abstract>
      <t>This document formally deprecates Transport Layer Security (TLS) versions 1.0 (RFC 2246) and 1.1 (RFC 4346). Accordingly, those documents have been moved to Historic status. These versions lack support for current and recommended cryptographic algorithms and mechanisms, and various government and industry profiles of applications using TLS now mandate avoiding these old TLS versions. TLS version 1.2 became the recommended version for IETF protocols in 2008 (subsequently being obsoleted by TLS version 1.3 in 2018), providing sufficient time to transition away from older versions. Removing support for older versions from implementations reduces the attack surface, reduces opportunity for misconfiguration, and streamlines library and product maintenance.</t>
      <t>This document also deprecates Datagram TLS (DTLS) version 1.0 (RFC 4347) but not DTLS version 1.2, and there is no DTLS version 1.1.</t>
      <t>This document updates many RFCs that normatively refer to TLS version 1.0 or TLS version 1.1, as described herein. This document also updates the best practices for TLS usage in RFC 7525; hence, it is part of BCP 195.</t>
    </abstract>
  </front>
  <seriesInfo name="BCP" value="195"/>
  <seriesInfo name="RFC" value="8996"/>
  <seriesInfo name="DOI" value="10.17487/RFC8996"/>
</reference>
<reference anchor="RFC8659">
  <front>
    <title>DNS Certification Authority Authorization (CAA) Resource Record</title>
    <author fullname="P. Hallam-Baker" initials="P." surname="Hallam-Baker"/>
    <author fullname="R. Stradling" initials="R." surname="Stradling"/>
    <author fullname="J. Hoffman-Andrews" initials="J." surname="Hoffman-Andrews"/>
    <date month="November" year="2019"/>
    <abstract>
      <t>The Certification Authority Authorization (CAA) DNS Resource Record allows a DNS domain name holder to specify one or more Certification Authorities (CAs) authorized to issue certificates for that domain name. CAA Resource Records allow a public CA to implement additional controls to reduce the risk of unintended certificate mis-issue. This document defines the syntax of the CAA record and rules for processing CAA records by CAs.</t>
      <t>This document obsoletes RFC 6844.</t>
    </abstract>
  </front>
  <seriesInfo name="RFC" value="8659"/>
  <seriesInfo name="DOI" value="10.17487/RFC8659"/>
</reference>
<reference anchor="RFC8657">
  <front>
    <title>Certification Authority Authorization (CAA) Record Extensions for Account URI and Automatic Certificate Management Environment (ACME) Method Binding</title>
    <author fullname="H. Landau" initials="H." surname="Landau"/>
    <date month="November" year="2019"/>
    <abstract>
      <t>The Certification Authority Authorization (CAA) DNS record allows a domain to communicate an issuance policy to Certification Authorities (CAs) but only allows a domain to define a policy with CA-level granularity. However, the CAA specification (RFC 8659) also provides facilities for an extension to admit a more granular, CA-specific policy. This specification defines two such parameters: one allowing specific accounts of a CA to be identified by URIs and one allowing specific methods of domain control validation as defined by the Automatic Certificate Management Environment (ACME) protocol to be required.</t>
    </abstract>
  </front>
  <seriesInfo name="RFC" value="8657"/>
  <seriesInfo name="DOI" value="10.17487/RFC8657"/>
</reference>
<reference anchor="RFC9495">
  <front>
    <title>Certification Authority Authorization (CAA) Processing for Email Addresses</title>
    <author fullname="C. Bonnell" initials="C." surname="Bonnell"/>
    <date month="October" year="2023"/>
    <abstract>
      <t>The Certification Authority Authorization (CAA) DNS resource record (RR) provides a mechanism for domains to express the allowed set of Certification Authorities that are authorized to issue certificates for the domain. RFC 8659 contains the core CAA specification, where Property Tags that restrict the issuance of certificates that certify domain names are defined. This specification defines a Property Tag that grants authorization to Certification Authorities to issue certificates that contain the id-kp-emailProtection key purpose in the extendedKeyUsage extension and at least one rfc822Name value or otherName value of type id-on-SmtpUTF8Mailbox that includes the domain name in the subjectAltName extension.</t>
    </abstract>
  </front>
  <seriesInfo name="RFC" value="9495"/>
  <seriesInfo name="DOI" value="10.17487/RFC9495"/>
</reference>

<reference anchor="I-D.ietf-acme-dns-persist">
   <front>
      <title>Automated Certificate Management Environment (ACME) Challenge for Persistent DNS TXT Record Validation</title>
      <author fullname="Shiloh Heurich" initials="S." surname="Heurich">
         <organization>Fastly</organization>
      </author>
      <author fullname="Henry Birge-Lee" initials="H." surname="Birge-Lee">
         <organization>Crosslayer Labs, Inc.</organization>
      </author>
      <author fullname="Michael Slaughter" initials="M." surname="Slaughter">
         <organization>Amazon Trust Services</organization>
      </author>
      <date day="20" month="September" year="2026"/>
      <abstract>
	 <t>   This document specifies &quot;dns-persist-01&quot;, a new validation method for
   the Automated Certificate Management Environment (ACME) protocol.
   This method allows a Certification Authority (CA) to verify control
   over a domain by confirming the presence of a persistent DNS TXT
   record containing CA and account identification information.  This
   method is particularly suited for environments where traditional
   challenge methods are impractical, such as multi-tenant hosting
   platforms, enterprise DNS environments, and IoT deployments.  The
   validation method is designed with a strong focus on security and
   robustness, incorporating widely adopted industry best practices for
   persistent domain control validation.  This design aims to make it
   suitable for Certification Authorities operating under various policy
   environments, including those that align with the CA/Browser Forum
   Baseline Requirements.

	 </t>
      </abstract>
   </front>
   <seriesInfo name="Internet-Draft" value="draft-ietf-acme-dns-persist-02"/>
   
</reference>

<reference anchor="I-D.ietf-stir-certificate-transparency">
   <front>
      <title>STI Certificate Transparency</title>
      <author fullname="Chris Wendt" initials="C." surname="Wendt">
         <organization>Somos, Inc.</organization>
      </author>
      <author fullname="Robert Śliwa" initials="R." surname="Śliwa">
         <organization>Somos, Inc.</organization>
      </author>
      <author fullname="Alec Fenichel" initials="A." surname="Fenichel">
         <organization>TransNexus</organization>
      </author>
      <author fullname="Vinit Anil Gaikwad" initials="V. A." surname="Gaikwad">
         <organization>Twilio</organization>
      </author>
      <date day="21" month="October" year="2026"/>
      <abstract>
	 <t>   This document describes a framework for the use of the Certificate
   Transparency (CT) protocol for publicly logging the existence of
   Secure Telephone Identity (STI) certificates as they are issued or
   observed.  This allows any interested party that is part of the STI
   ecosystem to audit STI certification authority (CA) activity and
   audit both the issuance of suspect certificates and the certificate
   logs themselves.  The intent is to establish a level of trust within
   the STI ecosystem that relies on the verification of telephone
   numbers.  This involves requiring STI certificates to be listed in an
   established log and refusing to honor those that are not.  This
   effectively establishes the precedent that STI CAs must add all
   issued certificates to the logs and thus establishes unique
   association of STI certificates to an authorized provider or assignee
   of a telephone number resource.  In the STI ecosystem, the primary
   role of CT is to provide verifiable trust by detecting the
   unauthorized issuance of duplicate telephone number level delegate
   certificates or provider level certificates.  This provides a robust
   auditable mechanism for the detection of unauthorized creation of
   certificate credentials for illegitimate spoofing of telephone
   numbers or service provider codes (SPC).

   The framework borrows the log structure and API model from RFC6962 to
   enable public auditing and verifiability of certificate issuance.
   While the foundational mechanisms for log operation, Merkle Tree
   construction, and Signed Certificate Timestamps (SCTs) are aligned
   with RFC6962, this document contextualizes their application in the
   STIR ecosystem, focusing on verifiable control over telephone number
   or service provider code resources.

	 </t>
      </abstract>
   </front>
   <seriesInfo name="Internet-Draft" value="draft-ietf-stir-certificate-transparency-05"/>
   
</reference>

<reference anchor="NIST.IR.8547" target="https://csrc.nist.gov/pubs/ir/8547/final">
  <front>
    <title>Transition to Post-Quantum Cryptography Standards</title>
    <author >
      <organization>National Institute of Standards and Technology</organization>
    </author>
    <date year="2026" month="October"/>
  </front>
  <seriesInfo name="NIST" value="IR 8547"/>
</reference>



<reference anchor="I-D.ietf-tls-ecdhe-mlkem">
   <front>
      <title>Post-quantum hybrid ECDHE-MLKEM Key Agreement for TLSv1.3</title>
      <author fullname="Kris Kwiatkowski" initials="K." surname="Kwiatkowski">
         <organization>PQShield</organization>
      </author>
      <author fullname="Panos Kampanakis" initials="P." surname="Kampanakis">
         <organization>AWS</organization>
      </author>
      <author fullname="Bas Westerbaan" initials="B." surname="Westerbaan">
         <organization>Cloudflare</organization>
      </author>
      <author fullname="Douglas Stebila" initials="D." surname="Stebila">
         <organization>University of Waterloo</organization>
      </author>
      <date day="26" month="May" year="2026"/>
      <abstract>
	 <t>   This draft defines three hybrid key agreement mechanisms for TLS 1.3
   - X25519MLKEM768, SecP256r1MLKEM768, and SecP384r1MLKEM1024 - that
   combine the post-quantum ML-KEM (Module-Lattice-Based Key
   Encapsulation Mechanism) with an ECDHE (Elliptic Curve Diffie-
   Hellman) exchange.

	 </t>
      </abstract>
   </front>
   <seriesInfo name="Internet-Draft" value="draft-ietf-tls-ecdhe-mlkem-05"/>
   
</reference>

<reference anchor="I-D.ietf-tls-mlkem">
   <front>
      <title>ML-KEM Post-Quantum Key Agreement for TLS 1.3</title>
      <author fullname="Deirdre Connolly" initials="D." surname="Connolly">
         <organization>Selkie Cryptography</organization>
      </author>
      <date day="16" month="September" year="2026"/>
      <abstract>
	 <t>   This memo defines ML-KEM-512, ML-KEM-768, and ML-KEM-1024 as
   NamedGroups and registers IANA values in the TLS Supported Groups
   registry for use in TLS 1.3 to achieve post-quantum (PQ) key
   establishment.

	 </t>
      </abstract>
   </front>
   <seriesInfo name="Internet-Draft" value="draft-ietf-tls-mlkem-11"/>
   
</reference>

<reference anchor="I-D.ietf-uta-pqc-app">
   <front>
      <title>Post-Quantum Cryptography Recommendations for TLS-based Applications</title>
      <author fullname="Tirumaleswar Reddy.K" initials="T." surname="Reddy.K">
         <organization>Nokia</organization>
      </author>
      <author fullname="Hannes Tschofenig" initials="H." surname="Tschofenig">
         <organization>University of the Bundeswehr Munich</organization>
      </author>
      <date day="4" month="July" year="2026"/>
      <abstract>
	 <t>   Post-quantum cryptography presents new challenges for device
   manufacturers, application developers, and service providers.  This
   document highlights the unique characteristics of applications and
   offers best practices for implementing quantum-ready usage profiles
   in applications that use TLS and supporting protocols such as DNS.

	 </t>
      </abstract>
   </front>
   <seriesInfo name="Internet-Draft" value="draft-ietf-uta-pqc-app-03"/>
   
</reference>

<reference anchor="I-D.ietf-stir-certificates-shortlived">
   <front>
      <title>Short-Lived Certificates for Secure Telephone Identity</title>
      <author fullname="Jon Peterson" initials="J." surname="Peterson">
         <organization>TransUnion</organization>
      </author>
      <date day="6" month="July" year="2026"/>
      <abstract>
	 <t>   When certificates are used as credentials to attest the assignment of
   ownership of telephone numbers, some mechanism is required to provide
   certificate freshness.  This document specifies short-lived
   certificates as a means of guaranteeing certificate freshness for
   secure telephone identity (STIR), potentially relying on the
   Automated Certificate Management Environment (ACME) or similar
   mechanisms to allow signers to acquire certificates as needed.

	 </t>
      </abstract>
   </front>
   <seriesInfo name="Internet-Draft" value="draft-ietf-stir-certificates-shortlived-06"/>
   
</reference>



    </references>

</references>



  </back>

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