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<rfc xmlns:xi="http://www.w3.org/2001/XInclude" category="std"
     number="8827" docName="draft-ietf-rtcweb-security-arch-20"
     ipr="pre5378Trust200902">
     ipr="pre5378Trust200902" obsoletes="" updates="" submissionType="IETF"
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  <!-- xml2rfc v2v3 conversion 2.33.0 -->
  <front>
    <title abbrev="WebRTC Sec. Arch.">WebRTC Security Architecture</title>
    <seriesInfo name="RFC" value="8827"/>
    <author fullname="Eric Rescorla" initials="E.K." initials="E." surname="Rescorla">
      <organization>RTFM, Inc.</organization>
      <address>
        <postal>
          <street>2064 Edgewood Drive</street>
          <city>Palo Alto</city>
          <region>CA</region>
          <code>94303</code>

          <country>USA</country>
          <country>United States of America</country>
        </postal>
        <phone>+1 650 678 2350</phone>
        <email>ekr@rtfm.com</email>
      </address>
    </author>

    <date/>

    <area>ART</area>

    <workgroup>RTCWEB</workgroup>
    <date month="June" year="2020"/>

<!-- [rfced] Please insert any keywords (beyond those that appear in the
title) for use on https://www.rfc-editor.org/search -->

    <abstract>
      <t>
<!-- [rfced] In this cluster, we have been expanding WebRTC in the body of the
document (but not the title) as Web Real-Time Communication.  Do you want to
include this expansion somewhere, or is not needed with the current
explanatory text?

Original (first occurrence):
   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>
        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>
  <middle>
    <section title="Introduction" anchor="sec.introduction"> anchor="sec.introduction" numbered="true" toc="default">
      <name>Introduction</name>
      <t>
        The Real-Time Communications on the Web (RTCWEB) working group Working Group
        standardized protocols for real-time communications between Web
        browsers, generally called "WebRTC" <xref target="I-D.ietf-rtcweb-overview"/>. target="RFC8825" format="default"/>.
        The major use cases for WebRTC technology are real-time audio
        and/or video calls, Web conferencing, and direct data transfer. Unlike
        most conventional real-time systems, systems (e.g., SIP-based <xref
        target="RFC3261"></xref> target="RFC3261" format="default"/> soft phones) phones), WebRTC communications are directly
        controlled by some Web server, via a JavaScript (JS) API as shown in
        <xref target="fig.simple"/>. target="fig.simple" format="default"/>.
      </t>
      <figure title="A simple WebRTC system" anchor="fig.simple">
        <artwork><![CDATA[
        <name>A Simple WebRTC System</name>
        <artwork name="" type="" align="left" alt=""><![CDATA[
                         +----------------+
                         |                |
                         |   Web Server   |
                         |                |
                         +----------------+
                             ^        ^
                            /          \
                    HTTP   /            \   HTTP
                          /              \
                         /                \
                        v                  v
                     JS API              JS API
               +-----------+            +-----------+
               |           |    Media   |           |
               |  Browser  |<---------->|  Browser  |
               |           |            |           |
               +-----------+            +-----------+ ]]></artwork>
      </figure>
      <t>
        A more complicated system might allow for interdomain calling, as shown
        in <xref target="fig.multidomain"/>. target="fig.multidomain" format="default"/>.  The protocol to be used between
        the domains is not standardized by WebRTC, but given the installed base
        and the form of the WebRTC API is likely to be something SDP-based like
        SIP or something like the Extensible Messaging and Presence Protocol (XMPP)
        <xref target="RFC6120"/>. target="RFC6120" format="default"/>.
      </t>
      <figure title="A multidomain WebRTC system" anchor="fig.multidomain">
        <artwork><![CDATA[
        <name>A Multidomain WebRTC System</name>
        <artwork name="" type="" align="left" alt=""><![CDATA[
          +--------------+                +--------------+
          |              | SIP,XMPP,...| SIP, XMPP, ... |              |
          |  Web Server  |<----------->|  |<-------------->|  Web Server  |
          |              |                |              |
          +--------------+                +--------------+
                 ^                                ^
                 |                                |
           HTTP  |                                |  HTTP
                 |                                |
                 v                                v
                 JS API                       JS API
           +-----------+                     +-----------+
           |           |        Media        |           |
           |  Browser  |<---------------->|  |<------------------->|  Browser  |
           |           |                     |           |
           +-----------+                     +-----------+ ]]></artwork>
      </figure>
      <t>
        This system presents a number of new security challenges, which are
        analyzed in <xref target="I-D.ietf-rtcweb-security"/>. target="RFC8826" format="default"/>.  This document
        describes a security architecture for WebRTC which addresses the threats
        and requirements described in that document.
      </t>
    </section>
    <section anchor="sec-term" title="Terminology">
      <t>
        The numbered="true" toc="default">
      <name>Terminology</name>
    <t>The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL
        NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED",
        "MAY", "<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 "OPTIONAL" "<bcp14>OPTIONAL</bcp14>" in this document are
    to be interpreted as described in BCP 14 BCP&nbsp;14 <xref target="RFC2119"/>
    <xref target="RFC8174"/> when, and only when, they appear in all capitals,
    as shown here.
      </t> here.</t>

    </section>
    <section title="Trust Model" anchor="sec.proposal.trusthierarchy"> anchor="sec.proposal.trusthierarchy" numbered="true" toc="default">
      <name>Trust Model</name>
      <t>
        The basic assumption of this architecture is that network resources
        exist in a hierarchy of trust, rooted in the browser, which serves as
        the user's Trusted Computing Base (TCB). Any security property which the
        user wishes to have enforced must be ultimately guaranteed by the
        browser (or transitively by some property the browser
        verifies). Conversely, if the browser is compromised, then no security
        guarantees are possible.  Note that there are cases (e.g., Internet
        kiosks) where the user can't really trust the browser that much. In
        these cases, the level of security provided is limited by how much they
        trust the browser.
      </t>
      <t>
        Optimally, we would not rely on trust in any entities other than the
        browser. However, this is unfortunately not possible if we wish to have
        a functional system.  Other network elements fall into two categories:
        those which can be authenticated by the browser and thus can be granted
        permissions to access sensitive resources, and those which cannot be
        authenticated and thus are untrusted.
      </t>
      <section title="Authenticated Entities" anchor="sec.proposal.authenticated"> anchor="sec.proposal.authenticated" numbered="true" toc="default">
        <name>Authenticated Entities</name>
        <t>
          There are two major classes of authenticated entities in the system:
        </t>
        <t>
          <list style="symbols">
            <t>
              Calling services: Web
        <dl newline="false" spacing="normal">
          <dt>Calling services:</dt>
           <dd>Web sites whose origin we can verify (optimally
              via HTTPS, but in some cases because we are on a topologically
              restricted network, such as behind a firewall, and can infer
              authentication from firewall behavior).
            </t>
            <t>
              Other users: WebRTC behavior).</dd>
           <dt>Other users:</dt>
            <dd>WebRTC peers whose origin we can verify
              cryptographically (optimally via DTLS-SRTP).
            </t>
          </list>
        </t> DTLS-SRTP).</dd>
        </dl>
        <t>
          Note that merely being authenticated does not make these entities
          trusted. For instance, just because we can verify that
          https://www.example.org/
          &lt;https://www.example.org/&gt; is owned by Dr. Evil does not mean that we can
          trust Dr. Evil to access our camera and microphone. However, it gives
          the user an opportunity to determine whether he wishes to trust
          Dr. Evil or not; after all, if he desires to contact Dr. Evil (perhaps
          to arrange for ransom payment), it's safe to temporarily give him
          access to the camera and microphone for the purpose of the call, but
          he doesn't want Dr. Evil to be able to access his camera and
          microphone other than during the call. The point here is that we must
          first identify other elements before we can determine whether and how
          much to trust them. Additionally, sometimes we need to identify the
          communicating peer before we know what policies to apply.
        </t>

      </section>

      <section title="Unauthenticated Entities" anchor="sec.proposal.unauthenticated">
        <t>
          Other than the above entities, we are not generally able to identify
          other network elements, thus we cannot trust them.  This does not mean
          that it is not possible to have any interaction with them, but it
          means that we must assume that they will behave maliciously and design
          a system which is secure even if they do so.
        </t>
      </section>
    </section>
    <!-- Not layered ? -->

    <section title="Overview" anchor="sec.proposal.overview">

<!-- TODO: Federated -->
      <t>
        This section describes a typical WebRTC session [rfced] Sections 3.1 and shows how subsequent:  Per the
        various security elements interact "Gender-Specific
Language" section of <https://www.rfc-editor.org/styleguide/part2/>,
please let us know if we may change these instances of "he," "him,"
and what guarantees are provided "his" to "they," "them," and "their."

Original:
 However, it
 gives the user. The example user an opportunity to determine whether he wishes to trust
 Dr. Evil or not; after all, if he desires to contact Dr. Evil
 (perhaps to arrange for ransom payment), it's safe to temporarily
 give him access to the camera and microphone for the purpose of the
 call, but he doesn't want Dr. Evil to be able to access his camera
 and microphone other than during the call.
...
 The
 idea behind this type of permissions is that a user might have a
 fairly narrow list of peers he is willing to communicate with, e.g.,
 "my mother" rather than "anyone on Facebook".
...
 Note that this
 does not mean that the IdP might not lie, but that is a
 trustworthiness judgement that the user can make at the time he looks
 at the identity.
...
 Note that
 this requires user consent in many cases but because the data channel
 does not need consent, he can use that directly.
...
 Fundamentally, the IdP proxy is just a piece of HTML and JS loaded by
 the browser, so nothing stops a Web attacker from creating their own
 IFRAME, loading the IdP proxy HTML/JS, and requesting a signature
 over his own keys rather than those generated in the browser. -->

        </t>
      </section>
      <section anchor="sec.proposal.unauthenticated" numbered="true" toc="default">
        <name>Unauthenticated Entities</name>
        <t>
          Other than the above entities, we are not generally able to identify
          other network elements; thus, we cannot trust them.  This does not mean
          that it is not possible to have any interaction with them, but it
          means that we must assume that they will behave maliciously and design
          a system which is secure even if they do so.
        </t>
      </section>
    </section>
    <section anchor="sec.proposal.overview" numbered="true" toc="default">
      <name>Overview</name>

<!-- [rfced] Section 4:  We found these comments in the original
approved XML file.  Were these items resolved?

Original:
    </section>
    <!- - Not layered ? - ->

    <section title="Overview" anchor="sec.proposal.overview">
      <!- - TODO: Federated - -> -->

      <t>
        This section describes a typical WebRTC session and shows how the
        various security elements interact and what guarantees are provided to
        the user. The example in this section is a "best case" scenario in which
        we provide the maximal amount of user authentication and media privacy
        with the minimal level of trust in the calling service. Simpler versions
        with lower levels of security are also possible and are noted in the
        text where applicable. It's also important to recognize the tension
        between security (or performance) and privacy. The example shown here is
        aimed towards settings where we are more concerned about secure calling
        than about privacy, but as we shall see, there are settings where one
        might wish to make different tradeoffs--this trade&nbhy;offs -- this architecture is still
        compatible with those settings.
      </t>
      <t>
        For the purposes of this example, we assume the topology shown in the
        figures below. This topology is derived from the topology shown in <xref
        target="fig.simple"/>, target="fig.simple" format="default"/>, but separates Alice Alice's and Bob's identities from the
        process of signaling.  Specifically, Alice and Bob have relationships
        with some Identity Provider (IdP) that supports a protocol (such as
        OpenID Connect) that can be used to demonstrate their identity to
        other parties. For instance, Alice might have an account with a social
        network which she can then use to authenticate to other web Web sites
        without explicitly having an account with those sites; this is a fairly
        conventional pattern on the Web. <xref
        target="sec.trust-relationships"/> &nbsp;<xref target="sec.trust-relationships" format="default"/> provides an overview of Identity
        Providers and the relevant terminology.  Alice and Bob might have
        relationships with different IdPs as well.
      </t>
      <t>
        This separation of identity provision and signaling isn't particularly
        important in "closed world" cases where Alice and Bob are users on the
        same social network and have identities based on that domain (<xref
        target="fig.proposal.idp"/>). target="fig.proposal.idp" format="default"/>). However, there are important settings where
        that is not the case, such as federation (calls from one domain to
        another; see <xref target="fig.proposal-federated.idp"/>) target="fig.proposal-federated.idp" format="default"/>) and calling on
        untrusted sites, such as where two users who have a relationship via a
        given social network want to call each other on another, untrusted,
        site, such as a poker site.
      </t>
      <t>
        Note that the servers themselves are also authenticated by an external
        identity service, the SSL/TLS certificate infrastructure (not shown).
        As is conventional in the Web, all identities are ultimately rooted in
        that system. For instance, when an IdP makes an identity assertion, the
        Relying Party consuming that assertion is able to verify because it is
        able to connect to the IdP via HTTPS.
      </t>
      <figure title="A call with IdP-based identity" anchor="fig.proposal.idp">
        <artwork><![CDATA[
        <name>A Call with IdP-Based Identity</name>
        <artwork name="" type="" align="left" alt=""><![CDATA[
                            +----------------+
                            |                |
                            |     Signaling  |
                            |     Server     |
                            |                |
                            +----------------+
                                ^        ^
                               /          \
                       HTTPS  /            \   HTTPS
                             /              \
                            /                \
                           v                  v
                        JS API              JS API
                  +-----------+            +-----------+
                  |           |    Media   |           |
            Alice |  Browser  |<---------->|  Browser  | Bob
                  |           | (DTLS+SRTP)|           |
                  +-----------+            +-----------+
                        ^      ^--+     +--^     ^
                        |         |     |        |
                        v         |     |        v
                  +-----------+   |     |  +-----------+
                  |           |<--------+  |           |
                  |   IdP1    |   |        |    IdP2   |
                  |           |   +------->|           |
                  +-----------+            +-----------+ ]]></artwork>
      </figure>
      <t>
        <xref target="fig.proposal-federated.idp"/> target="fig.proposal-federated.idp" format="default"/> shows essentially the same
        calling scenario but with a call between two separate domains (i.e., a
        federated case), as in <xref target="fig.multidomain"/>. target="fig.multidomain" format="default"/>. As mentioned
        above, the domains communicate by some unspecified protocol protocol, and
        providing separate signaling and identity allows for calls to be
        authenticated regardless of the details of the inter-domain protocol.
      </t>
      <figure title="A federated call with IdP-based identity" anchor="fig.proposal-federated.idp">
        <artwork><![CDATA[
        <name>A Federated Call with IdP-Based Identity</name>
        <artwork name="" type="" align="left" alt=""><![CDATA[
        +----------------+    Unspecified    +----------------+
        |                |      protocol     |                |
        |    Signaling   |<----------------->|    Signaling   |
        |    Server      |  (SIP, XMPP, ...) |    Server      |
        |                |                   |                |
        +----------------+                   +----------------+
                 ^                                   ^
                 |                                   |
           HTTPS |                                   | HTTPS
                 |                                   |
                 |                                   |
                 v                                   v
              JS API                               JS API
        +-----------+                             +-----------+
        |           |             Media           |           |
  Alice |  Browser  |<--------------------------->|  Browser  | Bob
        |           |           DTLS+SRTP         |           |
        +-----------+                             +-----------+
              ^      ^--+                      +--^     ^
              |         |                      |        |
              v         |                      |        v
        +-----------+   |                      |  +-----------+
        |           |<-------------------------+  |           |
        |   IdP1    |   |                         |    IdP2   |
        |           |   +------------------------>|           |
        +-----------+                             +-----------+ ]]></artwork>
      </figure>
      <section title="Initial Signaling"> numbered="true" toc="default">
        <name>Initial Signaling</name>
        <t>
          For simplicity, assume the topology in <xref
          target="fig.proposal.idp"/>. target="fig.proposal.idp" format="default"/>.  Alice and Bob are both users of a common
          calling service; they both have approved the calling service to make
          calls (we defer the discussion of device access permissions until
          later).  They are both connected to the calling service via HTTPS and
          so know the origin with some level of confidence. They also have
          accounts with some identity provider.  This sort of identity service
          is becoming increasingly common in the Web environment (with technologies
          such as Federated Google Login, Facebook Connect, OAuth,
          OpenID, WebFinger), and is often provided as a side effect service of
          a user's ordinary accounts with some service. In this example, we show
          Alice and Bob using a separate identity service, though the identity
          service may be the same entity as the calling service or there may be
          no identity service at all.
        </t>
        <t>
          Alice is logged onto the calling service and decides to call Bob.  She &nbsp;She
          can see from the calling service that he is online and the calling
          service presents a JS UI in the form of a button next to Bob's name
          which says "Call". Alice clicks the button, which initiates a JS
          callback that instantiates a PeerConnection object. This does not
          require a security check: JS from any origin is allowed to get this
          far.
        </t>
        <t>
          Once the PeerConnection is created, the calling service JS needs to
          set up some media. Because this is an audio/video call, it creates a
          MediaStream with two MediaStreamTracks, one connected to an audio
          input and one connected to a video input. At this point point, the first
          security check is required: untrusted origins are not allowed to
          access the camera and microphone, so the browser prompts Alice for
          permission.
        </t>
        <t>
          In the current W3C API, once some streams have been added, Alice's
          browser + JS generates a signaling message <xref
          target="I-D.ietf-rtcweb-jsep"/> target="RFC8829" format="default"/> containing:
        </t>
        <t>
          <list style="symbols">
            <t>
        <ul spacing="normal">
          <li>
              Media channel information
            </t>
            <t>
            </li>
          <li>
              Interactive Connectivity Establishment (ICE) <xref
              target="RFC8445"/> target="RFC8445" format="default"/> candidates
            </t>
            <t>
            </li>
          <li>
              A fingerprint "fingerprint" attribute binding the communication to a key pair
              <xref target="RFC5763"/>. target="RFC5763" format="default"/>. Note that this key may simply be
              ephemerally generated for this call or specific to this domain,
              and Alice may have a large number of such keys.
            </t>
          </list>
        </t>
            </li>
        </ul>
        <t>
          Prior to sending out the signaling message, the PeerConnection code
          contacts the identity service and obtains an assertion binding Alice's
          identity to her fingerprint. The exact details depend on the identity
          service (though as discussed in <xref target="sec.generic.idp"/> target="sec.generic.idp" format="default"/>
          PeerConnection can be agnostic to them), but for now it's easiest to
          think of as an OAuth token.  The assertion may bind other
          information to the identity besides the fingerprint, but at minimum it
          needs to bind the fingerprint.
        </t>
        <t>
          This message is sent to the signaling server, e.g., by XMLHttpRequest
          <xref target="XmlHttpRequest"/> target="XmlHttpRequest" format="default"/> or by WebSockets
	  <xref
          target="RFC6455"/>, target="RFC6455" format="default"/>, over TLS <xref target="RFC5246"/>.
	  target="RFC5246" format="default"/>.

<!-- [rfced] Section 4.1:  Because RFC 5246 has been obsoleted by
RFC 8446, would you like to (1) cite and list RFC 8446 instead,
(2) list both documents, or (3) leave the obsolete citation in place
(i.e., no changes)?

Original:
 This message is sent to the signaling server, e.g., by XMLHttpRequest
 [XmlHttpRequest] or by WebSockets [RFC6455], over TLS [RFC5246]. -->

          The signaling server processes the message from Alice's browser,
          determines that this is a call to Bob Bob, and sends a signaling message to
          Bob's browser (again, the format is currently undefined).  The JS on
          Bob's browser processes it, and alerts Bob to the incoming call and to
          Alice's identity. In this case, Alice has provided an identity
          assertion and so Bob's browser contacts Alice's identity provider
          (again, this is done in a generic way so the browser has no specific
          knowledge of the IdP) to verify the assertion. It is also possible
          to have IdPs with which the browser has a specific trustrelationship, trust relationship,
          as described in <xref target="sec.trust-relationships"/>. target="sec.trust-relationships" format="default"/>.
          This allows the browser
          to display a trusted element in the browser chrome indicating that a
          call is coming in from Alice. If Alice is in Bob's address book, then
          this interface might also include her real name, a picture, etc.  The
          calling site will also provide some user interface element (e.g., a
          button) to allow Bob to answer the call, though this is most likely
          not part of the trusted UI.
        </t>
        <t>
          If Bob agrees agrees, a PeerConnection is instantiated with the message from
          Alice's side.  Then, a similar process occurs as on Alice's browser:
          Bob's browser prompts him for device permission, the media streams are
          created, and a return signaling message containing media information,
          ICE candidates, and a fingerprint is sent back to Alice via the
          signaling service.  If Bob has a relationship with an IdP, the message
          will also come with an identity assertion.
        </t>
        <t>
          At this point, Alice and Bob each know that the other party wants to
          have a secure call with them. Based purely on the interface provided
          by the signaling server, they know that the signaling server claims
          that the call is from Alice to Bob. This &nbsp;This level of security is provided
          merely by having the fingerprint in the message and having that
          message received securely from the signaling server.  Because the far
          end sent an identity assertion along with their message, they know
          that this is verifiable from the IdP as well. Note that if the call is
          federated, as shown in <xref target="fig.proposal-federated.idp"/> target="fig.proposal-federated.idp" format="default"/>,
          then Alice is able to verify Bob's identity in a way that is not
          mediated by either her signaling server or Bob's. Rather, she verifies
          it directly with Bob's IdP.
        </t>
        <t>
          Of course, the call works perfectly well if either Alice or Bob
          doesn't have a relationship with an IdP; they just get a lower level
          of assurance. I.e., That is, they simply have whatever information their
          calling site claims about the caller/callee's identity.  Moreover,
          Alice might wish to make an anonymous call through an anonymous
          calling site, in which case she would of course just not provide any
          identity assertion and the calling site would mask her identity from
          Bob.
        </t>
      </section>
      <section title="Media numbered="true" toc="default">
        <name>Media Consent Verification"> Verification</name>
        <t>
          As described in (<xref target="I-D.ietf-rtcweb-security"/>; Section
          4.2) <xref target="RFC8826" sectionFormat="comma"
	  section="4.2"/>, media consent verification is provided via ICE.
  Thus, Alice and
          Bob perform ICE checks with each other.  At the completion of these
          checks, they are ready to send non-ICE data.
        </t>
        <t>
          At this point, Alice knows that (a) Bob (assuming he is verified via
          his IdP) or someone else who the signaling service is claiming is Bob
          is willing to exchange traffic with her and (b) that either Bob is at
          the IP address which she has verified via ICE or there is an attacker
          who is on-path to that IP address detouring the traffic. Note that it
          is not possible for an attacker who is on-path between Alice and Bob
          but not attached to the signaling service to spoof these checks
          because they do not have the ICE credentials. Bob has the same
          security guarantees with respect to Alice.
        </t>
      </section>
      <section title="DTLS Handshake"> numbered="true" toc="default">
        <name>DTLS Handshake</name>
        <t>
          Once the requisite ICE checks have completed, Alice and Bob can set
          up a secure channel or channels. This is performed via DTLS <xref target="RFC6347"/> target="RFC6347" format="default"/>
          and DTLS-SRTP <xref target="RFC5763"/> target="RFC5763" format="default"/> keying for SRTP
          <xref target="RFC3711"/> target="RFC3711" format="default"/> for the media channel and SCTP
	  the Stream Control Transmission Protocol (SCTP) over DTLS
          <xref target="RFC8261"/> target="RFC8261" format="default"/> for data
          channels. Specifically, Alice and Bob perform a DTLS handshake on
          every component which has been established by ICE. The total number of
          channels depends on the amount of muxing; in the most likely case case, we
          are using both RTP/RTCP mux and muxing multiple media streams on the
          same channel, in which case there is only one DTLS handshake. Once the
          DTLS handshake has completed, the keys are exported <xref
          target="RFC5705"/> target="RFC5705" format="default"/> and used to key SRTP for the media channels.
        </t>
        <t>
          At this point, Alice and Bob know that they share a set of secure data
          and/or media channels with keys which are not known to any third-party
          attacker. If Alice and Bob authenticated via their IdPs, then they
          also know that the signaling service is not mounting a
          man-in-the-middle attack on their traffic. Even if they do not use an
          IdP, as long as they have minimal trust in the signaling service not
          to perform a man-in-the-middle attack, they know that their
          communications are secure against the signaling service as well (i.e.,
          that the signaling service cannot mount a passive attack on the
          communications).
        </t>
      </section>
      <section title="Communications numbered="true" toc="default">
        <name>Communications and Consent Freshness"> Freshness</name>
        <t>
          From a security perspective, everything from here on in is a little
          anticlimactic: Alice and Bob exchange data protected by the keys
          negotiated by DTLS. Because of the security guarantees discussed in
          the previous sections, they know that the communications are encrypted
          and authenticated.
        </t>
        <t>
          The one remaining security property we need to establish is "consent
          freshness", i.e., allowing Alice to verify that Bob is still prepared
          to receive her communications so that Alice does not continue to send
          large traffic volumes to entities which went abruptly offline. ICE
          specifies periodic STUN Session Traversal Utilities for NAT (STUN) keepalives but only if media is not flowing.
          Because the consent issue is more difficult here, we require WebRTC
          implementations to periodically send keepalives.  As described in
          Section 5.3, these keepalives MUST <bcp14>MUST</bcp14> be based on the consent freshness
          mechanism specified in <xref target="RFC7675"/>. target="RFC7675" format="default"/>.

<!-- [rfced] Section 4.4:  This document does not have a Section 5.3.
Please let us know which section should be cited here.

Original:
 As described in
 Section 5.3, these keepalives MUST be based on the consent freshness
 mechanism specified in [RFC7675]. -->

  If a
          keepalive fails and no new ICE channels can be established, then the
          session is terminated.
        </t>
      </section>
    </section>
    <section title="SDP anchor="sec.sdp-id-attr" numbered="true" toc="default">
      <name>SDP Identity Attribute" anchor="sec.sdp-id-attr"> Attribute</name>
      <t>
        The SDP 'identity' "identity" attribute is a session-level attribute that
        is used by an endpoint to convey its identity assertion to its
        peer. The identity assertion identity-assertion value is encoded as Base-64, base64, as described
        in Section 4 of <xref target="RFC4648"/>. target="RFC4648" sectionFormat="of" section="4"/>.
      </t>
      <t>
        The procedures in this section are based on the assumption
        that the identity assertion of an endpoint is bound to identity assertion of an endpoint is bound to the
        fingerprints of the endpoint. This does not preclude the definition of
        alternative means of binding an assertion to the endpoint, but such
        means are outside the scope of this specification.
      </t>
      <t>
        The semantics of multiple "identity" attributes within an
        offer or answer are undefined.  Implementations <bcp14>SHOULD</bcp14> only include a
        single "identity" attribute in an offer or answer, and relying parties
        <bcp14>MAY</bcp14> elect to ignore all but the first "identity" attribute.
      </t>
      <dl newline="false" spacing="normal">
        <dt>Name:</dt>
        <dd>identity</dd>
        <dt>Value:</dt>
        <dd>identity-assertion</dd>
        <dt>Usage Level:</dt>
        <dd>session</dd>
        <dt>Charset Dependent:</dt>
        <dd>no</dd>
        <dt>Default Value:</dt>
        <dd>N/A</dd>
        <dt>Name:</dt>
        <dd>identity</dd>
      </dl>

<!-- [rfced] Section 5:  Are both "Name:  identity" entries needed in
this list?

Original:
 Name:  identity

 Value:  identity-assertion

 Usage Level:  session

 Charset Dependent:  no

 Default Value:  N/A

 Name:  identity -->

<t>Syntax:</t>
      <sourcecode name="abnf-1" type="abnf" ><![CDATA[
 identity-assertion       = identity-assertion-value
                            *(SP identity-extension)
 identity-assertion-value = base64
 identity-extension       = extension-name [ "=" extension-value ]
 extension-name           = token
 extension-value          = 1*(%x01-09 / %x0b-0c / %x0e-3a / %x3c-ff)
                            ; byte-string from [RFC4566]

 <ALPHA and DIGIT as defined in [RFC4566]>
 <base64 as defined in [RFC4566]>
]]></sourcecode>

<t>Example:</t>
<!-- [rfced] Section 5: We have split the <artwork> into 2 pieces: the
        fingerprints of
first has been tagged as <sourcecode type="abnf"> and the endpoint. This does not preclude second as
<sourcecode type="sdp" >. See
<https://www.rfc-editor.org/materials/sourcecode-types.txt> for the definition of
        alternative means preferred
list of binding an assertion to the endpoint, but such
        means "type" attributes.  Please review and let us know if ay updates are outside
needed.

For the scope definitions of ALPHA and DIGIT, RFC 4566 refers to RFC 4234, which has
been obsoleted by RFC 5234.  Should this specification.
      </t>
      <t>
        The semantics of multiple 'identity' attributes within an
        offer or answer are undefined.  Implementations SHOULD only include a
        single 'identity' attribute in an offer or answer document reference RFC 5234 for ALPHA
and relying parties
        MAY elect DIGIT?   Also, RFC 4566 will soon be obsoleted by RFC-to-be 8866
<draft-ietf-mmusic-rfc4566bis-37>; should this document be updated to ignore all but the first 'identity' attribute.
      </t>
      <t>
        <list style="hanging">
        <t hangText="Name:">identity</t>
        <t hangText="Value:">identity-assertion</t>
        <t hangText="Usage Level:">session</t>
        <t hangText="Charset Dependent:">no</t>
        <t hangText="Default Value:">N/A</t>
        <t hangText="Name:">identity</t>
        </list>
      </t>
      <figure> point to
RFC 8866?

Original:

      <artwork type="inline"><![CDATA[
Syntax:

  identity-assertion        = identity-assertion-value
                              *(SP identity-extension)
  identity-assertion-value  = base64
  identity-extension        = extension-name [ "=" extension-value ]
  extension-name            = token
  extension-value           = 1*(%x01-09 / %x0b-0c / %x0e-3a / %x3c-ff)
                              ; byte-string from [RFC4566]

  <ALPHA and DIGIT as defined in [RFC4566]>
  <base64 as defined in [RFC4566]>

 Example:

  a=identity:\
    eyJpZHAiOnsiZG9tYWluIjoiZXhhbXBsZS5vcmciLCJwcm90b2NvbCI6ImJvZ3Vz\
    In0sImFzc2VydGlvbiI6IntcImlkZW50aXR5XCI6XCJib2JAZXhhbXBsZS5vcmdc\
    IixcImNvbnRlbnRzXCI6XCJhYmNkZWZnaGlqa2xtbm9wcXJzdHV2d3l6XCIsXCJz\
    aWduYXR1cmVcIjpcIjAxMDIwMzA0MDUwNlwifSJ9

  Note
-->

<sourcecode name="sdp-1" type="sdp" ><![CDATA[
 a=identity:\
   eyJpZHAiOnsiZG9tYWluIjoiZXhhbXBsZS5vcmciLCJwcm90b2NvbCI6ImJvZ3Vz\
   In0sImFzc2VydGlvbiI6IntcImlkZW50aXR5XCI6XCJib2JAZXhhbXBsZS5vcmdc\
   IixcImNvbnRlbnRzXCI6XCJhYmNkZWZnaGlqa2xtbm9wcXJzdHV2d3l6XCIsXCJz\
   aWduYXR1cmVcIjpcIjAxMDIwMzA0MDUwNlwifSJ9 ]]></sourcecode>

  <aside><t>Note that long lines in the example are folded to meet the column
  width constraints of this document; the backslash ("\") at the end of
  a line, the carriage return that follows, and whitespace shall be ignored.

      ]]></artwork>
       </figure> ignored.</t></aside>
      <t>
         This specification does not define any extensions for the attribute.
      </t>
      <t>
         The identity-assertion value is a JSON <xref target="RFC8259"/> encoded string. string
	 <xref target="RFC8259" format="default"/>. The JSON object
         contains two keys: "assertion" and "idp". The <spanx style="verb">assertion</spanx> "assertion" key value contains
         an opaque string that is consumed by the IdP. The <spanx style="verb">idp</spanx> "idp" key value contains a
         dictionary with one or two further values that identify the IdP. See
         <xref target="sec.request-assert"/> target="sec.request-assert" format="default"/> for more details.
      </t>
      <section title="Offer/Answer Considerations" anchor="sec.sdp-id-attr-oa"> anchor="sec.sdp-id-attr-oa" numbered="true" toc="default">
        <name>Offer/Answer Considerations</name>

        <t>
           This section defines the SDP Offer/Answer offer/answer <xref target="RFC3264"/> target="RFC3264" format="default"/> considerations for the SDP
           'identity'
           "identity" attribute.
        </t>
        <t>
           Within this section, 'initial offer' refers to the first offer in the
           SDP session that contains an SDP <spanx style="verb">identity</spanx> "identity" attribute.
        </t>
        <section title="Generating anchor="sec.sdp-id-attr-oa-inio" numbered="true" toc="default">
          <name>Generating the Initial SDP Offer" anchor="sec.sdp-id-attr-oa-inio"> Offer</name>
          <t>
           When an offerer sends an offer, in order to provide its
           identity assertion to the peer, it includes an 'identity' "identity" attribute in
           the offer. In addition, the offerer includes one or more SDP
           'fingerprint'
           "fingerprint" attributes.  The 'identity' "identity" attribute MUST <bcp14>MUST</bcp14> be bound to
           all the 'fingerprint' "fingerprint" attributes in the session
           description.
          </t>
        </section>
        <section title="Generating of anchor="sec.sdp-id-attr-oa-ansa" numbered="true" toc="default">
          <name>Generating an SDP Answer" anchor="sec.sdp-id-attr-oa-ansa"> Answer</name>
          <t>
             If the answerer elects to include an 'identity' "identity" attribute, it follows
             the same steps as those in <xref target="sec.sdp-id-attr-oa-inio"/>. target="sec.sdp-id-attr-oa-inio" format="default"/>.
             The answerer can choose to include or omit an 'identity' "identity" attribute independently,
             regardless of whether the offerer did so.
          </t>
        </section>
        <section title="Processing anchor="sec.sdp-id-attr-oa-offa" numbered="true" toc="default">
          <name>Processing an SDP Offer or Answer" anchor="sec.sdp-id-attr-oa-offa"> Answer</name>
          <t>
             When an endpoint receives an offer or answer that contains an 'identity' "identity"
             attribute, the answerer can use the the attribute information to
             contact the IdP and verify the identity of the peer. If the identity
             requires a third-party IdP as described in <xref target="sec.trust-relationships"/> target="sec.trust-relationships" format="default"/>,
             then that IdP will need to have been specifically configured.
             If the identity verification fails, the answerer MUST <bcp14>MUST</bcp14> discard the
             offer or answer as malformed.
          </t>
        </section>
        <section title="Modifying anchor="sec.sdp-id-attr-oa-modi" numbered="true" toc="default">
          <name>Modifying the Session" anchor="sec.sdp-id-attr-oa-modi"> Session</name>
          <t>
             When modifying a session, if the set of fingerprints is
             unchanged, then the sender MAY <bcp14>MAY</bcp14> send the same 'identity' "identity" attribute. In
             this case, the established identity MUST <bcp14>MUST</bcp14> be applied to existing DTLS
             connections as well as new connections established using one of those
             fingerprints. Note that <xref target="I-D.ietf-rtcweb-jsep"/>, Section
             5.2.1 target="RFC8829" sectionFormat="comma" section="5.2.1"/> requires that each media section use the same set of
             fingerprints for every media section.
             If a new identity "identity" attribute is received, then the receiver MUST <bcp14>MUST</bcp14>
             apply that identity to all existing connections.
          </t>
          <t>
             If the set of fingerprints changes, then the sender MUST <bcp14>MUST</bcp14>
             either send a new 'identity' "identity" attribute or none at all.
             Because a change in fingerprints also causes a new DTLS
             connection to be established, the receiver MUST <bcp14>MUST</bcp14> discard
             all previously established identities.
          </t>
        </section>
      </section>
    </section>
    <section title="Detailed anchor="sec.proposal.detailed" numbered="true" toc="default">
      <name>Detailed Technical Description" anchor="sec.proposal.detailed"> Description</name>
      <section title="Origin anchor="sec.proposal.origin" numbered="true" toc="default">
        <name>Origin and Web Security Issues" anchor="sec.proposal.origin"> Issues</name>
        <t>
          The basic unit of permissions for WebRTC is the origin <xref
          target="RFC6454"/>. target="RFC6454" format="default"/>. Because the security of the origin depends on
          being able to authenticate content from that origin, the origin can
          only be securely established if data is transferred over HTTPS <xref
          target="RFC2818"/>. target="RFC2818" format="default"/>. Thus, clients MUST <bcp14>MUST</bcp14> treat HTTP and HTTPS origins as
          different permissions domains. Note: this This follows directly from the
          origin security model and is stated here merely for clarity.
        </t>
        <t>
          Many web Web browsers currently forbid by default any active mixed content
          on HTTPS pages. That is, when JavaScript is loaded from an HTTP origin
          onto an HTTPS page, an error is displayed and the HTTP content is not
          executed unless the user overrides the error. Any browser which
          enforces such a policy will also not permit access to WebRTC
          functionality from mixed content pages (because they never display
          mixed content).  Browsers which allow active mixed content MUST <bcp14>MUST</bcp14>
          nevertheless disable WebRTC functionality in mixed content settings.
        </t>
        <t>
          Note that it is possible for a page which that was not mixed content to
          become mixed content during the duration of the call.  The major risk
          here is that the newly arrived insecure JS might redirect media to a
          location controlled by the attacker.  Implementations MUST <bcp14>MUST</bcp14> either
          choose to terminate the call or display a warning at that point.
        </t>
        <t>
          Also note that the security architecture depends on the keying material
          not being available to move between origins.  But,  But it is assumed that
          the identity assertion can be passed to anyone that the page cares to.
        </t>
      </section>
      <section title="Device anchor="sec.proposal.device.permissions" numbered="true" toc="default">
        <name>Device Permissions Model" anchor="sec.proposal.device.permissions"> Model</name>
        <t>
          Implementations MUST <bcp14>MUST</bcp14> obtain explicit user consent prior to providing
          access to the camera and/or microphone. Implementations MUST <bcp14>MUST</bcp14> at
          minimum support the following two permissions models for HTTPS
          origins.
        </t>
        <t>
          <list style="symbols">
            <t>
        <ul spacing="normal">
          <li>
              Requests for one-time camera/microphone access.
            </t>
            <t>
            </li>
          <li>
              Requests for permanent access.
            </t>
          </list>
        </t>
            </li>
        </ul>
        <t>
          Because HTTP origins cannot be securely established against network
          attackers, implementations MUST <bcp14>MUST</bcp14> refuse all permissions grants for
          HTTP origins.
        </t>
        <t>
          In addition, they SHOULD <bcp14>SHOULD</bcp14> support requests for access that promise that
          media from this grant will be sent to a single communicating peer
          (obviously there could be other requests for other peers), eE.g., e.g.,
          "Call customerservice@example.org".  The semantics of this request are
          that the media stream from the camera and microphone will only be
          routed through a connection which has been cryptographically verified
          (through the IdP mechanism or an X.509 certificate in the DTLS-SRTP
          handshake) as being associated with the stated identity. Note that it
          is unlikely that browsers would have X.509 certificates, but servers
          might.  Browsers servicing such requests SHOULD <bcp14>SHOULD</bcp14> clearly indicate that
          identity to the user when asking for permission.  The idea behind this
          type of permissions is that a user might have a fairly narrow list of
          peers he is willing to communicate with, e.g., "my mother" rather than
          "anyone on Facebook". Narrow permissions grants allow the browser to
          do that enforcement.
        </t>

        <t>
          <list style="hanging">
            <t hangText="API Requirement:">
        <dl newline="false" spacing="normal">
          <dt>API Requirement:</dt>
          <dd>
              The API MUST <bcp14>MUST</bcp14> provide a mechanism for the requesting JS to
              relinquish the ability to see or modify the media (e.g., via
              MediaStream.record()).  Combined with secure authentication of the
              communicating peer, this allows a user to be sure that the calling
              site is not accessing or modifying their conversion.
            </t>
          </list>
        </t>

        <t>
          <list style="hanging">
            <t hangText="UI Requirement:">
            </dd>
        </dl>
        <dl newline="false" spacing="normal">
          <dt>UI Requirement:</dt>
          <dd>
              The UI MUST <bcp14>MUST</bcp14> clearly indicate when the user's camera and microphone
              are in use.  This indication MUST NOT <bcp14>MUST NOT</bcp14> be suppressable suppressible by the JS
              and MUST clearly indicate how JS
              and <bcp14>MUST</bcp14> clearly indicate how to terminate device access, and
              provide a UI means to immediately stop camera/microphone input
              without the JS being able to prevent it.
            </dd>
        </dl>
        <dl newline="false" spacing="normal">
          <dt>UI Requirement:</dt>
          <dd>
              If the UI indication of camera/microphone use is displayed in the
              browser such that minimizing the browser window would hide the
              indication, or the JS creating an overlapping window would hide
              the indication, then the browser <bcp14>SHOULD</bcp14> stop camera and microphone
              input when the indication is hidden.  (Note: This may not be
              necessary in systems that are non-windows-based but that have good
              notifications support, such as phones.)
            </dd>
        </dl>

<!-- [rfced] Section 6.2:  Is the bullet list after this "UI
Requirement:" list item supposed to terminate device access, and
              provide be a UI means to immediately stop camera/microphone input
              without "sub-list" (as was done,
for example, after the JS being able to prevent it.
            </t>
          </list>
        </t>

        <t>
          <list style="hanging">
            <t hangText="UI Requirement:"> "UI Requirements:" list item in Section 6.5),
or should it remain as a separate list?

Original:
 UI Requirement:  If the UI indication of camera/microphone use are
    displayed in the browser such that minimizing the browser window
    would hide the indication, or the JS creating an overlapping
    window would hide the indication, then the browser SHOULD stop
    camera and microphone input when the indication is hidden.  [Note:
    this may not be necessary in systems that are non-windows-based
    but that have good notifications support, such as phones.]
            </t>
          </list>
        </t>

        <t>
          <list style="symbols">
            <t>

 o  Browsers MUST NOT permit permanent screen or application sharing
    permissions to be installed as a response to a JS request for
    permissions.  Instead, they must require some other user action
    such as a permissions setting or an application install experience
    to grant permission to a site.
            </t>
            <t>
... -->

        <ul spacing="normal">
          <li>
              Browsers MUST <bcp14>MUST NOT</bcp14> permit permanent screen or application sharing
              permissions to be installed as a response to a JS request for
              permissions. Instead, they must require some other user action
              such as a permissions setting or an application install experience
              to grant permission to a site.
            </li>
          <li>
              Browsers <bcp14>MUST</bcp14> provide a separate dialog request for
              screen/application sharing permissions even if the media request
              is made at the same time as camera and microphone.
            </t>

            <t>

<!-- [rfced] Section 6.2:  Please clarify the meaning of "as camera
and microphone."

Original:
 o  Browsers MUST provide a separate dialog request for screen/
    application sharing permissions even if the media request is made
    at the same time as camera and microphone. -->

            </li>
          <li>
              The browser MUST <bcp14>MUST</bcp14> indicate any windows which are currently being
              shared in some unambiguous way. Windows which are not visible MUST
              NOT <bcp14>MUST
              NOT</bcp14> be shared even if the application is being shared.  If the
              screen is being shared, then that MUST <bcp14>MUST</bcp14> be indicated.
            </t>
          </list>
        </t>
            </li>
        </ul>
        <t>
          Browsers MAY <bcp14>MAY</bcp14> permit the formation of data channels without any direct
          user approval. Because sites can always tunnel data through the
          server, further restrictions on the data channel do not provide any
          additional security.  (See <xref
          target="sec.proposal.communications.consent"/> target="sec.proposal.communications.consent" format="default"/> for a related issue). issue.)
        </t>
        <t>
          Implementations which support some form of direct user authentication
          SHOULD
          <bcp14>SHOULD</bcp14> also provide a policy by which a user can authorize calls only
          to specific communicating peers. Specifically, the implementation
          SHOULD
          <bcp14>SHOULD</bcp14> provide the following interfaces/controls:
        </t>
        <t>
          <list style="symbols">
            <t>
        <ul spacing="normal">
          <li>
              Allow future calls to this verified user.
            </t>
            <t>
            </li>
          <li>
              Allow future calls to any verified user who is in my system
              address book (this only works with address book integration, of
              course).
            </t>
          </list>
        </t>
            </li>
        </ul>
        <t>
          Implementations SHOULD <bcp14>SHOULD</bcp14> also provide a different user interface
          indication when calls are in progress to users whose identities are
          directly verifiable.  <xref target="sec.proposal.comsec"/> target="sec.proposal.comsec" format="default"/> provides
          more on this.
        </t>
      </section>
      <section title="Communications Consent" anchor="sec.proposal.communications.consent"> anchor="sec.proposal.communications.consent" numbered="true" toc="default">
        <name>Communications Consent</name>
        <t>
          Browser client implementations of WebRTC MUST <bcp14>MUST</bcp14> implement ICE.  Server
          gateway implementations which operate only at public IP addresses MUST <bcp14>MUST</bcp14>
          implement either full ICE or ICE-Lite <xref target="RFC8445"/>. target="RFC8445" format="default"/>.
        </t>
        <t>
          Browser implementations MUST <bcp14>MUST</bcp14> verify reachability via ICE prior to
          sending any non-ICE packets to a given destination.  Implementations
          MUST NOT
          <bcp14>MUST NOT</bcp14> provide the ICE transaction ID to JavaScript during the
          lifetime of the transaction (i.e., during the period when the ICE
          stack would accept a new response for accept a new response for that transaction).  The JS <bcp14>MUST
          NOT</bcp14> be permitted to control the local ufrag and password, though it of
          course knows it.
        </t>
        <t>
          While continuing consent is required, the ICE <xref target="RFC8445" sectionFormat="comma" section="10"/> keepalives use STUN Binding Indications which are
          one-way and therefore not sufficient.

<!-- [rfced] Section 6.3:  Please advise regarding the following:

1. We do not see the word "keepalive" in Section 10 of RFC 8445, but
we do see it in 8445's Section 11.  Please confirm that "Section 10"
is correct here and will be clear to readers.

2. We found the use of "which" confusing here.  Are all STUN Binding
Indications one-way and therefore not sufficient (in which case "STUN
Binding Indications, which are" would be correct), or only some (in
which case "STUN Binding Indications that transaction).  The JS MUST
          NOT are" would be permitted to control correct)?

3. We found this comment, in the local ufrag and password, though it of
          course knows it.
        </t>
        <t> <!-- FIXME: XML file, just prior to this
sentence:  "FIXME: phrasing of first sentence still awkward --> awkward."
Please let us know how/if you want to fix the phrasing.

Original:
 While continuing consent is required, the ICE <xref
          target="RFC8445"/>; [RFC8445]; Section 10
 keepalives use STUN Binding Indications which are one-way and
 therefore not sufficient. -->

  The current WG consensus is to
          use ICE Binding Requests for continuing consent freshness. ICE already
          requires that implementations respond to such requests, so this
          approach is maximally compatible. A separate document will profile the
          ICE timers to be used; see <xref target="RFC7675"/>. target="RFC7675" format="default"/>.
        </t>
      </section>
      <section title="IP anchor="sec.proposal.ip.location.privacy" numbered="true" toc="default">
        <name>IP Location Privacy" anchor="sec.proposal.ip.location.privacy"> Privacy</name>
        <t>
          A side effect of the default ICE behavior is that the peer learns
          one's IP address, which leaks large amounts of location
          information. This has negative privacy consequences in some
          circumstances. The API requirements in this section are intended to
          mitigate this issue. Note that these requirements are not intended to
          protect the user's IP address from a malicious site. In general, the
          site will learn at least a user's server-reflexive address from any
          HTTP transaction.

<!-- [rfced] Section 6.4:  Per author feedback for RFC 8839 and per
other documents in this cluster, we hyphenated the term "server
reflexive".  Please let us know any objections.

Original:
 In general, the site will learn at
 least a user's server reflexive address from any HTTP transaction.

Currently:
 In general, the site will learn at
 least a user's server-reflexive address from any HTTP transaction. -->

  Rather, these requirements are intended to allow a
          site to cooperate with the user to hide the user's IP address from the
          other side of the call. Hiding the user's IP address from the server
          requires some sort of explicit privacy preserving privacy-preserving mechanism on the
          client (e.g., Tor Browser [https://www.torproject.org/projects/torbrowser.html.en]) <eref brackets="angle" target="https://www.torproject.org/projects/torbrowser.html.en"/>) and
          is out of scope for this specification.
        </t>

        <t>
          <list style="hanging">
            <t hangText="API Requirement:">
        <dl newline="false" spacing="normal">
          <dt>API Requirement:</dt>
          <dd>
              The API MUST <bcp14>MUST</bcp14> provide a mechanism to allow the JS to suppress ICE
              negotiation (though perhaps to allow candidate gathering) until
              the user has decided to answer the call [note: determining call. (Note: Determining when
              the call has been answered is a question for the JS.] JS.)  This
              enables a user to prevent a peer from learning their IP address if
              they elect not to answer a call and also from learning whether the
              user is online.
            </t>
          </list>
        </t>

        <t>
          <list style="hanging">
            <t hangText="API Requirement:">
            </dd>
        </dl>
        <dl newline="false" spacing="normal">
          <dt>API Requirement:</dt>
          <dd>
              The API MUST <bcp14>MUST</bcp14> provide a mechanism for the calling application JS to
              indicate that only TURN candidates are to be used. This prevents
              the peer from learning one's IP address at all.  This mechanism
              MUST
              <bcp14>MUST</bcp14> also permit suppression of the related address field, since
              that leaks local addresses.
            </t>
          </list>
        </t>

        <t>
          <list style="hanging">
            <t hangText="API Requirement:">
            </dd>
        </dl>
        <dl newline="false" spacing="normal">
          <dt>API Requirement:</dt>
          <dd>
              The API MUST <bcp14>MUST</bcp14> provide a mechanism for the calling application to
              reconfigure an existing call to add non-TURN candidates.  Taken
              together, this and the previous requirement allow ICE negotiation
              to start immediately on incoming call notification, thus reducing
              post-dial delay, but also to avoid disclosing the user's IP
              address until they have decided to answer. They also allow users
              to completely hide their IP address for the duration of the
              call. Finally, they allow a mechanism for the user to optimize
              performance by reconfiguring to allow non-TURN candidates during
              an active call if the user decides they no longer need to hide
              their IP address
            </t>
          </list>
        </t> address.
            </dd>
        </dl>
        <t>
          Note that some enterprises may operate proxies and/or NATs designed to
          hide internal IP addresses from the outside world. WebRTC provides no
          explicit mechanism to allow this function. Either such enterprises
          need to proxy the HTTP/HTTPS and modify the SDP and/or the JS, or
          there needs to be browser support to set the "TURN-only" policy
          regardless of the site's preferences.
        </t>
      </section>
      <section title="Communications Security" anchor="sec.proposal.comsec"> anchor="sec.proposal.comsec" numbered="true" toc="default">
        <name>Communications Security</name>
        <t>
          Implementations MUST <bcp14>MUST</bcp14> support SRTP <xref target="RFC3711"/>. target="RFC3711" format="default"/>.
          Implementations MUST <bcp14>MUST</bcp14> support DTLS <xref target="RFC6347"/> target="RFC6347" format="default"/> and
          DTLS-SRTP <xref target="RFC5763"/><xref target="RFC5764"/> target="RFC5763" format="default"/> <xref target="RFC5764" format="default"/> for SRTP
          keying. Implementations MUST <bcp14>MUST</bcp14> support SCTP over DTLS <xref
          target="RFC8261"/>. target="RFC8261" format="default"/>.
        </t>
        <t>
          All media channels MUST <bcp14>MUST</bcp14> be secured via SRTP and SRTCP. the
	  Secure Real-time Transport Control Protocol (SRTCP).  Media traffic MUST NOT <bcp14>MUST NOT</bcp14>
          be sent over plain (unencrypted) RTP or RTCP; that is, implementations MUST
          NOT <bcp14>MUST
          NOT</bcp14> negotiate cipher suites with NULL encryption modes.  DTLS-SRTP
          MUST
          <bcp14>MUST</bcp14> be offered for every media channel.  WebRTC implementations MUST NOT <bcp14>MUST NOT</bcp14>
          offer SDP Security Descriptions security descriptions <xref target="RFC4568"/> target="RFC4568" format="default"/> or select it if offered.
          A
          An SRTP MKI MUST NOT Master Key Identifier (MKI) <bcp14>MUST NOT</bcp14> be used.
        </t>
        <t>
          All data channels MUST <bcp14>MUST</bcp14> be secured via DTLS.
        </t>
        <t>
         All Implementations MUST implementations <bcp14>MUST</bcp14> support DTLS 1.2 with the
          TLS_ECDHE_ECDSA_WITH_AES_128_GCM_SHA256 cipher suite and the
          <xref target="FIPS186">P-256 target="FIPS186" format="default">P-256 curve</xref>.
          Earlier drafts of this specification required
          DTLS 1.0 with the cipher suite
          TLS_ECDHE_ECDSA_WITH_AES_128_CBC_SHA, and at the time of this
          writing some implementations do not support DTLS 1.2;
          endpoints which that support only DTLS 1.2 might encounter
          interoperability issues.
          The DTLS-SRTP protection profile
          SRTP_AES128_CM_HMAC_SHA1_80 MUST <bcp14>MUST</bcp14> be supported for
          SRTP.
          Implementations
          MUST
          <bcp14>MUST</bcp14> favor cipher suites which support (Perfect Perfect Forward Secrecy) PFS Secrecy (PFS)
          over non-PFS cipher suites and SHOULD <bcp14>SHOULD</bcp14> favor AEAD
	  Authenticated Encryption with Associated Data (AEAD) over non-AEAD cipher suites.
        </t>
        <t>
          Implementations MUST NOT <bcp14>MUST NOT</bcp14> implement DTLS renegotiation and MUST <bcp14>MUST</bcp14> reject
          it with a "no_renegotiation" alert if offered.</t>
        <t>
          Endpoints MUST NOT <bcp14>MUST NOT</bcp14> implement TLS False Start <xref target="RFC7918"/>.</t>

        <t>
          <list style="hanging">
            <t hangText="API Requirement:"> target="RFC7918" format="default"/>.</t>
        <dl newline="false" spacing="normal">
          <dt>API Requirement:</dt>
          <dd>
              The API MUST <bcp14>MUST</bcp14> generate a new authentication key pair for every new
              call by default.  This is intended to allow for unlinkability.
            </t>
            <t hangText="API Requirement:">
            </dd>
          <dt>API Requirement:</dt>
          <dd>
              The API MUST <bcp14>MUST</bcp14> provide a means to reuse a key pair for calls.  This
              can be used to enable key continuity-based authentication, and
              could be used to amortize key generation costs.
            </t>
            <t hangText="API Requirement:">
            </dd>
          <dt>API Requirement:</dt>
          <dd>
              Unless
              the user specifically configures an external key pair, different
              key pairs MUST <bcp14>MUST</bcp14> be used for each origin. (This avoids creating a
              super-cookie.)
            </t>
            <t hangText="API Requirement:">
            </dd>
          <dt>API Requirement:</dt>
          <dd>
              When DTLS-SRTP is used, the API MUST NOT <bcp14>MUST NOT</bcp14> permit the JS to obtain
              the negotiated keying material. This requirement preserves the
              end-to-end security of the media.
            </t>
          </list>
        </t>

        <t>
          <list style="hanging">
            <t hangText="UI Requirements: ">
            </dd>
        </dl>
        <dl newline="false" spacing="normal">
          <dt>UI Requirements:</dt>
          <dd>
              A user-oriented client MUST <bcp14>MUST</bcp14> provide an "inspector" interface which
              allows the user to determine the security characteristics of the
              media.
            </t>
            <t>
            </dd>
          <dt/>
          <dd>
              The following properties SHOULD <bcp14>SHOULD</bcp14> be displayed "up-front" in the
              browser chrome, i.e., without requiring the user to ask for them:
            </t>
            <t>
              <list style="symbols">
                <t>
            </dd>
          <dt/>
          <dd>
            <ul spacing="normal">
              <li>
                  A client MUST <bcp14>MUST</bcp14> provide a user interface through which a user
                  may determine the security characteristics for
                  currently-displayed
                  currently displayed audio and video stream(s)
                </t>

                <t> stream(s).
                </li>
              <li>
                  A client MUST <bcp14>MUST</bcp14> provide a user interface through which a user
                  may determine the security characteristics for transmissions
                  of their microphone audio and camera video.
                </t>

                <t>
                </li>
              <li>
                  If the far endpoint was directly verified, either via a
                  third-party verifiable X.509 certificate or via a Web IdP
                  mechanism (see <xref target="sec.generic.idp"/>) target="sec.generic.idp" format="default"/>), the "security
                  characteristics" MUST <bcp14>MUST</bcp14> include the verified information.  X.509
                  identities and Web IdP identities have similar semantics and
                  should be displayed in a similar way.
                </t>
              </list>
            </t>
            <t>
            </t>
            <t>
                </li>
            </ul>
          </dd>
          <dt/>
          <dd>
              The following properties are more likely to require some
              "drill-down" from the user:
            </t>
            <t>
              <list style="symbols">
                <t>
            </dd>
          <dt/>
          <dd>
            <ul spacing="normal">
              <li>
                  The "security characteristics" MUST <bcp14>MUST</bcp14> indicate the cryptographic
                  algorithms in use (For example: "AES-CBC".)
                </t>

                <t> (for example, "AES-CBC").
                </li>
              <li>
                  The "security characteristics" MUST <bcp14>MUST</bcp14> indicate whether PFS is
                  provided.
                </t>

                <t>
                </li>
              <li>
                  The "security characteristics" MUST <bcp14>MUST</bcp14> include some mechanism to
                  allow an out-of-band verification of the peer, such as a
                  certificate fingerprint or a Short Authentication String (SAS).
                  These are compared by the peers to authenticate one another.
                </t>
              </list>
            </t>
          </list>
        </t>
                </li>
            </ul>
          </dd>
        </dl>
      </section>
    </section>
    <section title="Web-Based anchor="sec.generic.idp" numbered="true" toc="default">
      <name>Web-Based Peer Authentication" anchor="sec.generic.idp"> Authentication</name>
      <t>
          In a number of cases, it is desirable for the endpoint (i.e., the
          browser) to be able to directly identify the endpoint on the other
          side without trusting the signaling service to which they are
          connected. For instance, users may be making a call via a federated
          system where they wish to get direct authentication of the other
          side. Alternately, they may be making a call on a site which they
          minimally trust (such as a poker site) but to someone who has an
          identity on a site they do trust (such as a social network.) network).
      </t>
      <t>
          Recently, a number of Web-based identity technologies (OAuth,
          Facebook Connect Connect, etc.) have been developed. While the
          details vary, what these technologies share is that they have a
          Web-based (i.e., HTTP/HTTPS) identity provider which that attests to Alice's
          identity. For instance, if Alice has an account at example.org, Alice could
          use the example.org identity provider to prove to others that Alice is
          alice@example.org.  The development of these technologies allows us to
          separate calling from identity provision: Alice could call you on a
          poker site but identify herself as alice@example.org.
      </t>
      <t>
          Whatever the underlying technology, the general principle is that the
          party which is being authenticated is NOT the signaling site but
          rather the user (and their browser). Similarly, the relying party is
          the browser and not the signaling site.  Thus, the browser MUST <bcp14>MUST</bcp14>
          generate the input to the IdP assertion process and
          display the results of the verification process to the user
          in a way which cannot be imitated by the calling site.
      </t>
      <t>
          The mechanisms defined in this document do not require the browser to
          implement any particular identity protocol or to support any
          particular IdP. Instead, this document provides a generic interface
          which any IdP can implement. Thus, new IdPs and protocols can be
          introduced without change to either the browser or the calling
          service. This avoids the need to make a commitment to any particular
          identity protocol, although browsers may opt to directly implement
          some identity protocols in order to provide superior performance or UI
          properties.
      </t>
      <section title="Trust anchor="sec.trust-relationships" numbered="true" toc="default">
        <name>Trust Relationships: IdPs, APs, and RPs" anchor="sec.trust-relationships"> RPs</name>
        <t>
            Any federated identity protocol has three major participants:
        </t>
          <t>
            <list style="hanging">
              <t hangText="Authenticating
        <dl newline="false" spacing="normal">
          <dt>Authenticating Party (AP):"> (AP):</dt>
          <dd>
                The entity which is trying to establish its identity.
              </t>
              <t>
              </t>

              <t hangText="Identity
              </dd>
          <dt>Identity Provider (IdP):"> (IdP):</dt>
          <dd>
                The entity which is vouching for the AP's identity.
              </t>

              <t>
              </t>

              <t hangText="Relying
              </dd>
          <dt>Relying Party (RP):"> (RP):</dt>
          <dd>
                The entity which is trying to verify the AP's identity.
              </t>
            </list>
          </t>
              </dd>
        </dl>
        <t>
            The AP and the IdP have an account relationship of some kind: the AP
            registers with the IdP and is able to subsequently authenticate
            directly to the IdP (e.g., with a password). This means that the
            browser must somehow know which IdP(s) the user has an account
            relationship with.  This can either be something that the user
            configures into the browser or that is configured at the calling
            site and then provided to the PeerConnection by the Web application
            at the calling site. The use case for having this information
            configured into the browser is that the user may "log into" the
            browser to bind it to some identity. This is becoming common in new
            browsers. However, it should also be possible for the IdP
            information to simply be provided by the calling application.
        </t>
        <t>
            At a high level level, there are two kinds of IdPs:
        </t>
          <t>
            <list style="hanging">
              <t hangText="Authoritative: ">
        <dl newline="false" spacing="normal">
          <dt>Authoritative:</dt>
          <dd>
                IdPs which have verifiable control of some section of the
                identity space. For instance, in the realm of e-mail, email, the
                operator of "example.com" has complete control of the namespace
                ending in "@example.com". Thus, "alice@example.com" is whoever
                the operator says operator says it is. Examples of systems with authoritative
                identity providers include DNSSEC, RFC 4474, and Facebook
                Connect (Facebook identities only make sense within the context
                of the Facebook system).

<!-- [rfced] Section 7.1:  May we cite RFC 8224 (which obsoletes
RFC 4474) here instead (with brackets, so that a hyperlink will be
available for the reader) and list it is. under Informative References?

Original:
 Examples of systems with authoritative
 identity providers include DNSSEC, RFC 4474, and Facebook Connect
 (Facebook identities only make sense within the context of the
 Facebook system).
              </t>

              <t>
              </t>
              <t hangText="Third-Party: ">

Possibly:
 Examples of systems with authoritative
 identity providers include DNSSEC, an identity system for SIP
 (see [RFC8224]), and Facebook Connect (Facebook identities only make
 sense within the context of the Facebook system).
...

 [RFC8224]  Peterson, J., Jennings, C., Rescorla, E., and C. Wendt,
            "Authenticated Identity Management in the Session
            Initiation Protocol (SIP)", RFC 8224, DOI 10.17487/RFC8224,
            February 2018, <https://www.rfc-editor.org/info/rfc8224>. -->

              </dd>
          <dt>Third-Party:</dt>
          <dd>
                IdPs which don't have control of their section of the identity
                space but instead verify a user's identities identity via some unspecified
                mechanism and then attest to it. Because the IdP doesn't
                actually control the namespace, RPs need to trust that the IdP
                is correctly verifying AP identities, and there can potentially
                be multiple IdPs attesting to the same section of the identity
                space. Probably the best-known example of a third-party identity
                provider is SSL/TLS certificates, where there are a large number of
                CAs
                certification authorities (CAs) all of whom can attest to any domain name.
              </t>
            </list>
          </t>
              </dd>
        </dl>
        <t>
            If an AP is authenticating via an authoritative IdP, then the RP
            does not need to explicitly configure trust in the IdP at all.  The
            identity mechanism can directly verify that the IdP indeed made the
            relevant identity assertion (a function provided by the mechanisms
            in this document), and any assertion it makes about an identity for
            which it is authoritative is directly verifiable. Note that this
            does not mean that the IdP might not lie, but that is a
            trustworthiness judgement that the user can make at the time he
            looks at the identity.
        </t>
        <t>
            By contrast, if an AP is authenticating via a third-party IdP, the
            RP needs to explicitly trust that IdP (hence the need for an
            explicit trust anchor list in PKI-based SSL/TLS clients). The list
            of trustable IdPs needs to be configured directly into the browser,
            either by the user or potentially by the browser manufacturer. This
            is a significant advantage of authoritative IdPs and implies that if
            third-party IdPs are to be supported, the potential number needs to
            be fairly small.
        </t>
      </section>
      <section title="Overview anchor="sec.overview" numbered="true" toc="default">
        <name>Overview of Operation" anchor="sec.overview"> Operation</name>
        <t>
            In order to provide security without trusting the calling site, the
            PeerConnection component of the browser must interact directly with
            the IdP. The details of the mechanism are described in the W3C API
            specification, but the general idea is that the PeerConnection
            component downloads JS from a specific location on the IdP dictated
            by the IdP domain name. That JS (the "IdP proxy") runs in an
            isolated security context within the browser browser, and the PeerConnection
            talks to it via a secure message passing channel.
        </t>
        <t>
            Note that there are two logically separate functions here:
            <list style="symbols">
              <t>
        </t>
        <ul spacing="normal">
          <li>
                Identity assertion generation.
              </t>
              <t>
              </li>
          <li>
                Identity assertion verification.
              </t>
            </list>
          </t>
              </li>
        </ul>
        <t>
            The same IdP JS "endpoint" is used for both functions functions, but of course
            a given IdP might behave differently and load new JS to perform one
            function or the other.
        </t>
          <figure>
            <artwork><![CDATA[
        <artwork name="" type="" align="left" alt=""><![CDATA[
     +--------------------------------------+
     | Browser                              |
     |                                      |
     | +----------------------------------+ |
     | | https://calling-site.example.com | |
     | |                                  | |
     | |        Calling JS Code           | |
     | |               ^                  | |
     | +---------------|------------------+ |
     |                 | API Calls          |
     |                 v                    |
     |          PeerConnection              |
     |                 ^                    |
     |                 | API Calls          |
     |     +-----------|-------------+      |   +---------------+
     |     |           v             |      |   |               |
     |     |       IdP Proxy         |<-------->|   Identity    |
     |     |                         |      |   |   Provider    |
     |     | https://idp.example.org |      |   |               |
     |     +-------------------------+      |   +---------------+
     |                                      |
     +--------------------------------------+ ]]></artwork>
          </figure>
        <t>
            When the PeerConnection object wants to interact with the IdP, the
            sequence of events is as follows:
            <list style="numbers">
              <t>
        </t>
        <ol spacing="normal" type="1">
          <li>
                The browser (the PeerConnection component) instantiates an IdP
                proxy. This allows the IdP to load whatever JS is necessary into
                the proxy.  The resulting code runs in the IdP's security
                context.
              </t>
              <t>
              </li>
          <li>
                The IdP registers an object with the browser that conforms to
                the API defined in <xref target="webrtc-api"/>.
              </t>
              <t> target="webrtc-api" format="default"/>.
              </li>
          <li>
                The browser invokes methods on the object registered by the IdP
                proxy to create or verify identity assertions.
              </t>
            </list>
          </t>
              </li>
        </ol>
        <t>
            This approach allows us to decouple the browser from any particular
            identity provider; the browser need only know how to load the IdP's
            JavaScript--the
            JavaScript -- the location of which is determined based on the IdP's
            identity--and
            identity -- and to call the generic API for requesting and verifying
            identity assertions. The IdP provides whatever logic is necessary to
            bridge the generic protocol to the IdP's specific
            requirements. Thus, a single browser can support any number of
            identity protocols, including being forward compatible with IdPs
            which did not exist at the time the browser was written.
        </t>
      </section>
      <section title="Items anchor="sec.standardized" numbered="true" toc="default">
        <name>Items for Standardization" anchor="sec.standardized"> Standardization</name>
        <t>
            There are two parts to this work:
        </t>
          <t>
            <list style="symbols">
              <t>
        <ul spacing="normal">
          <li>
                The precise information from the signaling message that must be
                cryptographically bound to the user's identity and a mechanism
                for carrying assertions in JSEP JavaScript Session Establishment
		Protocol (JSEP) messages. This is specified in
                <xref target="sec.jsep-binding"/>.
              </t>

              <t> target="sec.jsep-binding" format="default"/>.
              </li>
          <li>
                The interface to the IdP, which is defined in the companion W3C
                WebRTC API specification <xref target="webrtc-api"/>.
              </t>
            </list>
          </t> target="webrtc-api" format="default"/>.
              </li>
        </ul>
        <t>
            The WebRTC API specification also defines JavaScript interfaces that
            the calling application can use to specify which IdP to use.  That
            API also provides access to the assertion-generation capability and
            the status of the validation process.
        </t>
      </section>
      <section title="Binding anchor="sec.jsep-binding" numbered="true" toc="default">
        <name>Binding Identity Assertions to JSEP Offer/Answer Transactions" anchor="sec.jsep-binding"> Transactions</name>
        <t>
            An identity assertion binds the user's identity (as asserted by the
            IdP) to the SDP offer/answer exchange and specifically to the
            media. In order to achieve this, the PeerConnection must provide the
            DTLS-SRTP fingerprint to be bound to the identity. This is provided
            as a JavaScript object (also known as a dictionary or hash) with a
            single <spanx style="verb">fingerprint</spanx> "fingerprint" key, as shown below:
        </t>
          <figure>
            <artwork><![CDATA[
<!-- [rfced] Please review the type attribute set for each <sourcecode> and let
us know if any updates are needed.
-->
        <sourcecode name="json-1" type="json"><![CDATA[
{
  "fingerprint":
    [
      { "algorithm": "sha-256",
        "digest": "4A:AD:B9:B1:3F:...:E5:7C:AB" },
      { "algorithm": "sha-1",
        "digest": "74:E9:76:C8:19:...:F4:45:6B" }
    ]
}
]]></artwork>
          </figure> ]]></sourcecode>
        <t>
            The <spanx style="verb">fingerprint</spanx> "fingerprint" value is an array of
            objects.  Each object in the array contains <spanx
            style="verb">algorithm</spanx> "algorithm" and <spanx
            style="verb">digest</spanx> "digest" values, which correspond directly to
            the algorithm and digest values in the <spanx
            style="verb">fingerprint</spanx> "fingerprint" attribute of the SDP <xref
            target="RFC8122"/>. target="RFC8122" format="default"/>.
        </t>
        <t>
            This object is encoded in a <xref target="RFC8259">JSON</xref> target="RFC8259" format="default">JSON</xref>
            string for passing to the IdP.  The identity assertion returned by
            the IdP, which is encoded in the <spanx
            style="verb">identity</spanx> "identity" attribute, is a JSON object that is
            encoded as described in <xref target="sec.carry-assertion"/>. target="sec.carry-assertion" format="default"/>.
        </t>
        <t>
            This structure does not need to be interpreted by the IdP or the
            IdP proxy. It is consumed solely by the RP's browser.  The IdP
            merely treats it as an opaque value to be attested to.  Thus, new
            parameters can be added to the assertion without modifying the
            IdP.
        </t>
        <section title="Carrying anchor="sec.carry-assertion" numbered="true" toc="default">
          <name>Carrying Identity Assertions" anchor="sec.carry-assertion"> Assertions</name>
          <t>
              Once an IdP has generated an assertion (see <xref
              target="sec.request-assert"/>), target="sec.request-assert" format="default"/>), it is attached to the SDP
              offer/answer message. This is done by adding a new 'identity' "identity"
              attribute to the SDP. The sole contents of this value is the
              identity assertion.  The identity assertion produced by the IdP is
              encoded into a UTF-8 JSON text, then <xref
              target="RFC4648">Base64-encoded</xref> target="RFC4648" format="default">base64-encoded</xref> to produce this string.
              For example:
          </t>
            <figure>
              <artwork><![CDATA[

          <sourcecode name="sdp-1" type="sdp" ><![CDATA[
v=0
o=- 1181923068 1181923196 IN IP4 ua1.example.com
s=example1
c=IN IP4 ua1.example.com
a=fingerprint:sha-1 \
  4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB
a=identity:\
  eyJpZHAiOnsiZG9tYWluIjoiZXhhbXBsZS5vcmciLCJwcm90b2NvbCI6ImJvZ3Vz\
  In0sImFzc2VydGlvbiI6IntcImlkZW50aXR5XCI6XCJib2JAZXhhbXBsZS5vcmdc\
  IixcImNvbnRlbnRzXCI6XCJhYmNkZWZnaGlqa2xtbm9wcXJzdHV2d3l6XCIsXCJz\
  aWduYXR1cmVcIjpcIjAxMDIwMzA0MDUwNlwifSJ9
a=...
t=0 0
m=audio 6056 RTP/SAVP 0
a=sendrecv
...

  Note ]]></sourcecode>

  <aside><t>Note that long lines in the example are folded to meet the column
  width constraints of this document; the backslash ("\") at the end of
  a line, the carriage return that follows, and whitespace shall be ignored.

]]></artwork>
            </figure> ignored.</t></aside>

          <t>
              The 'identity' "identity" attribute attests to all <spanx
              style="verb">fingerprint</spanx> "fingerprint" attributes in the session
              description. It is therefore a session-level attribute.
          </t>
          <t>
              Multiple <spanx style="verb">fingerprint</spanx> "fingerprint" values can be
              used to offer alternative certificates for a peer.  The <spanx
              style="verb">identity</spanx> "identity" attribute MUST <bcp14>MUST</bcp14> include all
              fingerprint
              "fingerprint" values that are included in <spanx
              style="verb">fingerprint</spanx> "fingerprint" attributes of the session
              description.
          </t>
          <t>
              The RP browser MUST <bcp14>MUST</bcp14> verify that the in-use certificate for a DTLS
              connection is in the set of fingerprints returned from the IdP
              when verifying an assertion.
          </t>
        </section>
      </section>
      <section title="Determining anchor="sec.idp-uri" numbered="true" toc="default">
        <name>Determining the IdP URI" anchor="sec.idp-uri"> URI</name>
        <t>
                In order to ensure that the IdP is under control of the domain
                owner rather than someone who merely has an account on the
                domain owner's server (e.g., in shared hosting scenarios), the
                IdP JavaScript is hosted at a deterministic location based on
                the IdP's domain name.  Each IdP proxy instance is associated
                with two values:
        </t>
              <t>
                <list style="hanging">
                  <t hangText="Authority:">
        <dl newline="false" spacing="normal">
          <dt>authority:</dt>
          <dd>
                       The <xref target="RFC3986"> target="RFC3986" format="default"> authority</xref> at which the
                       IdP's service is hosted.
                  </t>
                  <t hangText="protocol:">
                  </dd>
          <dt>protocol:</dt>
          <dd>
                    The specific IdP protocol which the IdP is using. This is a
                    completely opaque IdP-specific string, but allows an IdP to
                    implement two protocols in parallel. This value may be the
                    empty string.  If no value for protocol is provided, a value
                    of "default" is used.
                  </t>
                </list>
              </t>
                  </dd>
        </dl>
        <t>
                Each IdP MUST <bcp14>MUST</bcp14> serve its initial entry page (i.e., the one loaded
                by the IdP proxy) from a <xref target="RFC5785">well-known target="RFC5785" format="default">well-known
                URI</xref>.

<!-- [rfced] Section 7.5:  RFC 5785 has been obsoleted by RFC 8615.
May we change both citations as well as the reference listing for
RFC 5785?

(It looks like <https://www.iana.org/assignments/well-known-uris/>
and <https://www.iana.org/assignments/uri-schemes/> might be related
to this text, and we see that both have been updated to refer to
RFC 8615.)

Original:
 Each IdP MUST serve its initial entry page (i.e., the one loaded by
 the IdP proxy) from a well-known URI [RFC5785].
...
This section reqisters the "idp-proxy" well-known URI from [RFC5785].
...
 [RFC5785]  Nottingham, M. and E. Hammer-Lahav, "Defining Well-Known
            Uniform Resource Identifiers (URIs)", RFC 5785,
            DOI 10.17487/RFC5785, April 2010,
            <https://www.rfc-editor.org/info/rfc5785>.

Suggested ("reqisters" has been fixed):
 Each IdP MUST serve its initial entry page (i.e., the one loaded by
 the IdP proxy) from a well-known URI [RFC8615].
...
 This section registers the "idp-proxy" well-known URI from [RFC8615].
...
 [RFC8615]  Nottingham, M., "Well-Known Uniform Resource Identifiers
            (URIs)", RFC 8615, DOI 10.17487/RFC8615, May 2019,
            <https://www.rfc-editor.org/info/rfc8615>. -->

  The well-known URI for an IdP proxy is formed from
                the following URI components:
                <list style="numbers">
                  <t>
        </t>
        <ol spacing="normal" type="1">
          <li>
                    The scheme, "https:".  An IdP MUST <bcp14>MUST</bcp14> be loaded using <xref
                    target="RFC2818">HTTPS</xref>.
                  </t>
                  <t> target="RFC2818" format="default">HTTPS</xref>.
                  </li>
          <li>
                    The <xref target="RFC3986">authority</xref>. target="RFC3986" format="default">authority</xref>.  As noted above,
                    the authority MAY <bcp14>MAY</bcp14> contain a  non-default port number or
                    userinfo sub-component.  Both are removed when determining
                    if an asserted identity matches the name of the IdP.
                  </t>
                  <t>
                  </li>
          <li>
                    The path, starting with "/.well-known/idp-proxy/" and
                    appended with the IdP protocol.  Note that the separator
                    characters '/' (%2F) and '\' (%5C) MUST NOT <bcp14>MUST NOT</bcp14> be permitted in
                    the protocol field, lest an attacker be able to direct
                    requests outside of the controlled "/.well-known/" prefix.
                    Query and fragment values MAY <bcp14>MAY</bcp14> be used by including '?' or
                    '#' characters.
                  </t>
                </list>
                  </li>
        </ol>
        <t>
                For example, for the IdP "identity.example.com" and the protocol
                "example", the URL would be:
        </t>
              <figure>
                <artwork><![CDATA[
  https://identity.example.com/.well-known/idp-proxy/example
  ]]></artwork>
              </figure>

  <ul empty="true"><li>&lt;https://identity.example.com/.well-known/idp-proxy/example&gt;</li></ul>

        <t>
                The IdP MAY <bcp14>MAY</bcp14> redirect requests to this URL, but they MUST <bcp14>MUST</bcp14> retain
                the "https" scheme.  This changes the effective origin of the
                IdP, but not the domain of the identities that the IdP is
                permitted to assert and validate. I.e., That is, the IdP is still
                regarded as authoritative for the original domain.
        </t>
        <section title="Authenticating Party"> numbered="true" toc="default">
          <name>Authenticating Party</name>
          <t>
                  How an AP determines the appropriate IdP domain is out of
                  scope of this specification. In general, however, the AP has
                  some actual account relationship with the IdP, as this
                  identity is what the IdP is attesting to. Thus, the AP somehow
                  supplies the IdP information to the browser. Some potential
                  mechanisms include:
                  <list style="symbols">
                    <t>
          </t>
          <ul spacing="normal">
            <li>
                      Provided by the user directly.
                    </t>
                    <t>
                    </li>
            <li>
                      Selected from some set of IdPs known to the calling site.
                      E.g., site
                      (e.g., a button that shows "Authenticate via Facebook
                      Connect"
                    </t>
                  </list>
                </t>
                      Connect").
                    </li>
          </ul>
        </section>
        <section title="Relying Party"> numbered="true" toc="default">
          <name>Relying Party</name>
          <t>
                  Unlike the AP, the RP need not have any particular
                  relationship with the IdP. Rather, it needs to be able to
                  process whatever assertion is provided by the AP.  As the
                  assertion contains the IdP's identity in the <spanx
                  style="verb">idp</spanx> "idp" field of the JSON-encoded object (see
                  <xref target="sec.request-assert"/>), target="sec.request-assert" format="default"/>), the URI can be
                  constructed directly from the assertion, and thus the RP can
                  directly verify the technical validity of the assertion with
                  no user interaction. Authoritative assertions need only be
                  verifiable. Third-party assertions also MUST <bcp14>MUST</bcp14> be verified
                  against local policy, as described in <xref
                  target="sec.id-format"/>. target="sec.id-format" format="default"/>.
          </t>
        </section>
      </section>
      <section title="Requesting Assertions" anchor="sec.request-assert"> anchor="sec.request-assert" numbered="true" toc="default">
        <name>Requesting Assertions</name>
        <t>
                The input to identity assertion is the JSON-encoded object
                described in <xref target="sec.jsep-binding"/> target="sec.jsep-binding" format="default"/> that contains the
                set of certificate fingerprints the browser intends to use.

<!-- [rfced] Section 7.6:  Should "input to identity assertion" be
"input to the IdP assertion process" (per Section 7) or possibly
"input to the assertion generation process" (per Section 8)?

Original:
 The input to identity assertion is the JSON-encoded object described
 in Section 7.4 that contains the set of certificate fingerprints the
 browser intends to use. -->

                This string is treated as opaque from the perspective of the
                IdP.
        </t>
        <t>
                The browser also identifies the origin that the PeerConnection
                is run in, which allows the IdP to make decisions based on who
                is requesting the assertion.
        </t>
        <t>
                An application can optionally provide a user identifier hint
                when specifying an IdP.  This value is a hint that the IdP can
                use to select amongst multiple identities, or to avoid providing
                assertions for unwanted identities.  The <spanx
                style="verb">username</spanx> "username" is a string that has no meaning to
                any entity other than the IdP, IdP; it can contain any data the IdP
                needs in order to correctly generate an assertion.
        </t>
        <t>
                An identity assertion that is successfully provided by the IdP
                consists of the following information:
        </t>
              <t>
                <list style="hanging">
                  <t hangText="idp:">
        <dl newline="false" spacing="normal">
          <dt>idp:</dt>
          <dd>
                    The domain name of an IdP and the protocol string.  This MAY <bcp14>MAY</bcp14>
                    identify a different IdP or protocol from the one that
                    generated the assertion.
                  </t>
                  <t hangText="assertion:">
                  </dd>
          <dt>assertion:</dt>
          <dd>
                    An opaque value containing the assertion itself. This is
                    only interpretable by the identified IdP or the IdP code
                    running in the client.
                  </t>
                </list>
              </t>
                  </dd>
        </dl>
        <t>
                <xref target="fig.assert-ex"/> target="fig.assert-ex" format="default"/> shows an example assertion
                formatted as JSON.  In this case, the message has presumably
                been digitally signed/MACed in some way that the IdP can later
                verify it, but this is an implementation detail and out of scope
                of this document.              </t>
        <figure title="Example assertion" anchor="fig.assert-ex">
                <artwork><![CDATA[
          <name>Example Assertion</name>
          <sourcecode name="json-2" type="json"><![CDATA[
{
  "idp":{
    "domain": "example.org",
    "protocol": "bogus"
  },
  "assertion": "{\"identity\":\"bob@example.org\",
                 \"contents\":\"abcdefghijklmnopqrstuvwyz\",
                 \"signature\":\"010203040506\"}"
}
]]></artwork> ]]></sourcecode>
        </figure>
        <t>
                For use in signaling, the assertion is serialized into JSON,
                <xref target="RFC4648">Base64-encoded</xref>, target="RFC4648" format="default">base64-encoded</xref>, and used as the
                value of the <spanx style="verb">identity</spanx> "identity" attribute.
                IdPs SHOULD <bcp14>SHOULD</bcp14> ensure that any assertions they
                generate cannot be interpreted in a different context. E.g., For example,
                they should use a distinct format or have separate cryptographic
                keys for assertion generation and other purposes.
                Line breaks are inserted solely for
                readability.
        </t>
      </section>
      <section title="Managing anchor="sec.user-login" numbered="true" toc="default">
        <name>Managing User Login" anchor="sec.user-login"> Login</name>
        <t>
                In order to generate an identity assertion, the IdP needs proof of
                the user's identity.  It is common practice to authenticate users
                (using passwords or multi-factor authentication), then use <xref
                target="RFC6265">Cookies</xref> target="RFC6265" format="default">cookies</xref> or <xref target="RFC7617">HTTP target="RFC7617" format="default">HTTP
                authentication</xref> for subsequent exchanges.
        </t>
        <t>
                The IdP proxy is able to access cookies, HTTP authentication data, or
                other persistent session data because it operates in the security
                context of the IdP origin.  Therefore, if a user is logged in, the
                IdP could have all the information needed to generate an
                assertion.
        </t>
        <t>
                An IdP proxy is unable to generate an assertion if the user is
                not logged in, or the IdP wants to interact with the user to
                acquire more information before generating the assertion.  If
                the IdP wants to interact with the user before generating an
                assertion, the IdP proxy can fail to generate an assertion and
                instead indicate a URL where login should proceed.
        </t>
        <t>
                The application can then load the provided URL to enable the
                user to enter credentials.  The communication between the
                application and the IdP is described in <xref
                target="webrtc-api"/>. target="webrtc-api" format="default"/>.
        </t>
      </section>
    </section>
    <section title="Verifying Assertions" anchor="sec.verify-assert"> anchor="sec.verify-assert" numbered="true" toc="default">
      <name>Verifying Assertions</name>
      <t>
                The input to identity validation is the assertion string taken
                from a decoded 'identity' "identity" attribute.
      </t>
      <t>
                The IdP proxy verifies the assertion. Depending on the identity
                protocol, the proxy might contact the IdP server or other
                servers.  For instance, an OAuth-based protocol will likely
                require using the IdP as an oracle, whereas with a
                signature-based scheme it might be able to verify the assertion
                without contacting the IdP, provided that it has cached the
                relevant public key.
      </t>
      <t>
                Regardless of the mechanism, if verification succeeds, a
                successful response from the IdP proxy consists of the following
                information:
                <list style="hanging">
                  <t hangText="identity:">
      </t>
      <dl newline="false" spacing="normal">
        <dt>identity:</dt>
        <dd>
                    The identity of the AP from the IdP's perspective. Details
                    of this are provided in <xref target="sec.id-format"/>.
                  </t>
                  <t hangText="contents:"> target="sec.id-format" format="default"/>.
                  </dd>
        <dt>contents:</dt>
        <dd>
                    The original unmodified string provided by the AP as input
                    to the assertion generation process.
                  </t>
                </list>
              </t>
                  </dd>
      </dl>
      <t>
                <xref target="fig.verify-ex"/> target="fig.verify-ex" format="default"/> shows an example response,
                which is JSON-formatted.
      </t>
      <figure title="Example verification result" anchor="fig.verify-ex">
                <artwork>
                  <![CDATA[
        <name>Example Verification Result</name>
        <sourcecode name="json-3" type="json"><![CDATA[
{
  "identity": "bob@example.org",
  "contents": "{\"fingerprint\":[ ... ]}"
}
]]></artwork> ]]></sourcecode>
      </figure>
      <section title="Identity Formats" anchor="sec.id-format"> anchor="sec.id-format" numbered="true" toc="default">
        <name>Identity Formats</name>
        <t>
                  The identity provided from the IdP to the RP browser MUST <bcp14>MUST</bcp14>
                  consist of a string representing the user's identity.  This
                  string is in the form "&lt;user>@&lt;domain>", "&lt;user&gt;@&lt;domain&gt;", where <spanx
                  style="verb">user</spanx> "user" consists of any character,
                  and domain is aninternationalized an internationalized
                  domain name <xref target="RFC5890"></xref> target="RFC5890" format="default"/> encoded as a sequence of U-labels.
        </t>
        <t>
                  The PeerConnection API MUST <bcp14>MUST</bcp14> check this string as follows:
                  <list style="numbers">
                    <t>
        </t>
        <ol spacing="normal" type="1">
          <li>
                      If the "domain" portion of the string is equal to the domain
                      name of the IdP proxy, then the assertion is valid, as the
                      IdP is authoritative for this domain.  Comparison of
                      domain names is done using the label equivalence rule
                      defined in Section 2.3.2.4 of <xref target="RFC5890"/>.
                    </t> target="RFC5890" sectionFormat="of" section="2.3.2.4"/>.
                    </li>
          <li>
            <t>
                      If the "domain" portion of the string is not equal to the
                      domain name of the IdP proxy, then the PeerConnection
                      object MUST <bcp14>MUST</bcp14> reject the assertion unless both:
                      <list style="numbers">
                        <t>
            </t>
            <ol spacing="normal" type="1">
              <li>
                          the IdP domain is trusted as an acceptable third-party
                          IdP; and
                        </t>
                        <t>
                        </li>
              <li>
                          local policy is configured to trust this IdP domain
                          for the domain portion of the identity string.
                        </t>
                      </list>
                    </t>
                  </list>
                </t>
                        </li>
            </ol>
          </li>
        </ol>
        <t>
                  Any "@" '@' or "%" '%' characters in the "user" portion of the
                  identity MUST <bcp14>MUST</bcp14> be escaped according to the "Percent-Encoding" "percent-encoding"
                  rules defined in Section 2.1 of <xref
                  target="RFC3986"/>. target="RFC3986" sectionFormat="of" section="2.1"/>. Characters other than "@" '@' and "%" MUST NOT '%' <bcp14>MUST NOT</bcp14>
                  be percent-encoded. For example, with a "user" of "user@133" and
                  a "domain" of "identity.example.com", the resulting string will
                  be encoded as "user%40133@identity.example.com".
        </t>
        <t>
                  Implementations are cautioned to take care when displaying
                  user identities containing escaped "@" '@' characters. If such
                  characters are unescaped prior to display, implementations
                  MUST
                  <bcp14>MUST</bcp14> distinguish between the domain of the IdP proxy and any
                  domain that might be implied by the portion of the
                  "&lt;user&gt;" portion that appears after the escaped "@"
                  sign.
        </t>
      </section>
    </section>
    <section title="Security Considerations" anchor="sec.sec-cons"> anchor="sec.sec-cons" numbered="true" toc="default">
      <name>Security Considerations</name>
      <t>
          Much of the security analysis of this problem is contained in <xref
          target="I-D.ietf-rtcweb-security"/> target="RFC8826" format="default"/> or in the discussion of the
          particular issues above.

<!-- [rfced] Section 9:  What does "this problem" refer to here?

Original:
 Much of the security analysis of this problem is contained in
 [I-D.ietf-rtcweb-security] or in the discussion of the particular
 issues above. -->

 In order to avoid repetition, this section
          focuses on (a) residual threats that are not addressed by this
          document and (b) threats produced by failure/misbehavior of one of the
          components in the system.
      </t>
      <section title="Communications Security"> numbered="true" toc="default">
        <name>Communications Security</name>
        <t>
            IF
            If HTTPS is not used to secure communications to the signaling
            server, and the identity mechanism used in
            <xref target="sec.generic.idp"/> target="sec.generic.idp" format="default"/> is not used,
            then any on-path attacker can replace the DTLS-SRTP fingerprints
            in the handshake and thus substitute its own identity for that
            of either endpoint.

<!-- [rfced] Section 9.1:  Should "the identity mechanism used in
Section 7" be "the identity mechanism used in Section 7.1" or
"the identity mechanisms used in Section 7"?  We ask because we see
"identity service mechanisms in Section 7" in the next paragraph.

Also, we changed "IF" to "If"; please let us know if the
capitalization was intentional.

Original:
 IF HTTPS is not used to secure communications to the signaling
 server, and the identity mechanism used in Section 7 is not used,
 then any on-path attacker can replace the DTLS-SRTP fingerprints in
 the handshake and thus substitute its own identity for that of either
 endpoint. -->

        </t>
        <t>
            Even if HTTPS is used, the signaling server can
            potentially mount a man-in-the-middle attack unless implementations
            have some mechanism for independently verifying keys. The UI
            requirements in <xref target="sec.proposal.comsec"/> target="sec.proposal.comsec" format="default"/> are designed to
            provide such a mechanism for motivated/security conscious users, but
            are not suitable for general use.  The identity service mechanisms
            in <xref target="sec.generic.idp"/> target="sec.generic.idp" format="default"/> are more suitable for general
            use. Note, however, that a malicious signaling service can strip off
            any such identity assertions, though it cannot forge new ones.  Note
            that all of the third-party security mechanisms available (whether
            X.509 certificates or a third-party IdP) rely on the security of the
            third party--this party -- this is of course also true of the user's connection to the
            Web site itself. Users who wish to assure themselves of security
            against a malicious identity provider can only do so by verifying
            peer credentials directly, e.g., by checking the peer's fingerprint
            against a value delivered out of band.
        </t>
        <t>
            In order to protect against malicious content JavaScript, that
            JavaScript MUST NOT <bcp14>MUST NOT</bcp14> be allowed to have direct
	    access to---or to -- or perform
            computations with---DTLS with -- DTLS keys. For instance, if content JS were able
            to compute digital signatures, then it would be possible for content
            JS to get an identity assertion for a browser's generated key and
            then use that assertion plus a signature by the key to authenticate
            a call protected under an ephemeral Diffie-Hellman (DH) key controlled by the content
            JS, thus violating the security guarantees otherwise provided by the
            IdP mechanism. Note that it is not sufficient merely to deny the
            content JS direct access to the keys, as some have suggested doing
            with the WebCrypto API <xref target="webcrypto"/>. target="webcrypto" format="default"/>.  The JS must
            also not be allowed to perform operations that would be valid for a
            DTLS endpoint. By far the safest approach is simply to deny the
            ability to perform any operations that depend on secret information
            associated with the key. Operations that depend on public
            information, such as exporting the public key key, are of course safe.
        </t>
      </section>
      <section title="Privacy"> numbered="true" toc="default">
        <name>Privacy</name>
        <t>
            The requirements in this document are intended to allow:
        </t>
          <t>
            <list style="symbols">
              <t>
        <ul spacing="normal">
          <li>
                Users to participate in calls without revealing their location.
              </t>
              <t>
              </li>
          <li>
                Potential callees to avoid revealing their location and even
                presence status prior to agreeing to answer a call.
              </t>
            </list>
          </t>
              </li>
        </ul>
        <t>
            However, these privacy protections come at a performance cost in
            terms of using TURN relays and, in the latter case, delaying
            ICE. Sites SHOULD <bcp14>SHOULD</bcp14> make users aware of these tradeoffs. trade&nbhy;offs.
        </t>
        <t>
            Note that the protections provided here assume a non-malicious
            calling service. As the calling service always knows the users user's
            status and (absent the use of a technology like Tor) their IP
            address, they can violate the users user's privacy at will.  Users who wish
            privacy against the calling sites they are using must use separate
            privacy enhancing
            privacy-enhancing technologies such as Tor. Combined &nbsp;Combined WebRTC/Tor
            implementations SHOULD <bcp14>SHOULD</bcp14> arrange to route the media as well as the
            signaling through Tor. Currently &nbsp;Currently this will produce very suboptimal
            performance.
        </t>
        <t>
            Additionally, any identifier which that persists across multiple calls is
            potentially a problem for privacy, especially for anonymous calling
            services. Such services SHOULD <bcp14>SHOULD</bcp14> instruct the browser to use separate
            DTLS keys for each call and also to use TURN throughout the
            call. Otherwise, the other side will learn linkable information that
            would allow them to correlate the browser across multiple calls.
            Additionally, browsers SHOULD <bcp14>SHOULD</bcp14> implement the privacy-preserving CNAME
            generation mode of <xref target="RFC7022"/>. target="RFC7022" format="default"/>.
        </t>
      </section>
      <section title="Denial numbered="true" toc="default">
        <name>Denial of Service"> Service</name>
        <t>
            The consent mechanisms described in this document are intended to
            mitigate denial of service DoS attacks in which an attacker uses clients
            to send large amounts of traffic to a victim without the consent of
            the victim. While these mechanisms are sufficient to protect victims
            who have not implemented WebRTC at all, WebRTC implementations need
            to be more careful.
        </t>
        <t>
            Consider the case of a call center which accepts calls via
            WebRTC. An attacker proxies the call center's front-end and arranges
            for multiple clients to initiate calls to the call center. Note that
            this requires user consent in many cases cases, but because the data
            channel does not need consent, he can use that directly. Since ICE
            will complete, browsers can then be induced to send large amounts of
            data to the victim call center if it supports the data channel at
            all. Preventing this attack requires that automated WebRTC
            implementations implement sensible flow control and have the ability
            to triage out (i.e., stop responding to ICE probes on) calls which
            are behaving badly, and especially to be prepared to remotely
            throttle the data channel in the absence of plausible audio and
            video (which the attacker cannot control).
        </t>
        <t>
            Another related attack is for the signaling service to swap the ICE
            candidates for the audio and video streams, thus forcing a browser
            to send video to the sink that the other victim expects will contain
            audio (perhaps it is only expecting audio!) audio!), potentially causing
            overload.  Muxing multiple media flows over a single transport makes
            it harder to individually suppress a single flow by denying ICE
            keepalives. Either media-level (RTCP) mechanisms must be used or the
            implementation must deny responses entirely, thus terminating the
            call.
        </t>
        <t>
            Yet another attack, suggested by Magnus Westerlund, is for the
            attacker to cross-connect offers and answers as follows. It induces
            the victim to make a call and then uses its control of other users users'
            browsers to get them to attempt a call to someone. It then
            translates their offers into apparent answers to the victim, which
            looks like large-scale parallel forking.  The victim still responds
            to ICE responses responses, and now the browsers all try to send media to the
            victim.  Implementations can defend themselves from this attack by
            only responding to ICE Binding Requests for a limited number of
            remote ufrags (this is the reason for the requirement that the JS
            not be able to control the ufrag and password).
        </t>
        <t>
            <xref target="I-D.ietf-rtcweb-rtp-usage"/> Section 13 target="RFC8834" sectionFormat="comma" section="13"/> documents a number
            of potential RTCP-based DoS attacks and countermeasures.
        </t>
        <t>
            Note that attacks based on confusing one end or the other about
            consent are possible even in the face of the third-party identity
            mechanism as long as major parts of the signaling messages are not
            signed. On the other hand, signing the entire message severely
            restricts the capabilities of the calling application, so there are
            difficult tradeoffs trade&nbhy;offs here.
        </t>
      </section>
      <section title="IdP numbered="true" toc="default">
        <name>IdP Authentication Mechanism"> Mechanism</name>
        <t>
            This mechanism relies for its security on the IdP and on the
            PeerConnection correctly enforcing the security invariants described
            above. At a high level, the IdP is attesting that the user
            identified in the assertion wishes to be associated with the
            assertion. Thus, it must not be possible for arbitrary third parties
            to get assertions tied to a user or to produce assertions that RPs
            will accept.
        </t>
        <section title="PeerConnection anchor="sec.pc-origin" numbered="true" toc="default">
          <name>PeerConnection Origin Check" anchor="sec.pc-origin"> Check</name>
          <t>
              Fundamentally, the IdP proxy is just a piece of HTML and JS loaded
              by the browser, so nothing stops a Web attacker from creating
              their own IFRAME, loading the IdP proxy HTML/JS, and requesting a
              signature over his own keys rather than those generated in
              the browser. However, that proxy would be in the
              attacker's origin, not the IdP's origin. Only the
              browser itself can instantiate a context that (a) is (a)&nbsp;is in the IdP's origin and
              (b) exposes
              (b)&nbsp;exposes the correct API surface. Thus, the IdP proxy on
              the sender's side MUST <bcp14>MUST</bcp14> ensure that it is running in the IdP's origin
              prior to issuing assertions.
          </t>
          <t>
              Note that this check only asserts that the browser (or some other
              entity with access to the user's authentication data) attests to
              the request and hence to the fingerprint.  It does not demonstrate
              that the browser has access to the associated private
              key, and therefore an attacker can attach their own identity
              to another party's keying material, thus making a call which
              comes from Alice appear to come from the attacker.
              See <xref target="I-D.ietf-mmusic-sdp-uks"/> target="RFC8844" format="default"/> for defenses against this
              form of attack.
          </t>
        </section>
        <section title="IdP Well-known URI" anchor="sec.sec-idp-uri"> anchor="sec.sec-idp-uri" numbered="true" toc="default">
          <name>IdP Well-Known URI</name>
          <t>
              As described in <xref target="sec.idp-uri"/> target="sec.idp-uri" format="default"/>, the IdP proxy HTML/JS
              landing page is located at a well-known URI based on the IdP's
              domain name. This requirement prevents an attacker who can write
              some resources at the IdP (e.g., on one's Facebook wall) from
              being able to impersonate the IdP.
          </t>
        </section>
        <section title="Privacy numbered="true" toc="default">
          <name>Privacy of IdP-generated identities IdP-Generated Identities and the hosting site"> Hosting Site</name>
          <t>
              Depending on the structure of the IdP's assertions, the calling
              site may learn the user's identity from the perspective of the
              IdP.  In many cases cases, this is not an issue because the user is
              authenticating to the site via the IdP in any case, case -- for instance instance,
              when the user has logged in with Facebook Connect and is then
              authenticating their call with a Facebook identity.  However, in
              other case, cases, the user may not have already revealed their identity
              to the site.  In general, IdPs SHOULD <bcp14>SHOULD</bcp14> either verify that the user
              is willing to have their identity revealed to the site (e.g.,
              through the usual IdP permissions dialog) or arrange that the
              identity information is only available to known RPs (e.g., social
              graph adjacencies) but not to the calling site. The "domain" field
              of the assertion request can be used to check that the user has
              agreed to disclose their identity to the calling site; because it
              is supplied by the PeerConnection it can be trusted to be correct.
          </t>
        </section>
        <section title="Security anchor="sec.sec-third-party" numbered="true" toc="default">
          <name>Security of Third-Party IdPs" anchor="sec.sec-third-party"> IdPs</name>
          <t>
              As discussed above, each third-party IdP represents a new
              universal trust point and therefore the number of these IdPs needs
              to be quite limited. Most IdPs, even those which issue unqualified
              identities such as Facebook, can be recast as authoritative IdPs
              (e.g., 123456@facebook.com). However, in such cases, the user
              interface implications are not entirely desirable.  One
              intermediate approach is to have a special (potentially user
              configurable) UI for large authoritative IdPs, thus allowing the
              user to instantly grasp that the call is being authenticated by
              Facebook, Google, etc.
          </t>
          <section title="Confusable Characters"> numbered="true" toc="default">
            <name>Confusable Characters</name>
            <t>
                Because a broad range of characters are permitted in identity
                strings, it may be possible for attackers to craft identities
                which are confusable with other identities (see
                <xref target="RFC6943"/> target="RFC6943" format="default"/> for more on this topic). This is
                a problem with any identifier space of this type
                (e.g., e-mail email addresses).
                Those minting identifers identifiers should avoid mixed scripts and similar
                confusable characters. Those presenting these identifiers to a
                user should consider highlighting cases of mixed script usage
                (see <xref target="RFC5890"/>, section 4.4). target="RFC5890" sectionFormat="comma" section="4.4"/>). Other best practices are still in development.
            </t>
          </section>
        </section>
        <section title="Web Security Feature Interactions">
            <t>
              A number of optional Web security features have the potential to
              cause issues for this mechanism, as discussed below.
            </t>

            <section title="Popup Blocking" anchor="sec.popup-blocking">
              <t>
                When popup blocking is in use, the IdP proxy is unable to generate popup windows, dialogs or
                any other form of user interactions.  This prevents the IdP
                proxy from being used to circumvent user interaction.  The
                "LOGINNEEDED" message allows the IdP proxy to inform the calling
                site of a need for user login, providing the information
                necessary to satisfy this requirement without resorting to
                direct user interaction from the IdP proxy itself.
              </t>
            </section>

            <section title="Third Party Cookies" anchor="sec.3rd-party-cookies">
              <t>
                Some browsers allow users to block third party cookies (cookies
                associated with origins other than the top level page) for
                privacy reasons.  Any IdP which uses cookies to persist logins
                will be broken by third-party cookie blocking. One option is to
                accept this as a limitation; another is to have the
                PeerConnection object disable third-party cookie blocking for
                the IdP proxy.
              </t>
            </section>

          </section>
        </section>
      </section>

      <section title="IANA Considerations" anchor="sec.iana-cons">
        <t>
          This specification defines the <spanx style="verb">identity</spanx>
          SDP attribute per the procedures of Section 8.2.4 of <xref
          target="RFC4566"/>.  The required information for the registration is
          included here:
          <list style="hanging">
            <t hangText="Contact Name:">IESG (iesg@ietf.org)</t>
            <t hangText="Attribute Name:">identity</t>
            <t hangText="Long Form:">identity</t>
            <t hangText="Type of Attribute:">session-level</t>
            <t hangText="Charset Considerations:">This attribute is not subject
            to the charset attribute.</t>
            <t hangText="Purpose:">This attribute carries an identity assertion,
            binding an identity to the transport-level security session.</t>
            <t hangText="Appropriate Values:">See <xref
            target="sec.sdp-id-attr"/> of RFCXXXX [[Editor Note: This
            document.]]</t>
            <t hangText="Mux Category:">NORMAL.</t>
          </list>
        </t>
        <t>
          This section reqisters the <spanx style="verb">idp-proxy</spanx> well-known
          URI from <xref target="RFC5785"/>.
          <list style="hanging">
             <t hangText="URI suffix:">idp-proxy</t>
             <t hangText="Change controller:">IETF</t>
          </list>
        </t>
      </section>

    <section title="Acknowledgements">
      <t>
        Bernard Aboba, Harald Alvestrand, Richard Barnes, Dan Druta, Cullen
        Jennings, Hadriel Kaplan, Matthew Kaufman, Jim McEachern, Martin
        Thomson, Magnus Westerland.  Matthew Kaufman provided the UI material in
        <xref target="sec.proposal.comsec"/>. Christer Holmberg provided
        the initial version of <xref target="sec.sdp-id-attr-oa"/>.
      </t>
    </section>

    <section title="Changes">
      <t> [RFC Editor: Please remove this section prior to publication.]</t>
      <section title="Changes since -15">
        <t>Rewrite the Identity section in more conventional offer/answer format.</t>
        <t>Clarify rules on changing identities.</t>
      </section>

      <section title="Changes since -11"> numbered="true" toc="default">
          <name>Web Security Feature Interactions</name>
          <t>
          Update discussion
              A number of IdP optional Web security model
        </t>

        <t>
          Replace "domain name" with RFC 3986 Authority
        </t>

        <t>
          Clean up discussion of how features have the potential to generate IdP URI.
        </t>

        <t>
          Remove obsolete text about null cipher suites.
        </t>

        <t>
          Remove obsolete appendixes about older IdP systems
        </t>

        <t>
          Require support
              cause issues for ECDSA, PFS, and AEAD this mechanism, as discussed below.
          </t>
      </section>
          <section title="Changes since -10">
        <t>
          Update cipher suite profiles.
        </t> anchor="sec.popup-blocking" numbered="true" toc="default">
            <name>Popup Blocking</name>
            <t>
          Rework IdP interaction based on implementation experience
                When popup blocking is in
          Firefox.
        </t>
      </section>

      <section title="Changes since -06">
        <t>
          Replaced RTCWEB and RTC-Web with WebRTC, except when referring to use, the
          IETF WG
        </t>
        <t>
          Forbade use in mixed content as discussed in Orlando.
        </t>
        <t>
          Added a requirement IdP proxy is unable to surface NULL ciphers generate popup windows, dialogs, or
                any other form of user interactions.  This prevents the IdP
                proxy from being used to circumvent user interaction.  The
                "LOGINNEEDED" message allows the top-level.
        </t>
        <t>
          Tried IdP proxy to clarify SRTP versus DTLS-SRTP.
        </t>
        <t>
          Added inform the calling
                site of a section on screen sharing permissions.
        </t>
        <t>
          Assorted editorial work.
        </t>
      </section>

      <section title="Changes since -05">
        <t>
          The following changes have been made since need for user login, providing the -05 draft.
        </t>
        <t>
          <list style="symbols">
            <t>
              Response information
                necessary to comments satisfy this requirement without resorting to
                direct user interaction from Richard Barnes
            </t>
            <t>
              More explanation of the IdP security properties and the federation
              use case.
            </t>
            <t>
              Editorial cleanup.
            </t>
          </list> proxy itself.
            </t>
          </section>
          <section title="Changes since -03">
        <t>
          Version -04 was a version control mistake. Please ignore.
        </t> anchor="sec.3rd-party-cookies" numbered="true" toc="default">
            <name>Third Party Cookies</name>
            <t>
          The following changes
                Some browsers allow users to block third party cookies (cookies
                associated with origins other than the top-level page) for
                privacy reasons.  Any IdP which uses cookies to persist logins
                will be broken by third-party cookie blocking. One option is to
                accept this as a limitation; another is to have been made since the -04 draft.
        </t>
        <t>
          <list style="symbols">
            <t>
              Move origin check from
                PeerConnection object disable third-party cookie blocking for
                the IdP to RP per discussion in YVR.
            </t>
            <t>
              Clarified treatment of X.509-level identities.
            </t>
            <t>
              Editorial cleanup.
            </t>
          </list> proxy.
            </t>
          </section>

      <section title="Changes since -03">
        </section>
      </section>
    </section>
    <section title="Changes since -02"> anchor="sec.iana-cons" numbered="true" toc="default">
      <name>IANA Considerations</name>
      <t>
          This specification defines the "identity"
          SDP attribute per the procedures of <xref target="RFC4566" sectionFormat="of" section="8.2.4"/>.  The following changes have been made since required information for the -02 draft.
        </t>
        <t>
          <list style="symbols">
            <t>
              Forbid persistent HTTP permissions. registration is
          included here:
      </t>
            <t>
              Clarified the text in S 5.4 to clearly refer
      <dl newline="false" spacing="normal">
        <dt>Contact Name:</dt>
        <dd>IESG (iesg@ietf.org)</dd>
        <dt>Attribute Name:</dt>
        <dd>identity</dd>
        <dt>Long Form:</dt>
        <dd>identity</dd>
        <dt>Type of Attribute:</dt>
        <dd>session</dd>
        <dt>Charset Considerations:</dt>
        <dd>This attribute is not subject
            to requirements on the API to provide functionality charset attribute.</dd>
        <dt>Purpose:</dt>
        <dd>This attribute carries an identity assertion,
            binding an identity to the site.
            </t>
            <t>
              Fold in the IETF portion transport-level security session.</dd>
        <dt>Appropriate Values:</dt>
        <dd>See <xref target="sec.sdp-id-attr" format="default"/> of draft-rescorla-rtcweb-generic-idp
            </t> RFC 8827.</dd>
        <dt>Mux Category:</dt>
        <dd>NORMAL</dd>
      </dl>
      <t>
              Retarget the continuing consent
          This section to assume Binding Requests
            </t>
            <t>
              Added some more privacy and linkage text in various places.
            </t>
            <t>
              Editorial improvements
            </t>
          </list> registers the "idp-proxy" well-known
          URI from <xref target="RFC5785" format="default"/>.
      </t>
      </section>
      <dl newline="false" spacing="normal">
        <dt>URI suffix:</dt>
        <dd>idp-proxy</dd>
        <dt>Change controller:</dt>
        <dd>IETF</dd>
      </dl>
    </section>
  </middle>
  <back>

    <references title="Normative References">
      &RFC2119;
      &RFC2818;
      &RFC3264;
      &RFC3711;
      &RFC3986;
      &RFC4566;
      &RFC4568;
      &RFC4648;
      &RFC5246;
      &RFC5763;
      &RFC5764;
      &RFC5785;
      &RFC5890;
      &RFC6347;
      &RFC6454;
      &RFC7022;
      &RFC7675;
      &RFC7918;
      &RFC8174;
      &RFC8122;
      &RFC8259;
      &RFC8261;
      &RFC8445;

      &I-D.ietf-rtcweb-overview;
      &I-D.ietf-rtcweb-security;
      &I-D.ietf-rtcweb-rtp-usage;
      &I-D.ietf-mmusic-sdp-uks;
      &I-D.ietf-rtcweb-jsep;
    <references>
      <name>References</name>
      <references>
        <name>Normative References</name>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.2119.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.2818.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.3264.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.3711.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.3986.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.4566.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.4568.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.4648.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.5246.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.5763.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.5764.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.5785.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.5890.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.6347.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.6454.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.7022.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.7675.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.7918.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.8174.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.8122.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.8259.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.8261.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.8445.xml"/>

<!-- draft-ietf-rtcweb-overview: RFC 8825 -->
<reference anchor="RFC8825" target="https://www.rfc-editor.org/info/rfc8825">
  <front>
    <title>Overview: Real-Time Protocols for Browser-Based Applications</title>
    <author initials="H." surname="Alvestrand" fullname="Harald T. Alvestrand">
      <organization />
    </author>
    <date month="June" year="2020" />
  </front>
  <seriesInfo name="RFC" value="8825" />
  <seriesInfo name="DOI" value="10.17487/RFC8825"/>
</reference>

 <!--draft-ietf-rtcweb-security: RFC 8826 -->
 <reference anchor="RFC8826" target="https://www.rfc-editor.org/info/rfc8826">
 <front>
 <title>Security Considerations for WebRTC</title>
 <author initials='E.' surname='Rescorla' fullname='Eric Rescorla'>
   <organization/>
 </author>
 <date month='June' year='2020'/>
 </front>
 <seriesInfo name="RFC" value="8826"/>
 <seriesInfo name="DOI" value="10.17487/RFC8826"/>
 </reference>

<!-- draft-ietf-rtcweb-rtp-usage; RFC 8834 -->
<reference anchor="RFC8834" target="https://www.rfc-editor.org/info/rfc8834">
  <front>
    <title>Media Transport and Use of RTP in WebRTC</title>
    <author initials="C." surname="Perkins" fullname="Colin Perkins">
      <organization />
    </author>
    <author initials="M." surname="Westerlund" fullname="Magnus Westerlund">
      <organization />
    </author>
    <author initials="J." surname="Ott" fullname="Jörg Ott">
      <organization />
    </author>
    <date month="June" year="2020" />
  </front>
  <seriesInfo name="RFC" value="8834" />
  <seriesInfo name="DOI" value="10.17487/RFC8834"/>
</reference>

<!-- draft-ietf-mmusic-sdp-uks; RFC 8844 -->
<reference anchor='RFC8844' target="https://www.rfc-editor.org/info/rfc8844">
<front>
<title>Unknown Key Share Attacks on uses of TLS with the Session Description Protocol (SDP)</title>

<author initials='M' surname='Thomson' fullname='Martin Thomson'>
    <organization />
</author>

<author initials='E' surname='Rescorla' fullname='Eric Rescorla'>
    <organization />
</author>

<date month="June" year="2020"/>

</front>
<seriesInfo name="RFC" value="8859"/>
<seriesInfo name="DOI" value="10.17487/RFC8859"/>

</reference>

<!-- draft-ietf-rtcweb-jsep; RFC 8829 -->
 <reference anchor="RFC8829" target="https://www.rfc-editor.org/info/rfc8829">
 <front>
 <title>JavaScript Session Establishment Protocol (JSEP)</title>
 <author initials='J.' surname='Uberti' fullname='Justin Uberti'>
   <organization/>
 </author>
 <author initials="C." surname="Jennings" fullname="Cullen Jennings">
   <organization/>
 </author>
 <author initials="E." surname="Rescorla" fullname="Eric Rescorla"
         role="editor">
 <organization/>
 </author>
 <date month='June' year='2020'/>
 </front>
 <seriesInfo name="RFC" value="8829"/>
 <seriesInfo name="DOI" value="10.17487/RFC8829"/>
 </reference>

        <reference anchor="webcrypto"> anchor="webcrypto" target="https://www.w3.org/TR/2017/REC-WebCryptoAPI-20170126/">
          <front>
            <title>Web Cryptography API</title>
            <author fullname="W3C editors"
                  surname="Dahl, Sleevi">
            <organization>W3C</organization> initials="M" surname="Watson" fullname="Mark Watson">
            </author>
            <date day="25" month="June" year="2013" /> month="January" year="2017" day="26"/>
          </front>

        <annotation>Available
            <refcontent>W3C Recommendation</refcontent>
        </reference>

<!-- [rfced] Normative References:  Because the citation for
[webcrypto] is used generally in text, we updated this listing per
<https://www.w3.org/TR/WebCryptoAPI/>.  Please let us know any
objections.

Original:
 [webcrypto]
            editors, W., "Web Cryptography API", June 2013.

            Available at
        http://www.w3.org/TR/WebCryptoAPI/</annotation>
      </reference> http://www.w3.org/TR/WebCryptoAPI/

Currently:
 [webcrypto]
            Watson, M., "Web Cryptography API", W3C Recommendation,
            26 January 2017,
            <https://www.w3.org/TR/2017/REC-WebCryptoAPI-20170126/>.
-->

        <reference anchor="webrtc-api"> anchor="webrtc-api" target="https://www.w3.org/TR/2019/CR-webrtc-20191213/">
          <front>
            <title>WebRTC 1.0: Real-time Communication Between Browsers</title>
            <author fullname="W3C editors"
                  surname="Bergkvist, Burnett, Jennings, Narayanan">
            <organization>W3C</organization> initials="C." surname="Jennings" fullname="Cullen Jennings">
              <organization/>
            </author>
            <author initials="H." surname="Boström" fullname="Henrik Boström">
              <organization/>
            </author>
            <author initials="J-I." surname="Bruaroey" fullname="Jan-Ivar Bruaroey">
              <organization/>
            </author>
            <date day="4" month="October" year="2011" /> year="2019" month="December" day="13"/>
          </front>

        <annotation>Available at
        http://dev.w3.org/2011/webrtc/editor/webrtc.html</annotation>
            <refcontent>W3C Candidate Recommendation</refcontent>
        </reference>

<!-- [rfced] Normative References:  The URL as provided for
[webrtc-api] in the original document -
<http://dev.w3.org/2011/webrtc/editor/webrtc.html> - steers to
<http://w3c.github.io/webrtc-pc/>, dated October 2019.  Please note
that this GitHub page says "Editor's draft" and also says
"Latest published version: https://www.w3.org/TR/webrtc/."  We have updated
this to refer to the "Latest published version".  Please let us know any
objections.

Original:
 [webrtc-api]
            editors, W., "WebRTC 1.0: Real-time Communication Between
            Browsers", October 2011.

            Available at http://dev.w3.org/2011/webrtc/editor/
            webrtc.html

Currently:
 [webrtc-api]
     Jennings, C., Boström, H., and J-I. Bruaroey, "WebRTC 1.0:
     Real-time Communication Between Browsers", W3C Candidate
     Recommendation, 13 December 2019,
     <https://www.w3.org/TR/2019/CR-webrtc-20191213/>.
-->

        <reference anchor="FIPS186">
          <front>
            <title>Digital Signature Standard (DSS)</title>
          <author >
            <author>
              <organization>National Institute of Standards and Technology (NIST)</organization>
            </author>
            <date year="2013" month="July"/>
          </front>
            <seriesInfo name="NIST PUB 186-4" value=""/> PUB" value="186-4"/>
            <seriesInfo name="DOI" value="10.6028/NIST.FIPS.186-4"/>
        </reference>
      </references>

   <references title="Informative References">
      &RFC7617;
      &RFC3261;
      &RFC5705;
      &RFC6455;
      &RFC6265;
      &RFC6943;
      &RFC6120;
      <references>
        <name>Informative References</name>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.7617.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.3261.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.5705.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.6455.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.6265.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.6943.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.6120.xml"/>
        <reference anchor="XmlHttpRequest"> anchor="XmlHttpRequest" target="https://www.w3.org/TR/XMLHttpRequest/">
          <front>
            <title>XMLHttpRequest Level 2</title>
            <author initials="A." surname="van Kesteren">
            <organization></organization>
              <organization/>
            </author>
            <date day="17" month="January" year="2012"/>
          </front>
        <format target="http://www.w3.org/TR/XMLHttpRequest/" type="TXT"/>
        </reference>

<!-- [rfced] Informative References:  The URL as provided for
[XmlHttpRequest] in the original document -
<http://www.w3.org/TR/XMLHttpRequest/> - steers to a page with the
title "XMLHttpRequest Level 1," dated October 2016.  When we did a
Google search for "XMLHttpRequest Level 2," we found
<https://www.w3.org/TR/2012/WD-XMLHttpRequest-20120117/>, which is
partially obscured by a red box that says "This version is
outdated!"  The link in the box in turn steers to the October 2016
"XMLHttpRequest Level 1" page.

Please advise.

Original:
 [XmlHttpRequest]
            van Kesteren, A., "XMLHttpRequest Level 2", January 2012. -->

      </references>
    </references>

    <section numbered="false" toc="default">
      <name>Acknowledgements</name>
      <t>
        <contact fullname="Bernard Aboba"/>, <contact fullname="Harald
	Alvestrand"/>, <contact fullname="Richard Barnes"/>, <contact
	fullname="Dan Druta"/>, <contact fullname="Cullen
        Jennings"/>, <contact fullname="Hadriel Kaplan"/>, <contact
	fullname="Matthew Kaufman"/>, <contact fullname="Jim McEachern"/>,
	<contact fullname="Martin Thomson"/>, <contact fullname="Magnus
	Westerlund"/>.  <contact fullname="Matthew Kaufman"/> provided the UI material in
        <xref target="sec.proposal.comsec" format="default"/>. <contact fullname="Christer Holmberg"/> provided
        the initial version of <xref target="sec.sdp-id-attr-oa" format="default"/>.
      </t>
    </section>
  </back>

<!-- [rfced] Please let us know if any changes are needed for the
following:

a) The following term was used inconsistently in this document.
We chose to use the latter form.  Please let us know any objections.

 Cookies ("use Cookies") / cookies ("access cookies") (Section 7.7)

b) In the v2 XML file, <spanx style="verb"> was used to create single quotes
for some keys, values, and attribute names.  Per RFC 7991, the xml2rfc v3
vocab, <spanx> has been deprecated:

   Deprecate <spanx>; replace it with <strong>, <em>, and <tt>.

C238 uses single or double quotes when referring to SDP attributes. Note that
we have replaced instances of <spanx> with double quotes.  Please let us know
if any updates are needed.

c) Please let us know how/if the following should be made consistent:

 interdomain ("interdomain calling") /
   inter-domain ("inter-domain protocol")
     (Usage post-RFC 6000 is mixed but leans heavily toward
     "inter-domain.")

 Identity Providers ("overview of Identity Providers and the relevant
   terminology") / identity providers

   The rest of this document, and the rest of the documents in
     Cluster 238, use the lowercase form.  Changing "overview of
     Identity Providers" to "overview of IdPs" in this document would
     resolve this issue.

 Relying Party (Section 4) / relying party (Section 7)

 security characteristics / "security characteristics"

 "https:" (The scheme, "https:") / "https" scheme (the "https" scheme)
-->

</rfc>