One document matched: draft-ietf-rtcweb-gateways-02.xml

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<rfc category="std" docName="draft-ietf-rtcweb-gateways-02" ipr="trust200902">
  <front>
    <title abbrev="WebRTC Gateways">WebRTC Gateways</title>

    <author fullname="Harald Alvestrand" initials="H. T." surname="Alvestrand">
      <organization>Google</organization>

      <address>
        <email>harald@alvestrand.no</email>
      </address>
    </author>

    <author fullname="Uwe Rauschenbach" initials="U." surname="Rauschenbach">
      <organization>Nokia Networks</organization>

      <address>
        <email>uwe.rauschenbach@nokia.com</email>
      </address>
    </author>

    <date day="21" month="January" year="2016"/>

    <workgroup>RTCWeb Working Group</workgroup>

    <abstract>
      <t>This document describes interoperability considerations for a class
      of WebRTC-compatible endpoints called "WebRTC gateways", which
      interconnect between WebRTC endpoints and devices that are not WebRTC
      endpoints.</t>
    </abstract>

    <note title="Requirements Language">
      <t>The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
      "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
      document are to be interpreted as described in <xref
      target="RFC2119">RFC 2119</xref>.</t>
    </note>
  </front>

  <middle>
    <section title="Introduction">
      <t>The WebRTC model described in <xref
      target="I-D.ietf-rtcweb-overview"/> is focused on direct browser to
      browser communication as its primary use case. Nevertheless, it is
      clearly interesting to have WebRTC endpoints connect to other types of
      devices, including but not limited to SIP phones, legacy phones,
      CLUE-based teleconferencing systems, XMPP-based conferencing systems,
      and entirely proprietary devices or systems.</t>

      <t>WebRTC gateways are middle boxes which enable the exchange of media
      streams between WebRTC endpoints on one side, and the other types of
      devices mentioned above on the other side. To a WebRTC endpoint, the
      gateway appears as a WebRTC-compatible endpoint.</t>

      <t>This document describes the requirements that need to be placed on
      such gateways, both the requirements on WebRTC endpoints that can be
      relaxed and the additional requirements that need to be applied.</t>

      <t>A WebRTC gateway appears as a WebRTC-compatible endpoint, and will
      thus not be conformant with all requirements for a WebRTC endpoint (it
      does not do everything a WebRTC endpoint does), but is able to
      interoperate with WebRTC endpoints.</t>

      <t>NOTE IN DRAFT: There is still not a WG consensus called on whether
      this document is Informational or standards-track. If it becomes
      informational, the use of RFC 2119 language is used to call attention to
      features where non-conformance will render a gateway unable to
      interoperate with WebRTC-based endpoints.</t>

      <section title="Implications of the gateway environment">
        <t>A gateway will be limited in the functionality it can offer by the
        system or class of devices it is gatewaying to. For instance, a
        gateway into the telephone system will not be able to relay data or
        video, no matter how much it is required. Therefore, a number of
        functions that are mandatory to support in WebRTC endpoints are not
        mandatory on gateways; the requirement on the gateway is that it is
        able to negotiate those features away correctly.</t>
      </section>

      <section title="Signalling model">
        <t>The WebRTC model is that signalling is outside the scope of the
        specification. This document does not change that.</t>

        <t>Nevertheless, any practical gateway needs to deal with signalling.
        For that, this document assumes that the overall system consists of an
        application running in the WebRTC browser, possibly one or more
        signalling relays that mediate signalling and thereby enable
        communication between the application and the gateway, and the actual
        gateway that is responsible for handling the media flows.</t>

        <t>The application, the signalling relays (if any) and the gateway
        together need to be able to:</t>

        <t><list style="symbols">
            <t>adhere to the offer/answer semantics</t>

            <t>deal with the description of configuration coming from the
            browser; this is specified in SDP format in the WebRTC browser
            API</t>

            <t>generate the information that is needed by the browser to set
            up the session, and express that information in the form of
            SDP.</t>
          </list> The shorthand notation "The gateway MUST/SHOULD/MAY support
        <SDP function xxx>" used below means that an application running
        in the Web browser, the signalling relays, and the gateway together
        MUST/SHOULD/MAY support this functionality; it is not a requirement
        that this happens at the media gateway itself.</t>
      </section>
    </section>

    <section title="WebRTC non-browser requirements that can be relaxed">
      <t>WebRTC gateways are intended to communicate with WebRTC
      endpoints<xref target="I-D.ietf-rtcweb-overview"/>. Some features that
      typical WebRTC endpoints are required to support may be meaningless or
      unneccesary for WebRTC gateways; some such things are noted in this
      section. This lack of conformance means that a gateway is considered a
      WebRTC-compatible endpoint, not a WebRTC endpoint (unless a particular
      gateway claims to be a WebRTC endpoint, which it is of course allowed to
      do).</t>

      <t>A WebRTC gateway which is expected to be deployed where it can be
      reached with a static IP address (as seen from the client) does not need
      to support full ICE; it therefore MAY implement ICE-Lite only.</t>

      <t>ICE-Lite implementations do not send consent checks, so a gateway MAY
      choose not to send consent checks too, but MUST respond to consent
      checks it receives.</t>

      <t>A gateway with a static IP address is expected to not need to hide
      its location, so it does not need to support functionality for operating
      only via a TURN server; instead it MAY choose to produce Host ICE
      candidates only.</t>

      <t>If a gateway serves as a media relay into another RTP domain, it MAY
      choose to support only features available in that network. This means
      that it MAY choose to not support Bundle and any of the RTP/RTCP
      extensions related to it, RTCP-Mux, or Trickle Ice. However, the gateway
      MUST support DTLS-SRTP, since this is required for interworking with
      WebRTC endpoints.</t>

      <t>Note that non-support of BUNDLE means that "bundle-only" tracks are
      not supported. This means that applications using an RTCBundlePolicy
      other than "max-compat" (<xref target="I-D.ietf-rtcweb-jsep"/> section
      4.1.1) can only use one track of each media type.</t>

      <t>If a gateway serves as a media relay into a network or to devices not
      implementing the WebRTC Datachannel, it MAY choose to not support the
      Datachannel.</t>
    </section>

    <section title="Additional WebRTC gateway requirements">
      <t>(nothing yet)</t>
    </section>

    <section title="Considerations for SDP-using networks">
      <t>Some networks that are gatewayed into, such as SIP networks, will
      also use SDP to represent the media configurations. Gateways will,
      however, need to inspect and probably modify the SDP passed between the
      SDP-using network and the WebRTC endpoints to achieve maximum
      interoperability.</t>

      <t>Considerations include:</t>

      <t><list style="symbols">
          <t>If a correspondent does not offer the features WebRTC depends on,
          connections will not complete. The support for dtls-srtp, shown by
          the "fingerprint" attribute, is the most obvious example. The
          gateway is probably better off either ending such calls early or
          acting as a full B2BUA (as defined in <xref target="RFC3261"/>) with
          media gatewaying.</t>

          <t>If a correspondent makes an offer using features that are not
          required by JSEP, these may not be understood by the WebRTC
          implementation. The gateway may choose to strip out some such
          features.</t>

          <t>Certain ancient practices (such as using port 0 to place a media
          section on hold with the intent of resuming it later) are not
          conformant with the SDP offer/answer spec (<xref target="RFC3264"/>
          section 8.2). Since WebRTC implementations are expected to be SDP
          offer/answer conformant, such practices may need to be stripped out
          by the gateway</t>
        </list>[NOTE IN DRAFT: This section may need expanding.]</t>

      <t/>
    </section>

    <section anchor="IANA" title="IANA Considerations">
      <t>This document makes no request of IANA.</t>

      <t>Note to RFC Editor: this section may be removed on publication as an
      RFC.</t>
    </section>

    <section anchor="Security" title="Security Considerations">
      <t>A WebRTC gateway may operate in two security modes: Security-context
      termination and security-context relaying.</t>

      <t>Relaying is only possible where signed and encrypted content can be
      passed through unchanged, and where keys can be exchanged directly
      between the endpoints.</t>

      <t>When the gateway terminates the security context, it means that the
      WebRTC user has to place trust in the gateway to perform all
      verification of identity and protection of content in the realm on the
      other side of the gateway; there is no way the end-user can detect a
      man-in-the-middle attack, an identity spoofing attack or a recording
      done at the gateway. For many scenarios, this is not going to be seen as
      a problem, but needs to be considered when one decides to use a
      gatewayed service.</t>
    </section>

    <section anchor="Acknowledgements" title="Acknowledgements">
      <t>Several comments from Christer Holmberg and Andrew Hutton were
      included.</t>
    </section>

    <section title="Change history">
      <t>Changes from draft-alvestrand-rtcweb-gateways-00 <list
          style="symbols">
          <t>Aligned terminology with draft-rtcweb-overview-12</t>

          <t>Rewrote text on signaling to improve clarity</t>

          <t>Editorial nits</t>
        </list></t>

      <t>Changes from draft-alvestrand-rtcweb-gateways-01 <list
          style="symbols">
          <t>Aligned terminology with draft-rtcweb-overview-13
          ("non-browser")</t>

          <t>Nits</t>
        </list></t>

      <t>Changes from draft-alvestrand-rtcweb-gateways-02 <list
          style="symbols">
          <t>Re-submitted as WG draft</t>

          <t>Addressed a comment from Andrew Hutton that deployment in open
          internet is an option, not a fact.</t>
        </list>Changes from draft-ietf-rtcweb-gateways-00</t>

      <t><list style="symbols">
          <t>Added note about impllications of non-support of BUNDLE</t>

          <t>Added "Considerations for SDP-using networks" section</t>
        </list>Changes from draft-ietf-rtcweb-gateways-01: None, this is a
      keepalive update.</t>
    </section>
  </middle>

  <back>
    <references title="Normative References">
      <?rfc include="reference.RFC.2119"?>

      <?rfc include='reference.I-D.ietf-rtcweb-jsep'?>

      <?rfc include='reference.I-D.ietf-rtcweb-overview'?>
    </references>

    <references title="Informative References">
      <?rfc include='reference.RFC.3261'?>

      <?rfc include='reference.RFC.3264'?>
    </references>
  </back>
</rfc>