rtcweb-overview.xml

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<?rfc toc="yes"?>
<?rfc tocompact="yes"?>
<?rfc tocdepth="3"?>
<?rfc tocindent="yes"?>
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<rfc category="std" docName="draft-ietf-rtcweb-overview-19" ipr="trust200902">
  <front>
    <title abbrev="WebRTC Overview">Overview: Real Time Protocols for
    Browser-based Applications</title>

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

      <address>
        <postal>
          <street>Kungsbron 2</street>

          <city>Stockholm</city>

          <region/>

          <code>11122</code>

          <country>Sweden</country>
        </postal>

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

    <date day="12" month="November" year="2017"/>

    <abstract>
      <t>This document gives an overview and context of a protocol suite
      intended for use with real-time applications that can be deployed in
      browsers - "real time communication on the Web".</t>

      <t>It intends to serve as a starting and coordination point to make sure
      all the parts that are needed to achieve this goal are findable, and
      that the parts that belong in the Internet protocol suite are fully
      specified and on the right publication track.</t>

      <t>This document is an Applicability Statement - it does not itself
      specify any protocol, but specifies which other specifications WebRTC
      compliant implementations are supposed to follow.</t>

      <t>This document is a work item of the RTCWEB working group.</t>
    </abstract>
  </front>

  <middle>
    <section title="Introduction">
      <t>The Internet was, from very early in its lifetime, considered a
      possible vehicle for the deployment of real-time, interactive
      applications - with the most easily imaginable being audio conversations
      (aka "Internet telephony") and video conferencing.</t>

      <t>The first attempts to build this were dependent on special networks,
      special hardware and custom-built software, often at very high prices or
      at low quality, placing great demands on the infrastructure.</t>

      <t>As the available bandwidth has increased, and as processors and other
      hardware has become ever faster, the barriers to participation have
      decreased, and it has become possible to deliver a satisfactory
      experience on commonly available computing hardware.</t>

      <t>Still, there are a number of barriers to the ability to communicate
      universally - one of these is that there is, as of yet, no single set of
      communication protocols that all agree should be made available for
      communication; another is the sheer lack of universal identification
      systems (such as is served by telephone numbers or email addresses in
      other communications systems).</t>

      <t>Development of The Universal Solution has, however, proved hard.</t>

      <t>The last few years have also seen a new platform rise for deployment
      of services: The browser-embedded application, or "Web application". It
      turns out that as long as the browser platform has the necessary
      interfaces, it is possible to deliver almost any kind of service on
      it.</t>

      <t>Traditionally, these interfaces have been delivered by plugins, which
      had to be downloaded and installed separately from the browser; in the
      development of HTML5, application developers see much promise in the
      possibility of making those interfaces available in a standardized way
      within the browser.</t>

      <t>This memo describes a set of building blocks that can be made
      accessible and controllable through a Javascript API in a browser, and
      which together form a sufficient set of functions to allow the use of
      interactive audio and video in applications that communicate directly
      between browsers across the Internet. The resulting protocol suite is
      intended to enable all the applications that are described as required
      scenarios in the use cases document <xref target="RFC7478"/>.</t>

      <t>Other efforts, for instance the W3C Web Real-Time Communications,
      Web Applications Security, and Device and Sensor working groups, focus
      on making standardized APIs and interfaces available, within or
      alongside the HTML5 effort, for those functions.  This memo concentrates
      on specifying the protocols and subprotocols that are needed to specify
      the interactions over the network.</t>

      <t>Operators should note that deployment of WebRTC will result in a
      change in the nature of signaling for real time media on the network,
      and may result in a shift in the kinds of devices used to create and
      consume such media. In the case of signaling, WebRTC session setup
      will typically occur over TLS-secured web technologies using
      application-specific protocols.  Operational techniques that involve
      inserting network elements to interpret SDP -- either through endpoint
      cooperation <xref target="RFC3361"/> or through the transparent
      insertion of SIP Application Level Gateways (ALGs) -- will not work
      with such signaling. In the case of networks using cooperative
      endpoints, the approaches defined in <xref target="RFC8155"/> may serve
      as a suitable replacement for <xref target="RFC3361"/>. The increase in
      browser-based communications may also lead to a shift away from
      dedicated real-time-communications hardware, such as SIP
      desk phones. This will diminish the efficacy of operational
      techniques that place dedicated real-time devices on their own
      network segment, address range, or VLAN for purposes such as
      applying traffic filtering and QoS. Applying the markings
      described in <xref target="I-D.ietf-tsvwg-rtcweb-qos"/> may be
      appropriate replacements for such techniques.</t> 

      <t>This memo uses the term "WebRTC" (note the case used) to refer to the
      overall effort consisting of both IETF and W3C efforts.</t>
    </section>

    <section title="Principles and Terminology">
      <t/>

      <section title="Goals of this document">
        <t>The goal of the WebRTC protocol specification is to specify a set
        of protocols that, if all are implemented, will allow an
        implementation to communicate with another implementation using audio,
        video and data sent along the most direct possible path between the
        participants.</t>

        <t>This document is intended to serve as the roadmap to the WebRTC
        specifications. It defines terms used by other parts of the WebRTC
        protocol specifications, lists references to other specifications that
        don't need further elaboration in the WebRTC context, and gives
        pointers to other documents that form part of the WebRTC suite.</t>

        <t>By reading this document and the documents it refers to, it should
        be possible to have all information needed to implement a WebRTC
        compatible implementation.</t>
      </section>

      <section title="Relationship between API and protocol">
        <t>The total WebRTC effort consists of two major parts, each
        consisting of multiple documents:</t>

        <t><list style="symbols">
            <t>A protocol specification, done in the IETF</t>

            <t>A Javascript API specification, defined in a series of W3C
            documents <xref target="W3C.WD-webrtc-20120209"/><xref
            target="W3C.WD-mediacapture-streams-20120628"/></t>
          </list>Together, these two specifications aim to provide an
        environment where Javascript embedded in any page, when suitably
        authorized by its user, is able to set up communication using audio,
        video and auxiliary data, as long as the browser supports this
        specification. The browser environment does not constrain the types of
        application in which this functionality can be used.</t>

        <t>The protocol specification does not assume that all implementations
        implement this API; it is not intended to be necessary for
        interoperation to know whether the entity one is communicating with is
        a browser or another device implementing this specification.</t>

        <t>The goal of cooperation between the protocol specification and the
        API specification is that for all options and features of the protocol
        specification, it should be clear which API calls to make to exercise
        that option or feature; similarly, for any sequence of API calls, it
        should be clear which protocol options and features will be invoked.
        Both subject to constraints of the implementation, of course.</t>

        <t>The following terms are used across the documents specifying the
        WebRTC suite, in the specific meanings given here. Not all terms are
        used in this document. Other terms are used in their commonly used
        meaning.</t>

        <t><list style="hanging">
            <t hangText="Agent:">Undefined term. See "SDP Agent" and "ICE
            Agent".</t>
             
            <t hangText="Application Programming Interface (API):">A
            specification of a set of calls and events, usually tied to a
            programming language or an abstract formal specification such as
            WebIDL, with its defined semantics.</t>
              
            <t hangText="Browser:">Used synonymously with "Interactive User
            Agent" as defined in the HTML specification <xref
            target="W3C.WD-html5-20110525"/>. See also "WebRTC User
            Agent".</t>
              
            <t hangText="Data Channel:">An abstraction that allows data to be
            sent between WebRTC endpoints in the form of messages. Two
            endpoints can have multiple data channels between them.</t>
              
            <t hangText="ICE Agent:">An implementation of the Interactive
            Connectivity Establishment (ICE) <xref
            target="RFC5245"/> protocol. An ICE Agent may also
            be an SDP Agent, but there exist ICE Agents that do not use SDP
            (for instance those that use Jingle <xref target="XEP-0166">
            </xref>).</t> 
              
            <t hangText="Interactive:">Communication between multiple parties,
            where the expectation is that an action from one party can cause a
            reaction by another party, and the reaction can be observed by the
            first party, with the total time required for the
            action/reaction/observation is on the order of no more than
            hundreds of milliseconds.</t>
              
            <t hangText="Media:">Audio and video content. Not to be confused
            with "transmission media" such as wires.</t>
              
            <t hangText="Media Path:">The path that media data follows from
            one WebRTC endpoint to another.</t>

            <t hangText="Protocol:">A specification of a set of data units,
            their representation, and rules for their transmission, with their
            defined semantics. A protocol is usually thought of as going
            between systems.</t>
 
            <t hangText="Real-time Media:">Media where generation of content
            and display of content are intended to occur closely together in
            time (on the order of no more than hundreds of milliseconds).
            Real-time media can be used to support interactive
            communication.</t>

            <t hangText="SDP Agent:">The protocol implementation involved in
            the Session Description Protocol (SDP) offer/answer exchange, as
            defined in <xref target="RFC3264"/> section 3.</t>

            <t hangText="Signaling:">Communication that happens in order to
            establish, manage and control media paths and data paths.</t>
 
            <t hangText="Signaling Path:">The communication channels used
            between entities participating in signaling to transfer signaling.
            There may be more entities in the signaling path than in the media
            path.</t>

            <t hangText="WebRTC Browser:">(also called a WebRTC User Agent
            or WebRTC UA) Something that conforms to both the protocol
            specification and the Javascript API cited above.</t>

            <t hangText="WebRTC non-Browser:"> Something that conforms to
            the protocol specification, but does not claim to implement the
            Javascript API.  This can also be called a "WebRTC device" or
            "WebRTC native application".</t>

            <t hangText="WebRTC Endpoint:"> Either a WebRTC browser or a
            WebRTC non-browser. It conforms to the protocol specification.</t>

            <t hangText="WebRTC-compatible Endpoint:"> An endpoint that is able
            to successfully communicate with a WebRTC endpoint, but may fail to
            meet some requirements of a WebRTC endpoint. This may limit where
            in the network such an endpoint can be attached, or may limit the
            security guarantees that it offers to others. It is not
            constrained by this specification; when it is mentioned at all, it
            is to note the implications on WebRTC-compatible endpoints of the
            requirements placed on WebRTC endpoints.</t>

            <t hangText="WebRTC Gateway:"> A WebRTC-compatible endpoint that
            mediates media traffic to non-WebRTC entities.</t>
          </list>All WebRTC browsers are WebRTC endpoints, so any requirement
        on a WebRTC endpoint also applies to a WebRTC browser.</t>

        <t>A WebRTC non-browser may be capable of hosting applications in a
        similar way to the way in which a browser can host Javascript
        applications, typically by offering APIs in other languages. For
        instance it may be implemented as a library that offers a C++ API
        intended to be loaded into applications. In this case, similar
        security considerations as for Javascript may be needed; however,
        since such APIs are not defined or referenced here, this document
        cannot give any specific rules for those interfaces.</t>

        <t>WebRTC gateways are described in a separate document, <xref
        target="I-D.ietf-rtcweb-gateways"/>.</t>
      </section>

      <section title="On interoperability and innovation">
        <t>The "Mission statement of the IETF" <xref target="RFC3935"/> states
        that "The benefit of a standard to the Internet is in interoperability
        - that multiple products implementing a standard are able to work
        together in order to deliver valuable functions to the Internet's
        users."</t>

        <t>Communication on the Internet frequently occurs in two phases:</t>

        <t><list style="symbols">
            <t>Two parties communicate, through some mechanism, what
            functionality they both are able to support</t>

            <t>They use that shared communicative functionality to
            communicate, or, failing to find anything in common, give up on
            communication.</t>
          </list>There are often many choices that can be made for
        communicative functionality; the history of the Internet is rife with
        the proposal, standardization, implementation, and success or failure
        of many types of options, in all sorts of protocols.</t>

        <t>The goal of having a mandatory to implement function set is to
        prevent negotiation failure, not to preempt or prevent
        negotiation.</t>

        <t>The presence of a mandatory to implement function set serves as a
        strong changer of the marketplace of deployment - in that it gives a
        guarantee that, as long as you conform to a specification, and the
        other party is willing to accept communication at the base level of
        that specification, you can communicate successfully.</t>

        <t>The alternative, that is having no mandatory to implement, does
        not mean that you cannot communicate, it merely means that in order to
        be part of the communications partnership, you have to implement the
        standard "and then some".  The "and then some" is usually called a
        profile of some sort; in the version most antithetical to the Internet
        ethos, that "and then some" consists of having to use a specific
        vendor's product only.</t>
      </section>

      <section title="Terminology">
        <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"/>.</t>
      </section>
    </section>

    <section title="Architecture and Functionality groups">
     <t>For browser-based applications, the model for real-time support does
     not assume that the browser will contain all the functions needed for
     an application such as a telephone or a video conference.  The vision is
     that the browser will have the functions needed for a Web application,
     working in conjunction with its backend servers, to implement these
     functions.</t>

      <t>This means that two vital interfaces need specification: The
      protocols that browsers use to talk to each other, without any
      intervening servers, and the APIs that are offered for a Javascript
      application to take advantage of the browser's functionality.</t>

      <figure anchor="fig-browser-model" title="Browser Model">
        <artwork><![CDATA[

                                                                          
                     +------------------------+  On-the-wire              
                     |                        |  Protocols                
                     |      Servers           |--------->                 
                     |                        |                           
                     |                        |                           
                     +------------------------+                           
                                 ^                                        
                                 |                                        
                                 |                                        
                                 | HTTPS/                                  
                                 | WebSockets                             
                                 |                                              
                                 |                                        
                   +----------------------------+                         
                   |    Javascript/HTML/CSS     |                         
                   +----------------------------+                         
                Other  ^                 ^ RTC                             
                APIs   |                 | APIs                            
                   +---|-----------------|------+                         
                   |   |                 |      |                         
                   |                 +---------+|                         
                   |                 | Browser ||  On-the-wire            
                   | Browser         | RTC     ||  Protocols              
                   |                 | Function|----------->              
                   |                 |         ||                         
                   |                 |         ||                         
                   |                 +---------+|                         
                   +---------------------|------+                         
                                         |                                
                                         V                                
                                    Native OS Services                    
                                                                          
                                                                          
                                                                          
                                                                          
                                                                          
                                                                          

]]></artwork>
      </figure>

      <t>Note that HTTPS and WebSockets are also offered to the Javascript
      application through browser APIs.</t>

      <t>As for all protocol and API specifications, there is no restriction
      that the protocols can only be used to talk to another browser; since
      they are fully specified, any endpoint that implements the protocols
      faithfully should be able to interoperate with the application running
      in the browser.</t>

      <t>A commonly imagined model of deployment is the one depicted
      below.  In the figure below JS is Javascript.</t>

      <figure anchor="fig-webtrapezoid" title="Browser RTC Trapezoid">
        <artwork><![CDATA[                                                                  
                                                                          
             +-----------+             +-----------+                      
             |   Web     |             |   Web     |                      
             |           |  Signaling  |           |                      
             |           |-------------|           |                      
             |  Server   |   path      |  Server   |                      
             |           |             |           |                      
             +-----------+             +-----------+                      
                  /                           \                           
                 /                             \ Application-defined      
                /                               \ over       
               /                                 \ HTTPS/WebSockets                    
              /  Application-defined over         \                       
             /   HTTPS/WebSockets                  \                      
            /                                       \                     
      +-----------+                           +-----------+               
      |JS/HTML/CSS|                           |JS/HTML/CSS|               
      +-----------+                           +-----------+               
      +-----------+                           +-----------+               
      |           |                           |           |               
      |           |                           |           |               
      |  Browser  | ------------------------- |  Browser  |               
      |           |          Media path       |           |               
      |           |                           |           |               
      +-----------+                           +-----------+               
]]></artwork>
      </figure>

      <t>On this drawing, the critical part to note is that the media path
      ("low path") goes directly between the browsers, so it has to be
      conformant to the specifications of the WebRTC protocol suite; the
      signaling path ("high path") goes via servers that can modify, translate
      or manipulate the signals as needed.</t>

      <t>If the two Web servers are operated by different entities, the
      inter-server signaling mechanism needs to be agreed upon, either by
      standardization or by other means of agreement. Existing protocols
      (e.g. SIP <xref target="RFC3261"/> or XMPP <xref target="RFC6120"/>)
      could be used between servers, while either a standards-based or
      proprietary protocol could be used between the browser and the web
      server.</t>

      <t>For example, if both operators' servers implement SIP, SIP could be
      used for communication between servers, along with either a standardized
      signaling mechanism (e.g. SIP over WebSockets) or a proprietary
      signaling mechanism used between the application running in the browser
      and the web server. Similarly, if both operators' servers implement
      Extensible Messaging and Presence Protocol (XMPP), XMPP could be used
      for communication between XMPP servers, with either a standardized
      signaling mechanism (e.g. XMPP over WebSockets or BOSH <xref
      target="XEP-0124"/> or a proprietary signaling mechanism used between the
      application running in the browser and the web server.</t>

      <t>The choice of protocols for client-server and inter-server
      signalling, and definition of the translation between them, is outside
      the scope of the WebRTC protocol suite described in the document.</t>

      <t>The functionality groups that are needed in the browser can be
      specified, more or less from the bottom up, as:</t>

      <t><list style="symbols">
          <t>Data transport: such as TCP, UDP and the means to securely set up
          connections between entities, as well as the functions for deciding
          when to send data: congestion management, bandwidth estimation and
          so on.</t>

          <t>Data framing: RTP, SCTP, DTLS, and other data formats that serve
          as containers, and their functions for data confidentiality and
          integrity.</t>

          <t>Data formats: Codec specifications, format specifications and
          functionality specifications for the data passed between systems.
          Audio and video codecs, as well as formats for data and document
          sharing, belong in this category. In order to make use of data
          formats, a way to describe them, a session description, is
          needed.</t>

          <t>Connection management: Setting up connections, agreeing on data
          formats, changing data formats during the duration of a call; SDP,
          SIP, and Jingle/XMPP belong in this category.</t>

          <t>Presentation and control: What needs to happen in order to ensure
          that interactions behave in a non-surprising manner. This can
          include floor control, screen layout, voice activated image
          switching and other such functions - where part of the system
          require the cooperation between parties. XCON and Cisco/Tandberg's
          TIP were some attempts at specifying this kind of functionality;
          many applications have been built without standardized interfaces to
          these functions.</t>

          <t>Local system support functions: These are things that need not be
          specified uniformly, because each participant may choose to do these
          in a way of the participant's choosing, without affecting the bits
          on the wire in a way that others have to be cognizant of. Examples
          in this category include echo cancellation (some forms of it), local
          authentication and authorization mechanisms, OS access control and
          the ability to do local recording of conversations.</t>
        </list>Within each functionality group, it is important to preserve
      both freedom to innovate and the ability for global communication.
      Freedom to innovate is helped by doing the specification in terms of
      interfaces, not implementation; any implementation able to communicate
      according to the interfaces is a valid implementation. Ability to
      communicate globally is helped both by having core specifications be
      unencumbered by IPR issues and by having the formats and protocols be
      fully enough specified to allow for independent implementation.</t>

      <t>One can think of the three first groups as forming a "media transport
      infrastructure", and of the three last groups as forming a "media
      service". In many contexts, it makes sense to use a common specification
      for the media transport infrastructure, which can be embedded in
      browsers and accessed using standard interfaces, and "let a thousand
      flowers bloom" in the "media service" layer; to achieve interoperable
      services, however, at least the first five of the six groups need to be
      specified.</t>
    </section>

    <section anchor="ch-transport" title="Data transport">
      <t>Data transport refers to the sending and receiving of data over the
      network interfaces, the choice of network-layer addresses at each end of
      the communication, and the interaction with any intermediate entities
      that handle the data, but do not modify it (such as TURN relays).</t>

      <t>It includes necessary functions for congestion control,
      retransmission, and in-order delivery.</t>

      <t>WebRTC endpoints MUST implement the transport protocols described in
      <xref target="I-D.ietf-rtcweb-transports"/>.</t>
    </section>

    <section title="Data framing and securing">
      <t>The format for media transport is RTP <xref target="RFC3550"/>.
      Implementation of SRTP <xref target="RFC3711"/> is REQUIRED for all
      implementations.</t>

      <t>The detailed considerations for usage of functions from RTP and SRTP
      are given in <xref target="I-D.ietf-rtcweb-rtp-usage"/>. The security
      considerations for the WebRTC use case are in <xref
      target="I-D.ietf-rtcweb-security"/>, and the resulting security
      functions are described in <xref
      target="I-D.ietf-rtcweb-security-arch"/>.</t>

      <t>Considerations for the transfer of data that is not in RTP format is
      described in <xref target="I-D.ietf-rtcweb-data-channel"/>, and a
      supporting protocol for establishing individual data channels is
      described in <xref target="I-D.ietf-rtcweb-data-protocol"/>. WebRTC
      endpoints MUST implement these two specifications.</t>

      <t>WebRTC endpoints MUST implement <xref
      target="I-D.ietf-rtcweb-rtp-usage"/>, <xref
      target="I-D.ietf-rtcweb-security"/>, <xref
      target="I-D.ietf-rtcweb-security-arch"/>, and the requirements they
      include.</t>
    </section>

    <section anchor="ch-data" title="Data formats">
      <t>The intent of this specification is to allow each communications
      event to use the data formats that are best suited for that particular
      instance, where a format is supported by both sides of the connection.
      However, a minimum standard is greatly helpful in order to ensure that
      communication can be achieved. This document specifies a minimum
      baseline that will be supported by all implementations of this
      specification, and leaves further codecs to be included at the will of
      the implementor.</t>

      <t>WebRTC endpoints that support audio and/or video MUST implement the
      codecs and profiles required in <xref target="RFC7874"/> and <xref
      target="RFC7742"/>.</t>
    </section>

    <section title="Connection management">
      <t>The methods, mechanisms and requirements for setting up, negotiating
      and tearing down connections is a large subject, and one where it is
      desirable to have both interoperability and freedom to innovate.</t>

      <t>The following principles apply:</t>

      <t><list style="numbers">
          <t>The WebRTC media negotiations will be capable of representing the
          same SDP offer/answer semantics <xref target="RFC3264"/> that are
          used in SIP, in such a way that it is possible to build a
          signaling gateway between SIP and the WebRTC media negotiation.</t>

          <t>It will be possible to gateway between legacy SIP devices that
          support ICE and appropriate RTP / SDP mechanisms, codecs and
          security mechanisms without using a media gateway. A signaling
          gateway to convert between the signaling on the web side to the SIP
          signaling may be needed.</t>

          <t>When an SDP for a new codec is specified, no other standardization
          should be required for it to be possible to use that in the web
          browsers. Adding new codecs which might have new SDP parameters should
          not change the APIs between the browser and Javascript application. As
          soon as the browsers support the new codecs, old applications
          written before the codecs were specified should automatically be
          able to use the new codecs where appropriate with no changes to the
          JS applications.</t>
        </list>The particular choices made for WebRTC, and their implications
      for the API offered by a browser implementing WebRTC, are described in
      <xref target="I-D.ietf-rtcweb-jsep"/>.</t>

      <t>WebRTC browsers MUST implement <xref
      target="I-D.ietf-rtcweb-jsep"/>.</t>

      <t>WebRTC endpoints MUST implement the functions described in that
      document that relate to the network layer (e.g. Bundle <xref
      target="I-D.ietf-mmusic-sdp-bundle-negotiation"/>, RTCP-mux <xref
      target="RFC5761"/> and Trickle ICE <xref
      target="I-D.ietf-ice-trickle"/>), but do not need to support the API
      functionality described there.</t>
    </section>

    <section title="Presentation and control">
      <t>The most important part of control is the user's control over the
      browser's interaction with input/output devices and communications
      channels. It is important that the user have some way of figuring out
      where his audio, video or texting is being sent, for what purported
      reason, and what guarantees are made by the parties that form part of
      this control channel. This is largely a local function between the
      browser, the underlying operating system and the user interface; this is
      specified in the peer connection API <xref
      target="W3C.WD-webrtc-20120209"/>, and the media capture API <xref
      target="W3C.WD-mediacapture-streams-20120628"/>.</t>

      <t>WebRTC browsers MUST implement these two specifications.</t>
    </section>

    <section title="Local system support functions">
      <t>These are characterized by the fact that the quality of these
      functions strongly influence the user experience, but the exact
      algorithm does not need coordination. In some cases (for instance echo
      cancellation, as described below), the overall system definition may
      need to specify that the overall system needs to have some
      characteristics for which these facilities are useful, without requiring
      them to be implemented a certain way.</t>

      <t>Local functions include echo cancellation, volume control, camera
      management including focus, zoom, pan/tilt controls (if available), and
      more.</t>

      <t>One would want to see certain parts of the system conform to certain
      properties, for instance:</t>

      <t><list style="symbols">
          <t>Echo cancellation should be good enough to achieve the
          suppression of acoustical feedback loops below a perceptually
          noticeable level.</t>

          <t>Privacy concerns MUST be satisfied; for instance, if remote
          control of camera is offered, the APIs should be available to let
          the local participant figure out who's controlling the camera, and
          possibly decide to revoke the permission for camera usage.</t>

          <t>Automatic gain control, if present, should normalize a speaking
          voice into a reasonable dB range.</t>
        </list>The requirements on WebRTC systems with regard to audio
      processing are found in <xref target="RFC7874"/> and includes more
      guidance about echo cancellation and AGC; the proposed API for control
      of local devices are found in <xref
      target="W3C.WD-mediacapture-streams-20120628"/>.</t>

      <t>WebRTC endpoints MUST implement the processing functions in <xref
      target="RFC7874"/>. (Together with the requirement in <xref
      target="ch-data"/>, this means that WebRTC endpoints MUST implement the
      whole document.)</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>Security of the web-enabled real time communications comes in several
      pieces:</t>

      <t><list style="symbols">
          <t>Security of the components: The browsers, and other servers
          involved. The most target-rich environment here is probably the
          browser; the aim here should be that the introduction of these
          components introduces no additional vulnerability.</t>

          <t>Security of the communication channels: It should be easy for a
          participant to reassure himself of the security of his communication
          - by verifying the crypto parameters of the links he himself
          participates in, and to get reassurances from the other parties to
          the communication that they promise that appropriate measures are
          taken.</t>

          <t>Security of the partners' identity: verifying that the
          participants are who they say they are (when positive identification
          is appropriate), or that their identity cannot be uncovered (when
          anonymity is a goal of the application).</t>
        </list>The security analysis, and the requirements derived from that
      analysis, is contained in <xref target="I-D.ietf-rtcweb-security"/>.</t>

      <t>It is also important to read the security sections of <xref
      target="W3C.WD-mediacapture-streams-20120628"/> and <xref
      target="W3C.WD-webrtc-20120209"/>.</t>
    </section>

    <section anchor="Acknowledgements" title="Acknowledgements">
      <t>The number of people who have taken part in the discussions
      surrounding this draft are too numerous to list, or even to identify.
      The ones below have made special, identifiable contributions; this does
      not mean that others' contributions are less important.</t>

      <t>Thanks to Cary Bran, Cullen Jennings, Colin Perkins, Magnus
      Westerlund and Joerg Ott, who offered technical contributions on various
      versions of the draft.</t>

      <t>Thanks to Jonathan Rosenberg, Matthew Kaufman and others at Skype for
      the ASCII drawings in section 1.</t>

      <t>Thanks to Alissa Cooper, Bjoern Hoehrmann, Colin Perkins,
      Colton Shields, Eric Rescorla, Heath Matlock, Henry Sinnreich,
      Justin Uberti, Keith Drage, Magnus Westerlund, Olle E. Johansson,
      Sean Turner and Simon Leinen for document review.</t>
    </section>
  </middle>

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

      <?rfc include='reference.RFC.3550'?>

      <?rfc include='reference.RFC.3264'?>

      <?rfc include='reference.RFC.3711'?>

      <?rfc include='reference.RFC.5245'?>

      <?rfc include='reference.RFC.7742'?>

      <?rfc include='reference.RFC.7874'?>

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

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

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

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

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

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

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

      <?rfc include='reference.W3C.WD-webrtc-20120209'?>

      <?rfc include='reference.W3C.WD-mediacapture-streams-20120628'?>

      <?rfc ?>
    </references>

    <references title="Informative References">
      <?rfc include='reference.RFC.3935'?>

      <?rfc include='reference.RFC.3261'?>

      <?rfc include='reference.RFC.3361'?>

      <?rfc include='reference.RFC.5761'?>

      <?rfc include='reference.RFC.6120'?>

      <?rfc include='reference.RFC.7478'?>

      <?rfc include='reference.RFC.8155'?>

      <?rfc include='reference.W3C.WD-html5-20110525'?>

      <?rfc include='reference.I-D.ietf-ice-trickle'?>

      <?rfc include='reference.I-D.ietf-mmusic-sdp-bundle-negotiation'?>

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

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

      <reference anchor="XEP-0166">
        <front>
          <title>Jingle</title>

          <author fullname="Scott Ludwig" initials="S." surname="Ludwig">
            <organization/>

            <address>
              <email>scottlu@google.com</email>
            </address>
          </author>

          <author fullname="Joe Beda" initials="J." surname="Beda">
            <organization/>

            <address>
              <email>jbeda@google.com</email>
            </address>
          </author>

          <author fullname="Peter Saint-Andre" initials="P."
                  surname="Saint-Andre">
            <organization/>

            <address>
              <email>stpeter@jabber.org</email>
            </address>
          </author>

          <author fullname="Robert McQueen" initials="R." surname="McQueen">
            <organization/>

            <address>
              <email>robert.mcqueen@collabora.co.uk</email>
            </address>
          </author>

          <author fullname="Sean Egan" initials="S." surname="Egan">
            <organization/>

            <address>
              <email>seanegan@google.com</email>
            </address>
          </author>

          <author fullname="Joe Hildebrand" initials="J." surname="Hildebrand">
            <organization/>

            <address>
              <email>jhildebr@cisco.com</email>
            </address>
          </author>

          <date day="20" month="June" year="2007"/>
        </front>

        <seriesInfo name="XSF XEP" value="0166"/>

        <format target="http://xmpp.org/extensions/xep-0166.html" type="HTML"/>
      </reference>

      <reference anchor="XEP-0124">
         <front>                                                                                                               <title>BOSH</title>

           <author fullname="Ian Paterson" initials="I." surname="Paterson">
             <organization/>

            <address>
               <email>ian.paterson@clientside.co.uk</email>
             </address>
           </author>
 
           <author fullname="Dave Smith" initials="D." surname="Smith">
             <organization/>
 
             <address>
               <email>dizzyd@jabber.org</email>
             </address>
           </author>
 
           <author fullname="Peter Saint-Andre" initials="P."
                   surname="Saint-Andre">
             <organization/>
 
             <address>
               <email>stpeter@jabber.org</email>
             </address>
           </author>
 
           <author fullname="Jack Moffitt" initials="J." surname="Moffitt">
             <organization/>
 
             <address>
               <email>jack@chesspark.com</email>
             </address>
           </author>
 
           <author fullname="Lance Stout" initials="L." surname="Stout">
             <organization/>
 
             <address>
               <email>lance@andyet.com</email>
             </address>
           </author>
 
           <author fullname="Winifried Tilanus" initials="W." surname="Tilanus">
              <organization/>
  
              <address>
                <email>winfried@tilanus.com</email>
              </address>
            </author>
 
           <date day="16" month="November" year="2016"/>

         </front>
 
        <seriesInfo name="XSF XEP" value="0124"/>
 
         <format target="http://xmpp.org/extensions/xep-0124.html" type="HTML"/>
       </reference>

    </references>

    <section title="Change log">
      <t>This section may be deleted by the RFC Editor when preparing for
      publication.</t>

      <section title="Changes from draft-alvestrand-dispatch-rtcweb-datagram-00 to -01">
        <t>Added section "On interoperability and innovation"</t>

        <t>Added data confidentiality and integrity to the "data framing"
        layer</t>

        <t>Added congestion management requirements in the "data transport"
        layer section</t>

        <t>Changed need for non-media data from "question: do we need this?"
        to "Open issue: How do we do this?"</t>

        <t>Strengthened disclaimer that listed codecs are placeholders, not
        decisions.</t>

        <t>More details on why the "local system support functions" section is
        there.</t>
      </section>

      <section title="Changes from draft-alvestrand-dispatch-01 to draft-alvestrand-rtcweb-overview-00">
        <t>Added section on "Relationship between API and protocol"</t>

        <t>Added terminology section</t>

        <t>Mentioned congestion management as part of the "data transport"
        layer in the layer list</t>
      </section>

      <section title="Changes from draft-alvestrand-rtcweb-00 to -01">
        <t>Removed most technical content, and replaced with pointers to
        drafts as requested and identified by the RTCWEB WG chairs.</t>

        <t>Added content to acknowledgments section.</t>

        <t>Added change log.</t>

        <t>Spell-checked document.</t>
      </section>

      <section title="Changes from draft-alvestrand-rtcweb-overview-01 to draft-ietf-rtcweb-overview-00">
        <t>Changed draft name and document date.</t>

        <t>Removed unused references</t>
      </section>

      <section title="Changes from -00 to -01 of draft-ietf-rtcweb-overview">
        <t>Added architecture figures to section 2.</t>

        <t>Changed the description of "echo cancellation" under "local system
        support functions".</t>

        <t>Added a few more definitions.</t>
      </section>

      <section title="Changes from -01 to -02 of draft-ietf-rtcweb-overview">
        <t>Added pointers to use cases, security and rtp-usage drafts (now WG
        drafts).</t>

        <t>Changed description of SRTP from mandatory-to-use to
        mandatory-to-implement.</t>

        <t>Added the "3 principles of negotiation" to the connection
        management section.</t>

        <t>Added an explicit statement that ICE is required for both NAT and
        consent-to-receive.</t>
      </section>

      <section title="Changes from -02 to -03 of draft-ietf-rtcweb-overview">
        <t>Added references to a number of new drafts.</t>

        <t>Expanded the description text under the "trapezoid" drawing with
        some more text discussed on the list.</t>

        <t>Changed the "Connection management" sentence from "will be done
        using SDP offer/answer" to "will be capable of representing SDP
        offer/answer" - this seems more consistent with JSEP.</t>

        <t>Added "security mechanisms" to the things a non-gatewayed SIP
        devices must support in order to not need a media gateway.</t>

        <t>Added a definition for "browser".</t>
      </section>

      <section title="Changes from -03 to -04 of draft-ietf-rtcweb-overview">
        <t>Made introduction more normative.</t>

        <t>Several wording changes in response to review comments from EKR</t>

        <t>Added an appendix to hold references and notes that are not yet in
        a separate document.</t>
      </section>

      <section title="Changes from -04 to -05 of draft-ietf-rtcweb-overview">
        <t>Minor grammatical fixes. This is mainly a "keepalive" refresh.</t>
      </section>

      <section title="Changes from -05 to -06">
        <t>Clarifications in response to Last Call review comments. Inserted
        reference to draft-ietf-rtcweb-audio.</t>
      </section>

      <section title="Changes from -06 to -07">
        <t>Added a reference to the "unified plan" draft, and updated some
        references.</t>

        <t>Otherwise, it's a "keepalive" draft.</t>
      </section>

      <section title="Changes from -07 to -08">
        <t>Removed the appendix that detailed transports, and replaced it with
        a reference to draft-ietf-rtcweb-transports. Removed now-unused
        references.</t>
      </section>

      <section title="Changes from -08 to -09">
        <t>Added text to the Abstract indicating that the intended status is
        an Applicability Statement.</t>

        <t/>
      </section>

      <section title="Changes from -09 to -10">
        <t>Defined "WebRTC Browser" and "WebRTC device" as things that do, or
        don't, conform to the API.</t>

        <t>Updated reference to data-protocol draft</t>

        <t>Updated data formats to reference -rtcweb-audio- and not the
        expired -cbran draft.</t>

        <t>Deleted references to -unified-plan</t>

        <t>Deleted reference to -generic-idp (draft expired)</t>

        <t>Added notes on which referenced documents WebRTC browsers or
        devices MUST conform to.</t>

        <t>Added pointer to the security section of the API drafts.</t>
      </section>

      <section title="Changes from -10 to -11">
        <t>Added "WebRTC Gateway" as a third class of device, and referenced
        the doc describing them.</t>

        <t>Made a number of text clarifications in response to document
        reviews.</t>
      </section>

      <section title="Changes from -11 to -12">
        <t>Refined entity definitions to define "WebRTC endpoint" and
        "WebRTC-compatible endpoint".</t>

        <t>Changed remaining usage of the term "RTCWEB" to "WebRTC", including
        in the page header.</t>
      </section>

      <section title="Changes from -12 to -13">
        <t>Changed "WebRTC device" to be "WebRTC non-browser", per decision at
        IETF 91. This led to the need for "WebRTC endpoint" as the common
        label for both, and the usage of that term in the rest of the
        document.</t>

        <t>Added words about WebRTC APIs in languages other than
        Javascript.</t>

        <t>Referenced draft-ietf-rtcweb-video for video codecs to support.</t>
      </section>

      <section title="Changes from -13 to -14">
        <t>None. This is a "keepalive" update.</t>
      </section>

      <section title="Changes from -14 to -15">
        <t>Changed "gateways" reference to point to the WG document.</t>
      </section>

      <section title="Changes from -15 to -16">
        <t>None. This is a "keepalive" publication.</t>
      </section>

      <section title="Changes from -16 to -17">
        <t>Addressed review comments by Olle E. Johansson and Magnus
        Westerlund</t>
      </section>

      <section title="Changes from -17 to -18">
        <t>Addressed review comments from Sean Turner and Alissa Cooper</t>
      </section>
      <section title="Changes from -18 to -19">
	<t>A number of grammatical issues were fixed.</t>
	<t>Added note on operational impact of WebRTC.</t>
	<t>Unified all definitions into the definitions list.</t>
	<t>Added a reference for BOSH.</t>
	<t>Changed ICE reference from 5245bis to RFC 5245.</t>
      </section>
    </section>
  </back>
</rfc>