rfc-xmldiff draft-ietf-rtcweb-overview-19-KC.v2v3.xml draft-ietf-rtcweb-overview-19-KC.v2v3.spell.xml

<?xml version='1.0' encoding='utf-8'?>
<!DOCTYPE rfc SYSTEM "rfc2629-xhtml.ent">
<rfc xmlns:xi="http://www.w3.org/2001/XInclude" category="std" number="0000" ipr="trust200902" obsoletes="" updates="" consensus="true" submissionType="IETF" xml:lang="en" tocInclude="true" symRefs="true" sortRefs="true" version="3">
  <!-- xml2rfc v2v3 conversion 2.23.1 -->
  <front>
    <title abbrev="WebRTC Overview">Overview: Real Time Protocols for
    Browser-based Applications</title>
    <seriesInfo name="RFC" value="0000"/>
    <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 month="July" year="2019"/>
    <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 numbered="true" toc="default" anchor="Intro">
      <name>Introduction</name>
      <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" format="default"/>.</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" format="default"/> 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" format="default"/> may serve
      as a suitable replacement for <xref target="RFC3361" format="default"/>. 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" format="default"/> 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 numbered="true" toc="default">
      <name>Principles and Terminology</name>
      <section numbered="true" toc="default">
        <name>Goals of this document</name>
        <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 numbered="true" toc="default">
        <name>Relationship between API and protocol</name>
        <t>The total WebRTC effort consists of two major parts, each
        consisting of multiple documents:</t>
        <ul spacing="normal">
          <li>A protocol specification, done in the IETF</li>
          <li>A Javascript API specification, defined in a series of W3C
            documents <xref target="W3C.WD-webrtc-20120209" format="default"/><xref target="W3C.WD-mediacapture-streams-20120628" format="default"/></li>
        </ul>
        <t>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>

<dl newline="false" spacing="normal">
          <dt>Agent:</dt>
          <dd>Undefined term. See "SDP Agent" and "ICE
            Agent".</dd>
          <dt>Application Programming Interface (API):</dt>
          <dd>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.</dd>
          <dt>Browser:</dt>
          <dd>Used synonymously with "Interactive User
            Agent" as defined in the HTML specification <xref target="W3C.WD-html5-20110525" format="default"/>. See also "WebRTC User
            Agent".</dd>
          <dt>Data Channel:</dt>
          <dd>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.</dd>
          <dt>ICE Agent:</dt>
          <dd>An implementation of the Interactive
            Connectivity Establishment (ICE) <xref target="RFC5245" format="default"/> 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" format="default">
            </xref>).</dd>
          <dt>Interactive:</dt>
          <dd>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.</dd>
          <dt>Media:</dt>
          <dd>Audio and video content. Not to be confused
            with "transmission media" such as wires.</dd>
          <dt>Media Path:</dt>
          <dd>The path that media data follows from
            one WebRTC endpoint to another.</dd>
          <dt>Protocol:</dt>
          <dd>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.</dd>
          <dt>Real-time Media:</dt>
          <dd>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.</dd>
          <dt>SDP Agent:</dt>
          <dd>The protocol implementation involved in
            the Session Description Protocol (SDP) offer/answer exchange, as
            defined in <xref target="RFC3264" sectionFormat="comma" section="3"/></dd>
          <dt>Signaling:</dt>
          <dd>Communication that happens in order to
            establish, manage and control media paths and data paths.</dd>
          <dt>Signaling Path:</dt>
          <dd>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.</dd>
          <dt>WebRTC Browser:</dt>
          <dd>(also called a WebRTC User Agent
            or WebRTC UA) Something that conforms to both the protocol
            specification and the Javascript API cited above.</dd>
          <dt>WebRTC non-Browser:</dt>
          <dd> 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".</dd>
          <dt>WebRTC Endpoint:</dt>
          <dd> Either a WebRTC browser or a
            WebRTC non-browser. It conforms to the protocol specification.</dd>
          <dt>WebRTC-compatible Endpoint:</dt>
          <dd> 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.</dd>
          <dt>WebRTC Gateway:</dt>
          <dd> A WebRTC-compatible endpoint that
            mediates media traffic to non-WebRTC entities.</dd>
        </dl>


        <t>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" format="default"/>.</t>
      </section>
      <section numbered="true" toc="default">
        <name>On interoperability and innovation</name>
        <t>The "Mission statement of the IETF" <xref target="RFC3935" format="default"/> 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>
        <ul spacing="normal">
          <li>Two parties communicate, through some mechanism, what
            functionality they both are able to support</li>
          <li>They use that shared communicative functionality to
            communicate, or, failing to find anything in common, give up on
            communication.</li>
        </ul>
        <t>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 numbered="true" toc="default">
        <name>Terminology</name>
        <t>The key words "<bcp14>MUST</bcp14>", "<bcp14>MUST NOT</bcp14>",
    "<bcp14>REQUIRED</bcp14>", "<bcp14>SHALL</bcp14>", "<bcp14>SHALL
    NOT</bcp14>", "<bcp14>SHOULD</bcp14>", "<bcp14>SHOULD NOT</bcp14>",
    "<bcp14>RECOMMENDED</bcp14>", "<bcp14>NOT RECOMMENDED</bcp14>",
    "<bcp14>MAY</bcp14>", and "<bcp14>OPTIONAL</bcp14>" in this document are
    to be interpreted as described in BCP&nbsp;14 <xref target="RFC2119"/>
    <xref target="RFC8174"/> when, and only when, they appear in all capitals,
    as shown here. 
</t>
      </section>
    </section>
    <section numbered="true" toc="default">
      <name>Architecture and Functionality groups</name>
      <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">
        <name>Browser Model</name>
        <artwork name="" type="" align="left" alt=""><![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 <xref target="fig-webtrapezoid"/> below, JS is Javascript.</t>
      <figure anchor="fig-webtrapezoid">
        <name>Browser RTC Trapezoid</name>
        <artwork name="" type="" align="left" alt=""><![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" format="default"/> or XMPP <xref target="RFC6120" format="default"/>)
      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" format="default"/> 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>
      <ul spacing="normal">
        <li>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.</li>
        <li>Data framing: RTP, SCTP, DTLS, and other data formats that serve
          as containers, and their functions for data confidentiality and
          integrity.</li>
        <li>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.</li>
        <li>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.</li>
        <li>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.</li>
        <li>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.</li>
      </ul>
      <t>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" numbered="true" toc="default">
      <name>Data transport</name>
      <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 <bcp14>MUST</bcp14> implement the transport protocols described in
      <xref target="RFCZZZZ" format="default"/>.</t>
    </section>
    <section numbered="true" toc="default">
      <name>Data framing and securing</name>
      <t>The format for media transport is RTP <xref target="RFC3550" format="default"/>.
      Implementation of SRTP <xref target="RFC3711" format="default"/> is <bcp14>REQUIRED</bcp14> for all
      implementations.</t>
      <t>The detailed considerations for usage of functions from RTP and SRTP
      are given in <xref target="RFCAAAA" format="default"/>. The security
      considerations for the WebRTC use case are in <xref target="RFCYYYY" format="default"/>, and the resulting security
      functions are described in <xref target="RFCDDDD" format="default"/>.</t>
      <t>Considerations for the transfer of data that is not in RTP format is
      described in <xref target="RFCBBBB" format="default"/>, and a
      supporting protocol for establishing individual data channels is
      described in <xref target="RFCCCCC" format="default"/>. WebRTC
      endpoints <bcp14>MUST</bcp14> implement these two specifications.</t>
      <t>WebRTC endpoints <bcp14>MUST</bcp14> implement <xref target="RFCAAAA" format="default"/>, <xref target="RFCYYYY" format="default"/>, <xref target="RFCDDDD" format="default"/>, and the requirements they
      include.</t>
    </section>
    <section anchor="ch-data" numbered="true" toc="default">
      <name>Data formats</name>
      <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 <bcp14>MUST</bcp14> implement the
      codecs and profiles required in <xref target="RFC7874" format="default"/> and <xref target="RFC7742" format="default"/>.</t>
    </section>
    <section numbered="true" toc="default">
      <name>Connection management</name>
      <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>
      <ol spacing="normal" type="1">
        <li>The WebRTC media negotiations will be capable of representing the
          same SDP offer/answer semantics <xref target="RFC3264" format="default"/> 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.</li>
        <li>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.</li>
        <li>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.</li>
      </ol>
      <t>The particular choices made for WebRTC, and their implications
      for the API offered by a browser implementing WebRTC, are described in
      <xref target="RFCEEEE" format="default"/>.</t>
      <t>WebRTC browsers <bcp14>MUST</bcp14> implement <xref target="RFCEEEE" format="default"/>.</t>
      <t>WebRTC endpoints <bcp14>MUST</bcp14> 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" format="default"/>, RTCP-mux <xref target="RFC5761" format="default"/> and Trickle ICE <xref target="I-D.ietf-ice-trickle" format="default"/>), but do not need to support the API
      functionality described there.</t>
    </section>
    <section numbered="true" toc="default">
      <name>Presentation and control</name>
      <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" format="default"/>, and the media capture API <xref target="W3C.WD-mediacapture-streams-20120628" format="default"/>.</t>
      <t>WebRTC browsers <bcp14>MUST</bcp14> implement these two specifications.</t>
    </section>
    <section numbered="true" toc="default">
      <name>Local system support functions</name>
      <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>
      <ul spacing="normal">
        <li>Echo cancellation should be good enough to achieve the
          suppression of acoustical feedback loops below a perceptually
          noticeable level.</li>
        <li>Privacy concerns <bcp14>MUST</bcp14> 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.</li>
        <li>Automatic gain control, if present, should normalize a speaking
          voice into a reasonable dB range.</li>
      </ul>
      <t>The requirements on WebRTC systems with regard to audio
      processing are found in <xref target="RFC7874" format="default"/> 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" format="default"/>.</t>
      <t>WebRTC endpoints <bcp14>MUST</bcp14> implement the processing functions in <xref target="RFC7874" format="default"/>. (Together with the requirement in <xref target="ch-data" format="default"/>, this means that WebRTC endpoints <bcp14>MUST</bcp14> implement the
      whole document.)</t>
    </section>
    <section anchor="IANA" numbered="true" toc="default">
      <name>IANA Considerations</name>
      <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" numbered="true" toc="default">
      <name>Security Considerations</name>
      <t>Security of the web-enabled real time communications comes in several
      pieces:</t>
      <ul spacing="normal">
        <li>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.</li>
        <li>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.</li>
        <li>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).</li>
      </ul>
      <t>The security analysis, and the requirements derived from that
      analysis, is contained in <xref target="RFCYYYY" format="default"/>.</t>
      <t>It is also important to read the security sections of <xref target="W3C.WD-mediacapture-streams-20120628" format="default"/> and <xref target="W3C.WD-webrtc-20120209" format="default"/>.</t>
    </section>
    <section anchor="Acknowledgements" numbered="true" toc="default">
      <name>Acknowledgements</name>
      <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 <xref target="Intro"/>.</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>
      <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.3264.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.3550.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.5245.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.7742.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.7874.xml"/>

   
    
        <reference anchor="RFCYYYY" target="https://www.rfc-editor.org/info/rfcYYYY">
          <front>
            <title>Security Considerations for WebRTC</title>
            <seriesInfo name="RFC" value="YYYY"/>
            <seriesInfo name="DOI" value="10.17487/RFCYYYY"/>
            <author initials="E" surname="Rescorla" fullname="Eric Rescorla">
              <organization/>
            </author>
            <date month="August" year="2019"/>
            <abstract>
              <t>WebRTC is a protocol suite for use with real-time applications that can be deployed in browsers - "real time communication on the Web". This document defines the WebRTC threat model and analyzes the security threats of WebRTC in that model.</t>
            </abstract>
          </front>
        </reference>

        <reference anchor="RFCZZZZ" target="https://www.rfc-editor.org/info/rfcZZZZ">
          <front>
            <title>Transports for WebRTC</title>
            <seriesInfo name="RFC" value="ZZZZ"/>
            <seriesInfo name="DOI" value="10.17487/RFCZZZZ"/>
            <author initials="H" surname="Alvestrand" fullname="Harald Alvestrand">
              <organization/>
            </author>
            <date month="August" year="2019"/>
            <abstract>
              <t>This document describes the data transport protocols used by WebRTC, including the protocols used for interaction with intermediate boxes such as firewalls, relays and NAT boxes.</t>
            </abstract>
          </front>
        </reference>

        <reference anchor="RFCAAAA" target="https://www.rfc-editor.org/info/rfcAAAA">
          <front>
            <title>Web Real-Time Communication (WebRTC): Media Transport and Use of RTP</title>
            <seriesInfo name="RFC" value="AAAA"/>
            <seriesInfo name="DOI" value="10.17487/RFCAAAA"/>
            <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="Joerg Ott">
              <organization/>
            </author>
            <date month="August" year="2019"/>
            <abstract>
              <t>The Web Real-Time Communication (WebRTC) framework provides support for direct interactive rich communication using audio, video, text, collaboration, games, etc. between two peers' web-browsers.  This memo describes the media transport aspects of the WebRTC framework. It specifies how the Real-time Transport Protocol (RTP) is used in the WebRTC context, and gives requirements for which RTP features, profiles, and extensions need to be supported.</t>
            </abstract>
          </front>
        </reference>

        <reference anchor="RFCBBBB" target="https://www.rfc-editor.org/info/rfcBBBB">
          <front>
            <title>WebRTC Data Channels</title>
            <seriesInfo name="RFC" value="BBBB"/>
            <seriesInfo name="DOI" value="10.17487/RFCBBBB"/>
            <author initials="R" surname="Jesup" fullname="Randell Jesup">
              <organization/>
            </author>
            <author initials="S" surname="Loreto" fullname="Salvatore Loreto">
              <organization/>
            </author>
            <author initials="M" surname="Tuexen" fullname="Michael Tuexen">
              <organization/>
            </author>
            <date month="August" year="2019"/>
            <abstract>
              <t>The WebRTC framework specifies protocol support for direct interactive rich communication using audio, video, and data between two peers' web-browsers.  This document specifies the non-media data transport aspects of the WebRTC framework.  It provides an architectural overview of how the Stream Control Transmission Protocol (SCTP) is used in the WebRTC context as a generic transport service allowing WEB-browsers to exchange generic data from peer to peer.</t>
            </abstract>
          </front>
        </reference>

        <reference anchor="RFCCCCC" target="https://www.rfc-editor.org/info/rfcCCCC">
          <front>
            <title>WebRTC Data Channel Establishment Protocol</title>
             <seriesInfo name="RFC" value="CCCC"/>
            <seriesInfo name="DOI" value="10.17487/RFCCCCC"/>
            <author initials="R" surname="Jesup" fullname="Randell Jesup">
              <organization/>
            </author>
            <author initials="S" surname="Loreto" fullname="Salvatore Loreto">
              <organization/>
            </author>
            <author initials="M" surname="Tuexen" fullname="Michael Tuexen">
              <organization/>
            </author>
            <date month="August" year="2019"/>
            <abstract>
              <t>The WebRTC framework specifies protocol support for direct interactive rich communication using audio, video, and data between two peers' web-browsers.  This document specifies a simple protocol for establishing symmetric Data Channels between the peers.  It uses a two way handshake and allows sending of user data without waiting for the handshake to complete.</t>
            </abstract>
          </front>
        </reference>

        <reference anchor="RFCDDDD" target="https://www.rfc-editor.org/info/rfcDDDD">
          <front>
            <title>WebRTC Security Architecture</title>
            <seriesInfo name="RFC" value="DDDD"/>
            <seriesInfo name="DOI" value="10.17487/RFCDDDD"/>
            <author initials="E" surname="Rescorla" fullname="Eric Rescorla">
              <organization/>
            </author>
            <date month="August" year="2019"/>
            <abstract>
              <t>This document defines the security architecture for WebRTC, a protocol suite intended for use with real-time applications that can be deployed in browsers - "real time communication on the Web".</t>
            </abstract>
          </front>
        </reference>

        <reference anchor="RFCEEEE" target="https://www.rfc-editor.org/info/rfcEEEE">
          <front>
            <title>JavaScript Session Establishment Protocol</title>
            <seriesInfo name="RFC" value="EEEE"/>
            <seriesInfo name="DOI" value="10.17487/RFCEEEE"/>
            <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">
              <organization/>
            </author>
            <date month="August" year="2019"/>
            <abstract>
              <t>This document describes the mechanisms for allowing a JavaScript application to control the signaling plane of a multimedia session via the interface specified in the W3C RTCPeerConnection API, and discusses how this relates to existing signaling protocols.</t>
            </abstract>
          </front>
        </reference>



        <reference anchor="W3C.WD-webrtc-20120209" target="http://www.w3.org/TR/2012/WD-webrtc-20120209" xml:base="https://xml2rfc.tools.ietf.org/public/rfc/bibxml4/reference.W3C.WD-webrtc-20120209.xml">
          <front>
            <title>WebRTC 1.0: Real-time Communication Between Browsers</title>
            <seriesInfo name="World Wide Web Consortium WD" value="WD-webrtc-20120209"/>
            <author initials="A." surname="Bergkvist" fullname="Adam Bergkvist">
              <organization/>
            </author>
            <author initials="D." surname="Burnett" fullname="Daniel C. Burnett">
              <organization/>
            </author>
            <author initials="C." surname="Jennings" fullname="Cullen Jennings">
              <organization/>
            </author>
            <author initials="A." surname="Narayanan" fullname="Anant Narayanan">
              <organization/>
            </author>
            <date month="February" year="2012"/>
          </front>
        </reference>
        <reference anchor="W3C.WD-mediacapture-streams-20120628" target="http://www.w3.org/TR/2012/WD-mediacapture-streams-20120628" xml:base="https://xml2rfc.tools.ietf.org/public/rfc/bibxml4/reference.W3C.WD-mediacapture-streams-20120628.xml">
          <front>
            <title>Media Capture and Streams</title>
            <seriesInfo name="World Wide Web Consortium WD" value="WD-mediacapture-streams-20120628"/>
            <author initials="D." surname="Burnett" fullname="Daniel C. Burnett">
              <organization/>
            </author>
            <author initials="A." surname="Narayanan" fullname="Anant Narayanan">
              <organization/>
            </author>
            <date month="June" year="2012"/>
          </front>
        </reference>
      </references>
      <references>
        <name>Informative References</name>

<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.3361.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.3935.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.5761.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.6120.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.7478.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.8155.xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.8174.xml"/>

        
        <reference anchor="W3C.WD-html5-20110525" target="http://www.w3.org/TR/2011/WD-html5-20110525" xml:base="https://xml2rfc.tools.ietf.org/public/rfc/bibxml4/reference.W3C.WD-html5-20110525.xml">
          <front>
            <title>HTML5</title>
            <seriesInfo name="World Wide Web Consortium LastCall," value="WD-html5-20110525"/>
            <author initials="I." surname="Hickson" fullname="Ian Hickson">
              <organization/>
            </author>
            <date month="May" year="2011"/>
          </front>
        </reference>
        <reference anchor="I-D.ietf-ice-trickle">
          <front>
            <title>Trickle ICE: Incremental Provisioning of Candidates for the Interactive Connectivity Establishment (ICE) Protocol</title>
            <seriesInfo name="Work in Progress," value="draft-ietf-ice-trickle-21"/>
            <author initials="E" surname="Ivov" fullname="Emil Ivov">
              <organization/>
            </author>
            <author initials="E" surname="Rescorla" fullname="Eric Rescorla">
              <organization/>
            </author>
            <author initials="J" surname="Uberti" fullname="Justin Uberti">
              <organization/>
            </author>
            <author initials="P" surname="Saint-Andre" fullname="Peter Saint-Andre">
              <organization/>
            </author>
            <date month="April" year="2018"/>
            <abstract>
              <t>This document describes "Trickle ICE", an extension to the Interactive Connectivity Establishment (ICE) protocol that enables ICE agents to begin connectivity checks while they are still gathering candidates, by incrementally exchanging candidates over time instead of all at once.  This method can considerably accelerate the process of establishing a communication session.</t>
            </abstract>
          </front>
        </reference>

        <reference anchor="I-D.ietf-mmusic-sdp-bundle-negotiation">
          <front>
            <title>Negotiating Media Multiplexing Using the Session Description Protocol (SDP)</title>
            <seriesInfo name="Work in Progress," value="draft-ietf-mmusic-sdp-bundle-negotiation-54"/>
            <author initials="C" surname="Holmberg" fullname="Christer Holmberg">
              <organization/>
            </author>
            <author initials="H" surname="Alvestrand" fullname="Harald Alvestrand">
              <organization/>
            </author>
            <author initials="C" surname="Jennings" fullname="Cullen Jennings">
              <organization/>
            </author>
            <date month="December" year="2018"/>
            <abstract>
              <t>This specification defines a new Session Description Protocol (SDP) Grouping Framework extension, 'BUNDLE'.  The extension can be used with the SDP Offer/Answer mechanism to negotiate the usage of a single transport (5-tuple) for sending and receiving media described by multiple SDP media descriptions ("m=" sections).  Such transport is referred to as a BUNDLE transport, and the media is referred to as bundled media.  The "m=" sections that use the BUNDLE transport form a BUNDLE group.  This specification updates RFC 3264, to also allow assigning a zero port value to a "m=" section in cases where the media described by the "m=" section is not disabled or rejected.  This specification updates RFC 5888, to also allow an SDP 'group' attribute to contain an identification-tag that identifies a "m=" section with the port set to zero.  This specification defines a new RTP Control Protocol (RTCP) source description (SDES) item and a new RTP header extension that can be used to correlate bundled RTP/RTCP packets with their appropriate "m=" section.  This specification updates RFC 7941, by adding an exception, for the MID RTP header extension, to the requirement regarding protection of an SDES RTP header extension carrying an SDES item for the MID RTP header extension.</t>
            </abstract>
          </front>
        </reference>

        <reference anchor="I-D.ietf-rtcweb-gateways">
          <front>
            <title>WebRTC Gateways</title>
            <seriesInfo name="Work in Progress," value="draft-ietf-rtcweb-gateways-02"/>
            <author initials="H" surname="Alvestrand" fullname="Harald T. Alvestrand">
              <organization/>
            </author>
            <author initials="U" surname="Rauschenbach" fullname="Uwe Rauschenbach">
              <organization/>
            </author>
            <date month="January" year="2016"/>
            <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>
          </front>
        </reference>

        <reference anchor="I-D.ietf-tsvwg-rtcweb-qos" >
          <front>
            <title>DSCP Packet Markings for WebRTC QoS</title>
            <seriesInfo name="Work in Progress," value="draft-ietf-tsvwg-rtcweb-qos-18"/>
            <author initials="P" surname="Jones" fullname="Paul Jones">
              <organization/>
            </author>
            <author initials="S" surname="Dhesikan" fullname="Subha Dhesikan">
              <organization/>
            </author>
            <author initials="C" surname="Jennings" fullname="Cullen Jennings">
              <organization/>
            </author>
            <author initials="D" surname="Druta" fullname="Dan Druta">
              <organization/>
            </author>
            <date month="August" year="2016"/>
            <abstract>
              <t>Many networks, such as service provider and enterprise networks, can provide different forwarding treatments for individual packets based on Differentiated Services Code Point (DSCP) values on a per-hop basis.  This document provides the recommended DSCP values for web browsers to use for various classes of WebRTC traffic.</t>
            </abstract>
          </front>
        </reference>

        <reference anchor="XEP-0166" target="http://xmpp.org/extensions/xep-0166.html">
          <front>
            <title>Jingle</title>
            <seriesInfo name="XSF XEP" value="0166"/>
            <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 month="June" year="2007"/>
          </front>
        </reference>

        <reference anchor="XEP-0124" target="http://xmpp.org/extensions/xep-0124.html">
          <front>
            <title>BOSH</title>
            <seriesInfo name="XSF XEP" value="0124"/>
            <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 month="November" year="2016"/>
          </front>
        </reference>
      </references>
    </references>
  </back>
</rfc>

<reference anchor="RFC2119" target="https://www.rfc-editor.org/info/rfc2119" xml:base="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.2119.xml">
<front>
<title>Key words for use in RFCs to Indicate Requirement Levels</title>
<author initials="S." surname="Bradner" fullname="S. Bradner"><organization/></author>
<date year="1997" month="March"/>
<abstract><t>In many standards track documents several words are used to signify the requirements in the specification.  These words are often capitalized. This document defines these words as they should be interpreted in IETF documents.  This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements.</t></abstract>
</front>
<seriesInfo name="BCP" value="14"/>
<seriesInfo name="RFC" value="2119"/>
<seriesInfo name="DOI" value="10.17487/RFC2119"/>
</reference>

<reference anchor="RFC3264" target="https://www.rfc-editor.org/info/rfc3264" xml:base="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.3264.xml">
<front>
<title>An Offer/Answer Model with Session Description Protocol (SDP)</title>
<author initials="J." surname="Rosenberg" fullname="J. Rosenberg"><organization/></author>
<author initials="H." surname="Schulzrinne" fullname="H. Schulzrinne"><organization/></author>
<date year="2002" month="June"/>
<abstract><t>This document defines a mechanism by which two entities can make use of the Session Description Protocol (SDP) to arrive at a common view of a multimedia session between them.  In the model, one participant offers the other a description of the desired session from their perspective, and the other participant answers with the desired session from their perspective.  This offer/answer model is most useful in unicast sessions where information from both participants is needed for the complete view of the session.  The offer/answer model is used by protocols like the Session Initiation Protocol (SIP).  [STANDARDS-TRACK]</t></abstract>
</front>
<seriesInfo name="RFC" value="3264"/>
<seriesInfo name="DOI" value="10.17487/RFC3264"/>
</reference>

<reference anchor="RFC3550" target="https://www.rfc-editor.org/info/rfc3550" xml:base="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.3550.xml">
<front>
<title>RTP: A Transport Protocol for Real-Time Applications</title>
<author initials="H." surname="Schulzrinne" fullname="H. Schulzrinne"><organization/></author>
<author initials="S." surname="Casner" fullname="S. Casner"><organization/></author>
<author initials="R." surname="Frederick" fullname="R. Frederick"><organization/></author>
<author initials="V." surname="Jacobson" fullname="V. Jacobson"><organization/></author>
<date year="2003" month="July"/>
<abstract><t>This memorandum describes RTP, the real-time transport protocol.  RTP provides end-to-end network transport functions suitable for applications transmitting real-time data, such as audio, video or simulation data, over multicast or unicast network services.  RTP does not address resource reservation and does not guarantee quality-of- service for real-time services.  The data transport is augmented by a control protocol (RTCP) to allow monitoring of the data delivery in a manner scalable to large multicast networks, and to provide minimal control and identification functionality.  RTP and RTCP are designed to be independent of the underlying transport and network layers.  The protocol supports the use of RTP-level translators and mixers. Most of the text in this memorandum is identical to RFC 1889 which it obsoletes.  There are no changes in the packet formats on the wire, only changes to the rules and algorithms governing how the protocol is used. The biggest change is an enhancement to the scalable timer algorithm for calculating when to send RTCP packets in order to minimize transmission in excess of the intended rate when many participants join a session simultaneously.  [STANDARDS-TRACK]</t></abstract>
</front>
<seriesInfo name="STD" value="64"/>
<seriesInfo name="RFC" value="3550"/>
<seriesInfo name="DOI" value="10.17487/RFC3550"/>
</reference>

<reference anchor="RFC3711" target="https://www.rfc-editor.org/info/rfc3711" xml:base="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.3711.xml">
<front>
<title>The Secure Real-time Transport Protocol (SRTP)</title>
<author initials="M." surname="Baugher" fullname="M. Baugher"><organization/></author>
<author initials="D." surname="McGrew" fullname="D. McGrew"><organization/></author>
<author initials="M." surname="Naslund" fullname="M. Naslund"><organization/></author>
<author initials="E." surname="Carrara" fullname="E. Carrara"><organization/></author>
<author initials="K." surname="Norrman" fullname="K. Norrman"><organization/></author>
<date year="2004" month="March"/>
<abstract><t>This document describes the Secure Real-time Transport Protocol (SRTP), a profile of the Real-time Transport Protocol (RTP), which can provide confidentiality, message authentication, and replay protection to the RTP traffic and to the control traffic for RTP, the Real-time Transport Control Protocol (RTCP).   [STANDARDS-TRACK]</t></abstract>
</front>
<seriesInfo name="RFC" value="3711"/>
<seriesInfo name="DOI" value="10.17487/RFC3711"/>
</reference>

<?xml version='1.0' encoding='UTF-8'?>

<reference  anchor='RFC5245' target='https://www.rfc-editor.org/info/rfc5245'>
<front>
<title>Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols</title>
<author initials='J.' surname='Rosenberg' fullname='J. Rosenberg'><organization /></author>
<date year='2010' month='April' />
<abstract><t>This document describes a protocol for Network Address Translator (NAT) traversal for UDP-based multimedia sessions established with the offer/answer model.  This protocol is called Interactive Connectivity Establishment (ICE).  ICE makes use of the Session Traversal Utilities for NAT (STUN) protocol and its extension, Traversal Using Relay NAT (TURN).  ICE can be used by any protocol utilizing the offer/answer model, such as the Session Initiation Protocol (SIP).  [STANDARDS-TRACK]</t></abstract>
</front>
<seriesInfo name='RFC' value='5245'/>
<seriesInfo name='DOI' value='10.17487/RFC5245'/>
</reference>

<?xml version='1.0' encoding='UTF-8'?>

<reference  anchor='RFC7742' target='https://www.rfc-editor.org/info/rfc7742'>
<front>
<title>WebRTC Video Processing and Codec Requirements</title>
<author initials='A.B.' surname='Roach' fullname='A.B. Roach'><organization /></author>
<date year='2016' month='March' />
<abstract><t>This specification provides the requirements and considerations for WebRTC applications to send and receive video across a network.  It specifies the video processing that is required as well as video codecs and their parameters.</t></abstract>
</front>
<seriesInfo name='RFC' value='7742'/>
<seriesInfo name='DOI' value='10.17487/RFC7742'/>
</reference>

<?xml version='1.0' encoding='UTF-8'?>

<reference  anchor='RFC7874' target='https://www.rfc-editor.org/info/rfc7874'>
<front>
<title>WebRTC Audio Codec and Processing Requirements</title>
<author initials='JM.' surname='Valin' fullname='JM. Valin'><organization /></author>
<author initials='C.' surname='Bran' fullname='C. Bran'><organization /></author>
<date year='2016' month='May' />
<abstract><t>This document outlines the audio codec and processing requirements for WebRTC endpoints.</t></abstract>
</front>
<seriesInfo name='RFC' value='7874'/>
<seriesInfo name='DOI' value='10.17487/RFC7874'/>
</reference>

<?xml version='1.0' encoding='UTF-8'?>

<reference anchor='W3C.WD-webrtc-20120209'
           target='http://www.w3.org/TR/2012/WD-webrtc-20120209'>
<front>
<title>WebRTC 1.0: Real-time Communication Between Browsers</title>

<author initials='A.' surname='Bergkvist' fullname='Adam Bergkvist'>
    <organization />
</author>

<author initials='D.' surname='Burnett' fullname='Daniel C. Burnett'>
    <organization />
</author>

<author initials='C.' surname='Jennings' fullname='Cullen Jennings'>
    <organization />
</author>

<author initials='A.' surname='Narayanan' fullname='Anant Narayanan'>
    <organization />
</author>

<date month='February' day='9' year='2012' />
</front>

<seriesInfo name='World Wide Web Consortium WD' value='WD-webrtc-20120209' />
<format type='HTML' target='http://www.w3.org/TR/2012/WD-webrtc-20120209' />
</reference>

<?xml version='1.0' encoding='UTF-8'?>

<reference anchor='W3C.WD-mediacapture-streams-20120628'
           target='http://www.w3.org/TR/2012/WD-mediacapture-streams-20120628'>
<front>
<title>Media Capture and Streams</title>

<author initials='D.' surname='Burnett' fullname='Daniel C. Burnett'>
    <organization />
</author>

<author initials='A.' surname='Narayanan' fullname='Anant Narayanan'>
    <organization />
</author>

<date month='June' day='28' year='2012' />
</front>

<seriesInfo name='World Wide Web Consortium WD' value='WD-mediacapture-streams-20120628' />
<format type='HTML' target='http://www.w3.org/TR/2012/WD-mediacapture-streams-20120628' />
</reference>

<?xml version='1.0' encoding='UTF-8'?>

<reference  anchor='RFC3261' target='https://www.rfc-editor.org/info/rfc3261'>
<front>
<title>SIP: Session Initiation Protocol</title>
<author initials='J.' surname='Rosenberg' fullname='J. Rosenberg'><organization /></author>
<author initials='H.' surname='Schulzrinne' fullname='H. Schulzrinne'><organization /></author>
<author initials='G.' surname='Camarillo' fullname='G. Camarillo'><organization /></author>
<author initials='A.' surname='Johnston' fullname='A. Johnston'><organization /></author>
<author initials='J.' surname='Peterson' fullname='J. Peterson'><organization /></author>
<author initials='R.' surname='Sparks' fullname='R. Sparks'><organization /></author>
<author initials='M.' surname='Handley' fullname='M. Handley'><organization /></author>
<author initials='E.' surname='Schooler' fullname='E. Schooler'><organization /></author>
<date year='2002' month='June' />
<abstract><t>This document describes Session Initiation Protocol (SIP), an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants.  These sessions include Internet telephone calls, multimedia distribution, and multimedia conferences.  [STANDARDS-TRACK]</t></abstract>
</front>
<seriesInfo name='RFC' value='3261'/>
<seriesInfo name='DOI' value='10.17487/RFC3261'/>
</reference>

<?xml version='1.0' encoding='UTF-8'?>

<reference  anchor='RFC3361' target='https://www.rfc-editor.org/info/rfc3361'>
<front>
<title>Dynamic Host Configuration Protocol (DHCP-for-IPv4) Option for Session Initiation Protocol (SIP) Servers</title>
<author initials='H.' surname='Schulzrinne' fullname='H. Schulzrinne'><organization /></author>
<date year='2002' month='August' />
</front>
<seriesInfo name='RFC' value='3361'/>
<seriesInfo name='DOI' value='10.17487/RFC3361'/>
</reference>

<?xml version='1.0' encoding='UTF-8'?>

<reference  anchor='RFC3935' target='https://www.rfc-editor.org/info/rfc3935'>
<front>
<title>A Mission Statement for the IETF</title>
<author initials='H.' surname='Alvestrand' fullname='H. Alvestrand'><organization /></author>
<date year='2004' month='October' />
<abstract><t>This memo gives a mission statement for the IETF, tries to define the terms used in the statement sufficiently to make the mission statement understandable and useful, argues why the IETF needs a mission statement, and tries to capture some of the debate that led to this point.  This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements.</t></abstract>
</front>
<seriesInfo name='BCP' value='95'/>
<seriesInfo name='RFC' value='3935'/>
<seriesInfo name='DOI' value='10.17487/RFC3935'/>
</reference>

<?xml version='1.0' encoding='UTF-8'?>

<reference  anchor='RFC5761' target='https://www.rfc-editor.org/info/rfc5761'>
<front>
<title>Multiplexing RTP Data and Control Packets on a Single Port</title>
<author initials='C.' surname='Perkins' fullname='C. Perkins'><organization /></author>
<author initials='M.' surname='Westerlund' fullname='M. Westerlund'><organization /></author>
<date year='2010' month='April' />
<abstract><t>This memo discusses issues that arise when multiplexing RTP data packets and RTP Control Protocol (RTCP) packets on a single UDP port. It updates RFC 3550 and RFC 3551 to describe when such multiplexing is and is not appropriate, and it explains how the Session Description Protocol (SDP) can be used to signal multiplexed sessions.  [STANDARDS-TRACK]</t></abstract>
</front>
<seriesInfo name='RFC' value='5761'/>
<seriesInfo name='DOI' value='10.17487/RFC5761'/>
</reference>

<?xml version='1.0' encoding='UTF-8'?>

<reference  anchor='RFC6120' target='https://www.rfc-editor.org/info/rfc6120'>
<front>
<title>Extensible Messaging and Presence Protocol (XMPP): Core</title>
<author initials='P.' surname='Saint-Andre' fullname='P. Saint-Andre'><organization /></author>
<date year='2011' month='March' />
<abstract><t>The Extensible Messaging and Presence Protocol (XMPP) is an application profile of the Extensible Markup Language (XML) that enables the near-real-time exchange of structured yet extensible data between any two or more network entities.  This document defines XMPP's core protocol methods: setup and teardown of XML streams, channel encryption, authentication, error handling, and communication primitives for messaging, network availability (&quot;presence&quot;), and request-response interactions.  This document obsoletes RFC 3920.  [STANDARDS-TRACK]</t></abstract>
</front>
<seriesInfo name='RFC' value='6120'/>
<seriesInfo name='DOI' value='10.17487/RFC6120'/>
</reference>

<?xml version='1.0' encoding='UTF-8'?>

<reference  anchor='RFC7478' target='https://www.rfc-editor.org/info/rfc7478'>
<front>
<title>Web Real-Time Communication Use Cases and Requirements</title>
<author initials='C.' surname='Holmberg' fullname='C. Holmberg'><organization /></author>
<author initials='S.' surname='Hakansson' fullname='S. Hakansson'><organization /></author>
<author initials='G.' surname='Eriksson' fullname='G. Eriksson'><organization /></author>
<date year='2015' month='March' />
<abstract><t>This document describes web-based real-time communication use cases. Requirements on the browser functionality are derived from the use cases.</t><t>This document was developed in an initial phase of the work with rather minor updates at later stages.  It has not really served as a tool in deciding features or scope for the WG's efforts so far.  It is being published to record the early conclusions of the WG.  It will not be used as a set of rigid guidelines that specifications and implementations will be held to in the future.</t></abstract>
</front>
<seriesInfo name='RFC' value='7478'/>
<seriesInfo name='DOI' value='10.17487/RFC7478'/>
</reference>

<?xml version='1.0' encoding='UTF-8'?>

<reference  anchor='RFC8155' target='https://www.rfc-editor.org/info/rfc8155'>
<front>
<title>Traversal Using Relays around NAT (TURN) Server Auto Discovery</title>
<author initials='P.' surname='Patil' fullname='P. Patil'><organization /></author>
<author initials='T.' surname='Reddy' fullname='T. Reddy'><organization /></author>
<author initials='D.' surname='Wing' fullname='D. Wing'><organization /></author>
<date year='2017' month='April' />
<abstract><t>Current Traversal Using Relays around NAT (TURN) server discovery mechanisms are relatively static and limited to explicit configuration.  These are usually under the administrative control of the application or TURN service provider, and not the enterprise, ISP, or the network in which the client is located.  Enterprises and ISPs wishing to provide their own TURN servers need auto-discovery mechanisms that a TURN client could use with minimal or no configuration.  This document describes three such mechanisms for TURN server discovery.</t><t>This document updates RFC 5766 to relax the requirement for mutual authentication in certain cases.</t></abstract>
</front>
<seriesInfo name='RFC' value='8155'/>
<seriesInfo name='DOI' value='10.17487/RFC8155'/>
</reference>

<reference anchor="RFC8174" target="https://www.rfc-editor.org/info/rfc8174" xml:base="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.8174.xml">
<front>
<title>Ambiguity of Uppercase vs Lowercase in RFC 2119 Key Words</title>
<author initials="B." surname="Leiba" fullname="B. Leiba"><organization/></author>
<date year="2017" month="May"/>
<abstract><t>RFC 2119 specifies common key words that may be used in protocol  specifications.  This document aims to reduce the ambiguity by clarifying that only UPPERCASE usage of the key words have the  defined special meanings.</t></abstract>
</front>
<seriesInfo name="BCP" value="14"/>
<seriesInfo name="RFC" value="8174"/>
<seriesInfo name="DOI" value="10.17487/RFC8174"/>
</reference>

<?xml version='1.0' encoding='UTF-8'?>

<reference anchor='W3C.WD-html5-20110525'
           target='http://www.w3.org/TR/2011/WD-html5-20110525'>
<front>
<title>HTML5</title>

<author initials='I.' surname='Hickson' fullname='Ian Hickson'>
    <organization />
</author>

<date month='May' day='25' year='2011' />
</front>

<seriesInfo name='World Wide Web Consortium LastCall' value='WD-html5-20110525' />
<format type='HTML' target='http://www.w3.org/TR/2011/WD-html5-20110525' />
</reference>

<?xml version="1.0" encoding="utf-8"?>
<!DOCTYPE rfc SYSTEM "rfc2629-xhtml.ent">
<rfc category="std" number="0000" ipr="trust200902" obsoletes="" updates="" consensus="true" submissionType="IETF" xml:lang="en" tocInclude="true" symRefs="true" sortRefs="true" version="3">
- <-- xml2rfc v2v3 conversion 2.23.1 -->
- <front>
  - <title abbrev="WebRTC Overview">
    - Overview: Real Time Protocols for Browser-based Applications
    - </title>
  - <seriesInfo name="RFC" value="0000"/>
  - <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 month="July" year="2019"/>
  - <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 numbered="true" toc="default" anchor="Intro">
    - <name>
      - Introduction
      - </name>
    - <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 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 xmlns:xi="http://www.w3.org/2001/XInclude" target="RFC7478" format="default"/>.
      - </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 sub-protocols 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 xmlns:xi="http://www.w3.org/2001/XInclude" target="RFC3361" format="default"/> 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 xmlns:xi="http://www.w3.org/2001/XInclude" target="RFC8155" format="default"/> may serve as a suitable replacement for <xref xmlns:xi="http://www.w3.org/2001/XInclude" target="RFC3361" format="default"/>. 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 xmlns:xi="http://www.w3.org/2001/XInclude" target="I-D.ietf-tsvwg-rtcweb-qos" format="default"/> 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 numbered="true" toc="default">
    - <name>
      - Principles and Terminology
      - </name>
    - <section numbered="true" toc="default">
      - <name>
        
        Goals of this document
        
        </name>
      - <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 numbered="true" toc="default">
      - <name>
        
        Relationship between API and protocol
        
        </name>
      - <t>
        
        The total WebRTC effort consists of two major parts, each consisting of multiple documents:
        
        </t>
      - <ul spacing="normal">
        
        <li>
        
        A protocol specification, done in the IETF
        
        </li>
        
        <li>
        
        A Javascript JavaScript API specification, defined in a series of W3C documents <xref xmlns:xi="http://www.w3.org/2001/XInclude" target="W3C.WD-webrtc-20120209" format="default"/><xref xmlns:xi="http://www.w3.org/2001/XInclude" target="W3C.WD-mediacapture-streams-20120628" format="default"/>
        
        </li>
        
        </ul>
      - <t>
        
        Together, these two specifications aim to provide an environment where Javascript 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>
      - <dl newline="false" spacing="normal">
        
        <dt>
        
        Agent:
        
        </dt>
        
        <dd>
        
        Undefined term. See "SDP Agent" and "ICE Agent".
        
        </dd>
        
        <dt>
        
        Application Programming Interface (API):
        
        </dt>
        
        <dd>
        
        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.
        
        </dd>
        
        <dt>
        
        Browser:
        
        </dt>
        
        <dd>
        
        Used synonymously with "Interactive User Agent" as defined in the HTML specification <xref xmlns:xi="http://www.w3.org/2001/XInclude" target="W3C.WD-html5-20110525" format="default"/>. See also "WebRTC User Agent".
        
        </dd>
        
        <dt>
        
        Data Channel:
        
        </dt>
        
        <dd>
        
        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.
        
        </dd>
        
        <dt>
        
        ICE Agent:
        
        </dt>
        
        <dd>
        
        An implementation of the Interactive Connectivity Establishment (ICE) <xref xmlns:xi="http://www.w3.org/2001/XInclude" target="RFC5245" format="default"/> 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 xmlns:xi="http://www.w3.org/2001/XInclude" target="XEP-0166" format="default"></xref>).
        
        </dd>
        
        <dt>
        
        Interactive:
        
        </dt>
        
        <dd>
        
        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.
        
        </dd>
        
        <dt>
        
        Media:
        
        </dt>
        
        <dd>
        
        Audio and video content. Not to be confused with "transmission media" such as wires.
        
        </dd>
        
        <dt>
        
        Media Path:
        
        </dt>
        
        <dd>
        
        The path that media data follows from one WebRTC endpoint to another.
        
        </dd>
        
        <dt>
        
        Protocol:
        
        </dt>
        
        <dd>
        
        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.
        
        </dd>
        
        <dt>
        
        Real-time Media:
        
        </dt>
        
        <dd>
        
        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.
        
        </dd>
        
        <dt>
        
        SDP Agent:
        
        </dt>
        
        <dd>
        
        The protocol implementation involved in the Session Description Protocol (SDP) offer/answer exchange, as defined in <xref xmlns:xi="http://www.w3.org/2001/XInclude" target="RFC3264" sectionFormat="comma" section="3"/>
        
        </dd>
        
        <dt>
        
        Signaling:
        
        </dt>
        
        <dd>
        
        Communication that happens in order to establish, manage and control media paths and data paths.
        
        </dd>
        
        <dt>
        
        Signaling Path:
        
        </dt>
        
        <dd>
        
        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.
        
        </dd>
        
        <dt>
        
        WebRTC Browser:
        
        </dt>
        
        <dd>
        
        (also called a WebRTC User Agent or WebRTC UA) Something that conforms to both the protocol specification and the Javascript JavaScript API cited above.
        
        </dd>
        
        <dt>
        
        WebRTC non-Browser:
        
        </dt>
        
        <dd>
        
        Something that conforms to the protocol specification, but does not claim to implement the Javascript JavaScript API. This can also be called a "WebRTC device" or "WebRTC native application".
        
        </dd>
        
        <dt>
        
        WebRTC Endpoint:
        
        </dt>
        
        <dd>
        
        Either a WebRTC browser or a WebRTC non-browser. It conforms to the protocol specification.
        
        </dd>
        
        <dt>
        
        WebRTC-compatible Endpoint:
        
        </dt>
        
        <dd>
        
        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.
        
        </dd>
        
        <dt>
        
        WebRTC Gateway:
        
        </dt>
        
        <dd>
        
        A WebRTC-compatible endpoint that mediates media traffic to non-WebRTC entities.
        
        </dd>
        
        </dl>
      - <t>
        
        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 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 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 xmlns:xi="http://www.w3.org/2001/XInclude" target="I-D.ietf-rtcweb-gateways" format="default"/>.
        
        </t>
      - </section>
    - <section numbered="true" toc="default">
      - <name>
        
        On interoperability and innovation
        
        </name>
      - <t>
        
        The "Mission statement of the IETF" <xref xmlns:xi="http://www.w3.org/2001/XInclude" target="RFC3935" format="default"/> 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>
      - <ul spacing="normal">
        
        <li>
        
        Two parties communicate, through some mechanism, what functionality they both are able to support
        
        </li>
        
        <li>
        
        They use that shared communicative functionality to communicate, or, failing to find anything in common, give up on communication.
        
        </li>
        
        </ul>
      - <t>
        
        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 numbered="true" toc="default">
      - <name>
        
        Terminology
        
        </name>
      - <t>
        
        The key words "<bcp14 xmlns:xi="http://www.w3.org/2001/XInclude">MUST</bcp14>", "<bcp14 xmlns:xi="http://www.w3.org/2001/XInclude">MUST NOT</bcp14>", "<bcp14 xmlns:xi="http://www.w3.org/2001/XInclude">REQUIRED</bcp14>", "<bcp14 xmlns:xi="http://www.w3.org/2001/XInclude">SHALL</bcp14>", "<bcp14 xmlns:xi="http://www.w3.org/2001/XInclude">SHALL NOT</bcp14>", "<bcp14 xmlns:xi="http://www.w3.org/2001/XInclude">SHOULD</bcp14>", "<bcp14 xmlns:xi="http://www.w3.org/2001/XInclude">SHOULD NOT</bcp14>", "<bcp14 xmlns:xi="http://www.w3.org/2001/XInclude">RECOMMENDED</bcp14>", "<bcp14 xmlns:xi="http://www.w3.org/2001/XInclude">NOT RECOMMENDED</bcp14>", "<bcp14 xmlns:xi="http://www.w3.org/2001/XInclude">MAY</bcp14>", and "<bcp14 xmlns:xi="http://www.w3.org/2001/XInclude">OPTIONAL</bcp14>" in this document are to be interpreted as described in BCP 14 <xref xmlns:xi="http://www.w3.org/2001/XInclude" target="RFC2119"/> <xref xmlns:xi="http://www.w3.org/2001/XInclude" target="RFC8174"/> when, and only when, they appear in all capitals, as shown here.
        
        </t>
      - </section>
    - </section>
  - <section numbered="true" toc="default">
    - <name>
      - Architecture and Functionality groups
      - </name>
    - <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 JavaScript application to take advantage of the browser's functionality.
      - </t>
    - <figure anchor="fig-browser-model">
      - <name>
        
        Browser Model
        
        </name>
      - <artwork name="" type="" align="left" alt="">
        
        
                             +------------------------+  On-the-wire
                             |                        |  Protocols
                             |      Servers           |--------->
                             |                        |
                             |                        |
                             +------------------------+
                                         ^
                                         |
                                         |
                                         | HTTPS/
                                         | WebSockets
                                         |
                                         |
                           +----------------------------+
                           |    Javascript/HTML/CSS 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 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 <xref xmlns:xi="http://www.w3.org/2001/XInclude" target="fig-webtrapezoid"/> below, JS is Javascript. JavaScript.
      - </t>
    - <figure anchor="fig-webtrapezoid">
      - <name>
        
        Browser RTC Trapezoid
        
        </name>
      - <artwork name="" type="" align="left" alt="">
        
        
        
                     +-----------+             +-----------+
                     |   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 xmlns:xi="http://www.w3.org/2001/XInclude" target="RFC3261" format="default"/> or XMPP <xref xmlns:xi="http://www.w3.org/2001/XInclude" target="RFC6120" format="default"/>) 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 xmlns:xi="http://www.w3.org/2001/XInclude" target="XEP-0124" format="default"/> 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, signaling, 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>
    - <ul spacing="normal">
      - <li>
        
        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.
        
        </li>
      - <li>
        
        Data framing: RTP, SCTP, DTLS, and other data formats that serve as containers, and their functions for data confidentiality and integrity.
        
        </li>
      - <li>
        
        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.
        
        </li>
      - <li>
        
        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.
        
        </li>
      - <li>
        
        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.
        
        </li>
      - <li>
        
        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.
        
        </li>
      - </ul>
    - <t>
      - 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" numbered="true" toc="default">
    - <name>
      - Data transport
      - </name>
    - <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 <bcp14 xmlns:xi="http://www.w3.org/2001/XInclude">MUST</bcp14> implement the transport protocols described in <xref xmlns:xi="http://www.w3.org/2001/XInclude" target="RFCZZZZ" format="default"/>.
      - </t>
    - </section>
  - <section numbered="true" toc="default">
    - <name>
      - Data framing and securing
      - </name>
    - <t>
      - The format for media transport is RTP <xref xmlns:xi="http://www.w3.org/2001/XInclude" target="RFC3550" format="default"/>. Implementation of SRTP <xref xmlns:xi="http://www.w3.org/2001/XInclude" target="RFC3711" format="default"/> is <bcp14 xmlns:xi="http://www.w3.org/2001/XInclude">REQUIRED</bcp14> for all implementations.
      - </t>
    - <t>
      - The detailed considerations for usage of functions from RTP and SRTP are given in <xref xmlns:xi="http://www.w3.org/2001/XInclude" target="RFCAAAA" format="default"/>. The security considerations for the WebRTC use case are in <xref xmlns:xi="http://www.w3.org/2001/XInclude" target="RFCYYYY" format="default"/>, and the resulting security functions are described in <xref xmlns:xi="http://www.w3.org/2001/XInclude" target="RFCDDDD" format="default"/>.
      - </t>
    - <t>
      - Considerations for the transfer of data that is not in RTP format is described in <xref xmlns:xi="http://www.w3.org/2001/XInclude" target="RFCBBBB" format="default"/>, and a supporting protocol for establishing individual data channels is described in <xref xmlns:xi="http://www.w3.org/2001/XInclude" target="RFCCCCC" format="default"/>. WebRTC endpoints <bcp14 xmlns:xi="http://www.w3.org/2001/XInclude">MUST</bcp14> implement these two specifications.
      - </t>
    - <t>
      - WebRTC endpoints <bcp14 xmlns:xi="http://www.w3.org/2001/XInclude">MUST</bcp14> implement <xref xmlns:xi="http://www.w3.org/2001/XInclude" target="RFCAAAA" format="default"/>, <xref xmlns:xi="http://www.w3.org/2001/XInclude" target="RFCYYYY" format="default"/>, <xref xmlns:xi="http://www.w3.org/2001/XInclude" target="RFCDDDD" format="default"/>, and the requirements they include.
      - </t>
    - </section>
  - <section anchor="ch-data" numbered="true" toc="default">
    - <name>
      - Data formats
      - </name>
    - <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 <bcp14 xmlns:xi="http://www.w3.org/2001/XInclude">MUST</bcp14> implement the codecs and profiles required in <xref xmlns:xi="http://www.w3.org/2001/XInclude" target="RFC7874" format="default"/> and <xref xmlns:xi="http://www.w3.org/2001/XInclude" target="RFC7742" format="default"/>.
      - </t>
    - </section>
  - <section numbered="true" toc="default">
    - <name>
      - Connection management
      - </name>
    - <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>
    - <ol spacing="normal" type="1">
      - <li>
        
        The WebRTC media negotiations will be capable of representing the same SDP offer/answer semantics <xref xmlns:xi="http://www.w3.org/2001/XInclude" target="RFC3264" format="default"/> 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.
        
        </li>
      - <li>
        
        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.
        
        </li>
      - <li>
        
        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 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.
        
        </li>
      - </ol>
    - <t>
      - The particular choices made for WebRTC, and their implications for the API offered by a browser implementing WebRTC, are described in <xref xmlns:xi="http://www.w3.org/2001/XInclude" target="RFCEEEE" format="default"/>.
      - </t>
    - <t>
      - WebRTC browsers <bcp14 xmlns:xi="http://www.w3.org/2001/XInclude">MUST</bcp14> implement <xref xmlns:xi="http://www.w3.org/2001/XInclude" target="RFCEEEE" format="default"/>.
      - </t>
    - <t>
      - WebRTC endpoints <bcp14 xmlns:xi="http://www.w3.org/2001/XInclude">MUST</bcp14> implement the functions described in that document that relate to the network layer (e.g. Bundle <xref xmlns:xi="http://www.w3.org/2001/XInclude" target="I-D.ietf-mmusic-sdp-bundle-negotiation" format="default"/>, RTCP-mux <xref xmlns:xi="http://www.w3.org/2001/XInclude" target="RFC5761" format="default"/> and Trickle ICE <xref xmlns:xi="http://www.w3.org/2001/XInclude" target="I-D.ietf-ice-trickle" format="default"/>), but do not need to support the API functionality described there.
      - </t>
    - </section>
  - <section numbered="true" toc="default">
    - <name>
      - Presentation and control
      - </name>
    - <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 xmlns:xi="http://www.w3.org/2001/XInclude" target="W3C.WD-webrtc-20120209" format="default"/>, and the media capture API <xref xmlns:xi="http://www.w3.org/2001/XInclude" target="W3C.WD-mediacapture-streams-20120628" format="default"/>.
      - </t>
    - <t>
      - WebRTC browsers <bcp14 xmlns:xi="http://www.w3.org/2001/XInclude">MUST</bcp14> implement these two specifications.
      - </t>
    - </section>
  - <section numbered="true" toc="default">
    - <name>
      - Local system support functions
      - </name>
    - <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>
    - <ul spacing="normal">
      - <li>
        
        Echo cancellation should be good enough to achieve the suppression of acoustical feedback loops below a perceptually noticeable level.
        
        </li>
      - <li>
        
        Privacy concerns <bcp14 xmlns:xi="http://www.w3.org/2001/XInclude">MUST</bcp14> 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.
        
        </li>
      - <li>
        
        Automatic gain control, if present, should normalize a speaking voice into a reasonable dB range.
        
        </li>
      - </ul>
    - <t>
      - The requirements on WebRTC systems with regard to audio processing are found in <xref xmlns:xi="http://www.w3.org/2001/XInclude" target="RFC7874" format="default"/> and includes more guidance about echo cancellation and AGC; the proposed API for control of local devices are found in <xref xmlns:xi="http://www.w3.org/2001/XInclude" target="W3C.WD-mediacapture-streams-20120628" format="default"/>.
      - </t>
    - <t>
      - WebRTC endpoints <bcp14 xmlns:xi="http://www.w3.org/2001/XInclude">MUST</bcp14> implement the processing functions in <xref xmlns:xi="http://www.w3.org/2001/XInclude" target="RFC7874" format="default"/>. (Together with the requirement in <xref xmlns:xi="http://www.w3.org/2001/XInclude" target="ch-data" format="default"/>, this means that WebRTC endpoints <bcp14 xmlns:xi="http://www.w3.org/2001/XInclude">MUST</bcp14> implement the whole document.)
      - </t>
    - </section>
  - <section anchor="IANA" numbered="true" toc="default">
    - <name>
      - IANA Considerations
      - </name>
    - <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" numbered="true" toc="default">
    - <name>
      - Security Considerations
      - </name>
    - <t>
      - Security of the web-enabled real time communications comes in several pieces:
      - </t>
    - <ul spacing="normal">
      - <li>
        
        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.
        
        </li>
      - <li>
        
        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.
        
        </li>
      - <li>
        
        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).
        
        </li>
      - </ul>
    - <t>
      - The security analysis, and the requirements derived from that analysis, is contained in <xref xmlns:xi="http://www.w3.org/2001/XInclude" target="RFCYYYY" format="default"/>.
      - </t>
    - <t>
      - It is also important to read the security sections of <xref xmlns:xi="http://www.w3.org/2001/XInclude" target="W3C.WD-mediacapture-streams-20120628" format="default"/> and <xref xmlns:xi="http://www.w3.org/2001/XInclude" target="W3C.WD-webrtc-20120209" format="default"/>.
      - </t>
    - </section>
  - <section anchor="Acknowledgements" numbered="true" toc="default">
    - <name>
      - Acknowledgements
      - </name>
    - <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 <xref xmlns:xi="http://www.w3.org/2001/XInclude" target="Intro"/>.
      - </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>
    - <name>
      - References
      - </name>
    - <references>
      - <name>
        
        Normative References
        
        </name>
      - <reference anchor="RFC2119" target="https://www.rfc-editor.org/info/rfc2119" xml:base="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.2119.xml">
        
        <front>
        
        <title>
        
        Key words for use in RFCs to Indicate Requirement Levels
        
        </title>
        
        <author initials="S." surname="Bradner" fullname="S. Bradner">
        
        <organization/>
        
        </author>
        
        <date year="1997" month="March"/>
        
        <abstract>
        
        <t>
        
        In many standards track documents several words are used to signify the requirements in the specification. These words are often capitalized. This document defines these words as they should be interpreted in IETF documents. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements.
        
        </t>
        
        </abstract>
        
        </front>
        
        <seriesInfo name="BCP" value="14"/>
        
        <seriesInfo name="RFC" value="2119"/>
        
        <seriesInfo name="DOI" value="10.17487/RFC2119"/>
        
        </reference>
      - <reference anchor="RFC3264" target="https://www.rfc-editor.org/info/rfc3264" xml:base="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.3264.xml">
        
        <front>
        
        <title>
        
        An Offer/Answer Model with Session Description Protocol (SDP)
        
        </title>
        
        <author initials="J." surname="Rosenberg" fullname="J. Rosenberg">
        
        <organization/>
        
        </author>
        
        <author initials="H." surname="Schulzrinne" fullname="H. Schulzrinne">
        
        <organization/>
        
        </author>
        
        <date year="2002" month="June"/>
        
        <abstract>
        
        <t>
        
        This document defines a mechanism by which two entities can make use of the Session Description Protocol (SDP) to arrive at a common view of a multimedia session between them. In the model, one participant offers the other a description of the desired session from their perspective, and the other participant answers with the desired session from their perspective. This offer/answer model is most useful in unicast sessions where information from both participants is needed for the complete view of the session. The offer/answer model is used by protocols like the Session Initiation Protocol (SIP). [STANDARDS-TRACK]
        
        </t>
        
        </abstract>
        
        </front>
        
        <seriesInfo name="RFC" value="3264"/>
        
        <seriesInfo name="DOI" value="10.17487/RFC3264"/>
        
        </reference>
      - <reference anchor="RFC3550" target="https://www.rfc-editor.org/info/rfc3550" xml:base="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.3550.xml">
        
        <front>
        
        <title>
        
        RTP: A Transport Protocol for Real-Time Applications
        
        </title>
        
        <author initials="H." surname="Schulzrinne" fullname="H. Schulzrinne">
        
        <organization/>
        
        </author>
        
        <author initials="S." surname="Casner" fullname="S. Casner">
        
        <organization/>
        
        </author>
        
        <author initials="R." surname="Frederick" fullname="R. Frederick">
        
        <organization/>
        
        </author>
        
        <author initials="V." surname="Jacobson" fullname="V. Jacobson">
        
        <organization/>
        
        </author>
        
        <date year="2003" month="July"/>
        
        <abstract>
        
        <t>
        
        This memorandum describes RTP, the real-time transport protocol. RTP provides end-to-end network transport functions suitable for applications transmitting real-time data, such as audio, video or simulation data, over multicast or unicast network services. RTP does not address resource reservation and does not guarantee quality-of- service for real-time services. The data transport is augmented by a control protocol (RTCP) to allow monitoring of the data delivery in a manner scalable to large multicast networks, and to provide minimal control and identification functionality. RTP and RTCP are designed to be independent of the underlying transport and network layers. The protocol supports the use of RTP-level translators and mixers. Most of the text in this memorandum is identical to RFC 1889 which it obsoletes. There are no changes in the packet formats on the wire, only changes to the rules and algorithms governing how the protocol is used. The biggest change is an enhancement to the scalable timer algorithm for calculating when to send RTCP packets in order to minimize transmission in excess of the intended rate when many participants join a session simultaneously. [STANDARDS-TRACK]
        
        </t>
        
        </abstract>
        
        </front>
        
        <seriesInfo name="STD" value="64"/>
        
        <seriesInfo name="RFC" value="3550"/>
        
        <seriesInfo name="DOI" value="10.17487/RFC3550"/>
        
        </reference>
      - <reference anchor="RFC3711" target="https://www.rfc-editor.org/info/rfc3711" xml:base="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.3711.xml">
        
        <front>
        
        <title>
        
        The Secure Real-time Transport Protocol (SRTP)
        
        </title>
        
        <author initials="M." surname="Baugher" fullname="M. Baugher">
        
        <organization/>
        
        </author>
        
        <author initials="D." surname="McGrew" fullname="D. McGrew">
        
        <organization/>
        
        </author>
        
        <author initials="M." surname="Naslund" fullname="M. Naslund">
        
        <organization/>
        
        </author>
        
        <author initials="E." surname="Carrara" fullname="E. Carrara">
        
        <organization/>
        
        </author>
        
        <author initials="K." surname="Norrman" fullname="K. Norrman">
        
        <organization/>
        
        </author>
        
        <date year="2004" month="March"/>
        
        <abstract>
        
        <t>
        
        This document describes the Secure Real-time Transport Protocol (SRTP), a profile of the Real-time Transport Protocol (RTP), which can provide confidentiality, message authentication, and replay protection to the RTP traffic and to the control traffic for RTP, the Real-time Transport Control Protocol (RTCP). [STANDARDS-TRACK]
        
        </t>
        
        </abstract>
        
        </front>
        
        <seriesInfo name="RFC" value="3711"/>
        
        <seriesInfo name="DOI" value="10.17487/RFC3711"/>
        
        </reference>
      - <reference anchor="RFC5245" target="https://www.rfc-editor.org/info/rfc5245" xml:base="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.5245.xml" "https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.5245.xml" >
        
        <front>
        
        <title>
        
        Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols
        
        </title>
        
        <author initials="J." surname="Rosenberg" fullname="J. Rosenberg">
        
        <organization/>
        
        </author>
        
        <date year="2010" month="April"/>
        
        <abstract>
        
        <t>
        
        This document describes a protocol for Network Address Translator (NAT) traversal for UDP-based multimedia sessions established with the offer/answer model. This protocol is called Interactive Connectivity Establishment (ICE). ICE makes use of the Session Traversal Utilities for NAT (STUN) protocol and its extension, Traversal Using Relay NAT (TURN). ICE can be used by any protocol utilizing the offer/answer model, such as the Session Initiation Protocol (SIP). [STANDARDS-TRACK]
        
        </t>
        
        </abstract>
        
        </front>
        
        <seriesInfo name="RFC" value="5245"/>
        
        <seriesInfo name="DOI" value="10.17487/RFC5245"/>
        
        </reference>
      - <reference anchor="RFC7742" target="https://www.rfc-editor.org/info/rfc7742" xml:base="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.7742.xml" "https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.7742.xml" >
        
        <front>
        
        <title>
        
        WebRTC Video Processing and Codec Requirements
        
        </title>
        
        <author initials="A.B." surname="Roach" fullname="A.B. Roach">
        
        <organization/>
        
        </author>
        
        <date year="2016" month="March"/>
        
        <abstract>
        
        <t>
        
        This specification provides the requirements and considerations for WebRTC applications to send and receive video across a network. It specifies the video processing that is required as well as video codecs and their parameters.
        
        </t>
        
        </abstract>
        
        </front>
        
        <seriesInfo name="RFC" value="7742"/>
        
        <seriesInfo name="DOI" value="10.17487/RFC7742"/>
        
        </reference>
      - <reference anchor="RFC7874" target="https://www.rfc-editor.org/info/rfc7874" xml:base="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.7874.xml" "https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.7874.xml" >
        
        <front>
        
        <title>
        
        WebRTC Audio Codec and Processing Requirements
        
        </title>
        
        <author initials="JM." surname="Valin" fullname="JM. Valin">
        
        <organization/>
        
        </author>
        
        <author initials="C." surname="Bran" fullname="C. Bran">
        
        <organization/>
        
        </author>
        
        <date year="2016" month="May"/>
        
        <abstract>
        
        <t>
        
        This document outlines the audio codec and processing requirements for WebRTC endpoints.
        
        </t>
        
        </abstract>
        
        </front>
        
        <seriesInfo name="RFC" value="7874"/>
        
        <seriesInfo name="DOI" value="10.17487/RFC7874"/>
        
        </reference>
      - <reference anchor="RFCYYYY" target="https://www.rfc-editor.org/info/rfcYYYY">
        
        <front>
        
        <title>
        
        Security Considerations for WebRTC
        
        </title>
        
        <seriesInfo name="RFC" value="YYYY"/>
        
        <seriesInfo name="DOI" value="10.17487/RFCYYYY"/>
        
        <author initials="E" surname="Rescorla" fullname="Eric Rescorla">
        
        <organization/>
        
        </author>
        
        <date month="August" year="2019"/>
        
        <abstract>
        
        <t>
        
        WebRTC is a protocol suite for use with real-time applications that can be deployed in browsers - "real time communication on the Web". This document defines the WebRTC threat model and analyzes the security threats of WebRTC in that model.
        
        </t>
        
        </abstract>
        
        </front>
        
        </reference>
      - <reference anchor="RFCZZZZ" target="https://www.rfc-editor.org/info/rfcZZZZ">
        
        <front>
        
        <title>
        
        Transports for WebRTC
        
        </title>
        
        <seriesInfo name="RFC" value="ZZZZ"/>
        
        <seriesInfo name="DOI" value="10.17487/RFCZZZZ"/>
        
        <author initials="H" surname="Alvestrand" fullname="Harald Alvestrand">
        
        <organization/>
        
        </author>
        
        <date month="August" year="2019"/>
        
        <abstract>
        
        <t>
        
        This document describes the data transport protocols used by WebRTC, including the protocols used for interaction with intermediate boxes such as firewalls, relays and NAT boxes.
        
        </t>
        
        </abstract>
        
        </front>
        
        </reference>
      - <reference anchor="RFCAAAA" target="https://www.rfc-editor.org/info/rfcAAAA">
        
        <front>
        
        <title>
        
        Web Real-Time Communication (WebRTC): Media Transport and Use of RTP
        
        </title>
        
        <seriesInfo name="RFC" value="AAAA"/>
        
        <seriesInfo name="DOI" value="10.17487/RFCAAAA"/>
        
        <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="Joerg Ott">
        
        <organization/>
        
        </author>
        
        <date month="August" year="2019"/>
        
        <abstract>
        
        <t>
        
        The Web Real-Time Communication (WebRTC) framework provides support for direct interactive rich communication using audio, video, text, collaboration, games, etc. between two peers' web-browsers. This memo describes the media transport aspects of the WebRTC framework. It specifies how the Real-time Transport Protocol (RTP) is used in the WebRTC context, and gives requirements for which RTP features, profiles, and extensions need to be supported.
        
        </t>
        
        </abstract>
        
        </front>
        
        </reference>
      - <reference anchor="RFCBBBB" target="https://www.rfc-editor.org/info/rfcBBBB">
        
        <front>
        
        <title>
        
        WebRTC Data Channels
        
        </title>
        
        <seriesInfo name="RFC" value="BBBB"/>
        
        <seriesInfo name="DOI" value="10.17487/RFCBBBB"/>
        
        <author initials="R" surname="Jesup" fullname="Randell Jesup">
        
        <organization/>
        
        </author>
        
        <author initials="S" surname="Loreto" fullname="Salvatore Loreto">
        
        <organization/>
        
        </author>
        
        <author initials="M" surname="Tuexen" fullname="Michael Tuexen">
        
        <organization/>
        
        </author>
        
        <date month="August" year="2019"/>
        
        <abstract>
        
        <t>
        
        The WebRTC framework specifies protocol support for direct interactive rich communication using audio, video, and data between two peers' web-browsers. This document specifies the non-media data transport aspects of the WebRTC framework. It provides an architectural overview of how the Stream Control Transmission Protocol (SCTP) is used in the WebRTC context as a generic transport service allowing WEB-browsers to exchange generic data from peer to peer.
        
        </t>
        
        </abstract>
        
        </front>
        
        </reference>
      - <reference anchor="RFCCCCC" target="https://www.rfc-editor.org/info/rfcCCCC">
        
        <front>
        
        <title>
        
        WebRTC Data Channel Establishment Protocol
        
        </title>
        
        <seriesInfo name="RFC" value="CCCC"/>
        
        <seriesInfo name="DOI" value="10.17487/RFCCCCC"/>
        
        <author initials="R" surname="Jesup" fullname="Randell Jesup">
        
        <organization/>
        
        </author>
        
        <author initials="S" surname="Loreto" fullname="Salvatore Loreto">
        
        <organization/>
        
        </author>
        
        <author initials="M" surname="Tuexen" fullname="Michael Tuexen">
        
        <organization/>
        
        </author>
        
        <date month="August" year="2019"/>
        
        <abstract>
        
        <t>
        
        The WebRTC framework specifies protocol support for direct interactive rich communication using audio, video, and data between two peers' web-browsers. This document specifies a simple protocol for establishing symmetric Data Channels between the peers. It uses a two way handshake and allows sending of user data without waiting for the handshake to complete.
        
        </t>
        
        </abstract>
        
        </front>
        
        </reference>
      - <reference anchor="RFCDDDD" target="https://www.rfc-editor.org/info/rfcDDDD">
        
        <front>
        
        <title>
        
        WebRTC Security Architecture
        
        </title>
        
        <seriesInfo name="RFC" value="DDDD"/>
        
        <seriesInfo name="DOI" value="10.17487/RFCDDDD"/>
        
        <author initials="E" surname="Rescorla" fullname="Eric Rescorla">
        
        <organization/>
        
        </author>
        
        <date month="August" year="2019"/>
        
        <abstract>
        
        <t>
        
        This document defines the security architecture for WebRTC, a protocol suite intended for use with real-time applications that can be deployed in browsers - "real time communication on the Web".
        
        </t>
        
        </abstract>
        
        </front>
        
        </reference>
      - <reference anchor="RFCEEEE" target="https://www.rfc-editor.org/info/rfcEEEE">
        
        <front>
        
        <title>
        
        JavaScript Session Establishment Protocol
        
        </title>
        
        <seriesInfo name="RFC" value="EEEE"/>
        
        <seriesInfo name="DOI" value="10.17487/RFCEEEE"/>
        
        <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">
        
        <organization/>
        
        </author>
        
        <date month="August" year="2019"/>
        
        <abstract>
        
        <t>
        
        This document describes the mechanisms for allowing a JavaScript application to control the signaling plane of a multimedia session via the interface specified in the W3C RTCPeerConnection API, and discusses how this relates to existing signaling protocols.
        
        </t>
        
        </abstract>
        
        </front>
        
        </reference>
      - <reference anchor="W3C.WD-webrtc-20120209" target="http://www.w3.org/TR/2012/WD-webrtc-20120209" xml:base="https://xml2rfc.tools.ietf.org/public/rfc/bibxml4/reference.W3C.WD-webrtc-20120209.xml">
        
        <front>
        
        <title>
        
        WebRTC 1.0: Real-time Communication Between Browsers
        
        </title>
        
        <seriesInfo name="World Wide Web Consortium WD" value="WD-webrtc-20120209"/>
        
        <author initials="A." surname="Bergkvist" fullname="Adam Bergkvist">
        
        <organization/>
        
        </author>
        
        <author initials="D." surname="Burnett" fullname="Daniel C. Burnett">
        
        <organization/>
        
        </author>
        
        <author initials="C." surname="Jennings" fullname="Cullen Jennings">
        
        <organization/>
        
        </author>
        
        <author initials="A." surname="Narayanan" fullname="Anant Narayanan">
        
        <organization/>
        
        </author>
        
        <date month="February" year="2012"/>
        
        </front>
        
        </reference>
      - <reference anchor="W3C.WD-mediacapture-streams-20120628" target="http://www.w3.org/TR/2012/WD-mediacapture-streams-20120628" xml:base="https://xml2rfc.tools.ietf.org/public/rfc/bibxml4/reference.W3C.WD-mediacapture-streams-20120628.xml">
        
        <front>
        
        <title>
        
        Media Capture and Streams
        
        </title>
        
        <seriesInfo name="World Wide Web Consortium WD" value="WD-mediacapture-streams-20120628"/>
        
        <author initials="D." surname="Burnett" fullname="Daniel C. Burnett">
        
        <organization/>
        
        </author>
        
        <author initials="A." surname="Narayanan" fullname="Anant Narayanan">
        
        <organization/>
        
        </author>
        
        <date month="June" year="2012"/>
        
        </front>
        
        </reference>
      - </references>
    - <references>
      - <name>
        
        Informative References
        
        </name>
      - <reference anchor="RFC3261" target="https://www.rfc-editor.org/info/rfc3261" xml:base="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.3261.xml" "https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.3261.xml" >
        
        <front>
        
        <title>
        
        SIP: Session Initiation Protocol
        
        </title>
        
        <author initials="J." surname="Rosenberg" fullname="J. Rosenberg">
        
        <organization/>
        
        </author>
        
        <author initials="H." surname="Schulzrinne" fullname="H. Schulzrinne">
        
        <organization/>
        
        </author>
        
        <author initials="G." surname="Camarillo" fullname="G. Camarillo">
        
        <organization/>
        
        </author>
        
        <author initials="A." surname="Johnston" fullname="A. Johnston">
        
        <organization/>
        
        </author>
        
        <author initials="J." surname="Peterson" fullname="J. Peterson">
        
        <organization/>
        
        </author>
        
        <author initials="R." surname="Sparks" fullname="R. Sparks">
        
        <organization/>
        
        </author>
        
        <author initials="M." surname="Handley" fullname="M. Handley">
        
        <organization/>
        
        </author>
        
        <author initials="E." surname="Schooler" fullname="E. Schooler">
        
        <organization/>
        
        </author>
        
        <date year="2002" month="June"/>
        
        <abstract>
        
        <t>
        
        This document describes Session Initiation Protocol (SIP), an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants. These sessions include Internet telephone calls, multimedia distribution, and multimedia conferences. [STANDARDS-TRACK]
        
        </t>
        
        </abstract>
        
        </front>
        
        <seriesInfo name="RFC" value="3261"/>
        
        <seriesInfo name="DOI" value="10.17487/RFC3261"/>
        
        </reference>
      - <reference anchor="RFC3361" target="https://www.rfc-editor.org/info/rfc3361" xml:base="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.3361.xml" "https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.3361.xml" >
        
        <front>
        
        <title>
        
        Dynamic Host Configuration Protocol (DHCP-for-IPv4) Option for Session Initiation Protocol (SIP) Servers
        
        </title>
        
        <author initials="H." surname="Schulzrinne" fullname="H. Schulzrinne">
        
        <organization/>
        
        </author>
        
        <date year="2002" month="August"/>
        
        </front>
        
        <seriesInfo name="RFC" value="3361"/>
        
        <seriesInfo name="DOI" value="10.17487/RFC3361"/>
        
        </reference>
      - <reference anchor="RFC3935" target="https://www.rfc-editor.org/info/rfc3935" xml:base="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.3935.xml" "https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.3935.xml" >
        
        <front>
        
        <title>
        
        A Mission Statement for the IETF
        
        </title>
        
        <author initials="H." surname="Alvestrand" fullname="H. Alvestrand">
        
        <organization/>
        
        </author>
        
        <date year="2004" month="October"/>
        
        <abstract>
        
        <t>
        
        This memo gives a mission statement for the IETF, tries to define the terms used in the statement sufficiently to make the mission statement understandable and useful, argues why the IETF needs a mission statement, and tries to capture some of the debate that led to this point. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements.
        
        </t>
        
        </abstract>
        
        </front>
        
        <seriesInfo name="BCP" value="95"/>
        
        <seriesInfo name="RFC" value="3935"/>
        
        <seriesInfo name="DOI" value="10.17487/RFC3935"/>
        
        </reference>
      - <reference anchor="RFC5761" target="https://www.rfc-editor.org/info/rfc5761" xml:base="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.5761.xml" "https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.5761.xml" >
        
        <front>
        
        <title>
        
        Multiplexing RTP Data and Control Packets on a Single Port
        
        </title>
        
        <author initials="C." surname="Perkins" fullname="C. Perkins">
        
        <organization/>
        
        </author>
        
        <author initials="M." surname="Westerlund" fullname="M. Westerlund">
        
        <organization/>
        
        </author>
        
        <date year="2010" month="April"/>
        
        <abstract>
        
        <t>
        
        This memo discusses issues that arise when multiplexing RTP data packets and RTP Control Protocol (RTCP) packets on a single UDP port. It updates RFC 3550 and RFC 3551 to describe when such multiplexing is and is not appropriate, and it explains how the Session Description Protocol (SDP) can be used to signal multiplexed sessions. [STANDARDS-TRACK]
        
        </t>
        
        </abstract>
        
        </front>
        
        <seriesInfo name="RFC" value="5761"/>
        
        <seriesInfo name="DOI" value="10.17487/RFC5761"/>
        
        </reference>
      - <reference anchor="RFC6120" target="https://www.rfc-editor.org/info/rfc6120" xml:base="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.6120.xml" "https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.6120.xml" >
        
        <front>
        
        <title>
        
        Extensible Messaging and Presence Protocol (XMPP): Core
        
        </title>
        
        <author initials="P." surname="Saint-Andre" fullname="P. Saint-Andre">
        
        <organization/>
        
        </author>
        
        <date year="2011" month="March"/>
        
        <abstract>
        
        <t>
        
        The Extensible Messaging and Presence Protocol (XMPP) is an application profile of the Extensible Markup Language (XML) that enables the near-real-time exchange of structured yet extensible data between any two or more network entities. This document defines XMPP's core protocol methods: setup and teardown of XML streams, channel encryption, authentication, error handling, and communication primitives for messaging, network availability ("presence"), and request-response interactions. This document obsoletes RFC 3920. [STANDARDS-TRACK]
        
        </t>
        
        </abstract>
        
        </front>
        
        <seriesInfo name="RFC" value="6120"/>
        
        <seriesInfo name="DOI" value="10.17487/RFC6120"/>
        
        </reference>
      - <reference anchor="RFC7478" target="https://www.rfc-editor.org/info/rfc7478" xml:base="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.7478.xml" "https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.7478.xml" >
        
        <front>
        
        <title>
        
        Web Real-Time Communication Use Cases and Requirements
        
        </title>
        
        <author initials="C." surname="Holmberg" fullname="C. Holmberg">
        
        <organization/>
        
        </author>
        
        <author initials="S." surname="Hakansson" fullname="S. Hakansson">
        
        <organization/>
        
        </author>
        
        <author initials="G." surname="Eriksson" fullname="G. Eriksson">
        
        <organization/>
        
        </author>
        
        <date year="2015" month="March"/>
        
        <abstract>
        
        <t>
        
        This document describes web-based real-time communication use cases. Requirements on the browser functionality are derived from the use cases.
        
        </t>
        
        <t>
        
        This document was developed in an initial phase of the work with rather minor updates at later stages. It has not really served as a tool in deciding features or scope for the WG's efforts so far. It is being published to record the early conclusions of the WG. It will not be used as a set of rigid guidelines that specifications and implementations will be held to in the future.
        
        </t>
        
        </abstract>
        
        </front>
        
        <seriesInfo name="RFC" value="7478"/>
        
        <seriesInfo name="DOI" value="10.17487/RFC7478"/>
        
        </reference>
      - <reference anchor="RFC8155" target="https://www.rfc-editor.org/info/rfc8155" xml:base="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.8155.xml" "https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.8155.xml" >
        
        <front>
        
        <title>
        
        Traversal Using Relays around NAT (TURN) Server Auto Discovery
        
        </title>
        
        <author initials="P." surname="Patil" fullname="P. Patil">
        
        <organization/>
        
        </author>
        
        <author initials="T." surname="Reddy" fullname="T. Reddy">
        
        <organization/>
        
        </author>
        
        <author initials="D." surname="Wing" fullname="D. Wing">
        
        <organization/>
        
        </author>
        
        <date year="2017" month="April"/>
        
        <abstract>
        
        <t>
        
        Current Traversal Using Relays around NAT (TURN) server discovery mechanisms are relatively static and limited to explicit configuration. These are usually under the administrative control of the application or TURN service provider, and not the enterprise, ISP, or the network in which the client is located. Enterprises and ISPs wishing to provide their own TURN servers need auto-discovery mechanisms that a TURN client could use with minimal or no configuration. This document describes three such mechanisms for TURN server discovery.
        
        </t>
        
        <t>
        
        This document updates RFC 5766 to relax the requirement for mutual authentication in certain cases.
        
        </t>
        
        </abstract>
        
        </front>
        
        <seriesInfo name="RFC" value="8155"/>
        
        <seriesInfo name="DOI" value="10.17487/RFC8155"/>
        
        </reference>
      - <reference anchor="RFC8174" target="https://www.rfc-editor.org/info/rfc8174" xml:base="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.8174.xml">
        
        <front>
        
        <title>
        
        Ambiguity of Uppercase vs Lowercase in RFC 2119 Key Words
        
        </title>
        
        <author initials="B." surname="Leiba" fullname="B. Leiba">
        
        <organization/>
        
        </author>
        
        <date year="2017" month="May"/>
        
        <abstract>
        
        <t>
        
        RFC 2119 specifies common key words that may be used in protocol specifications. This document aims to reduce the ambiguity by clarifying that only UPPERCASE usage of the key words have the defined special meanings.
        
        </t>
        
        </abstract>
        
        </front>
        
        <seriesInfo name="BCP" value="14"/>
        
        <seriesInfo name="RFC" value="8174"/>
        
        <seriesInfo name="DOI" value="10.17487/RFC8174"/>
        
        </reference>
      - <reference anchor="W3C.WD-html5-20110525" target="http://www.w3.org/TR/2011/WD-html5-20110525" xml:base="https://xml2rfc.tools.ietf.org/public/rfc/bibxml4/reference.W3C.WD-html5-20110525.xml">
        
        <front>
        
        <title>
        
        HTML5
        
        </title>
        
        <seriesInfo name="World Wide Web Consortium LastCall," value="WD-html5-20110525"/>
        
        <author initials="I." surname="Hickson" fullname="Ian Hickson">
        
        <organization/>
        
        </author>
        
        <date month="May" year="2011"/>
        
        </front>
        
        </reference>
      - <reference anchor="I-D.ietf-ice-trickle">
        
        <front>
        
        <title>
        
        Trickle ICE: Incremental Provisioning of Candidates for the Interactive Connectivity Establishment (ICE) Protocol
        
        </title>
        
        <seriesInfo name="Work in Progress," value="draft-ietf-ice-trickle-21"/>
        
        <author initials="E" surname="Ivov" fullname="Emil Ivov">
        
        <organization/>
        
        </author>
        
        <author initials="E" surname="Rescorla" fullname="Eric Rescorla">
        
        <organization/>
        
        </author>
        
        <author initials="J" surname="Uberti" fullname="Justin Uberti">
        
        <organization/>
        
        </author>
        
        <author initials="P" surname="Saint-Andre" fullname="Peter Saint-Andre">
        
        <organization/>
        
        </author>
        
        <date month="April" year="2018"/>
        
        <abstract>
        
        <t>
        
        This document describes "Trickle ICE", an extension to the Interactive Connectivity Establishment (ICE) protocol that enables ICE agents to begin connectivity checks while they are still gathering candidates, by incrementally exchanging candidates over time instead of all at once. This method can considerably accelerate the process of establishing a communication session.
        
        </t>
        
        </abstract>
        
        </front>
        
        </reference>
      - <reference anchor="I-D.ietf-mmusic-sdp-bundle-negotiation">
        
        <front>
        
        <title>
        
        Negotiating Media Multiplexing Using the Session Description Protocol (SDP)
        
        </title>
        
        <seriesInfo name="Work in Progress," value="draft-ietf-mmusic-sdp-bundle-negotiation-54"/>
        
        <author initials="C" surname="Holmberg" fullname="Christer Holmberg">
        
        <organization/>
        
        </author>
        
        <author initials="H" surname="Alvestrand" fullname="Harald Alvestrand">
        
        <organization/>
        
        </author>
        
        <author initials="C" surname="Jennings" fullname="Cullen Jennings">
        
        <organization/>
        
        </author>
        
        <date month="December" year="2018"/>
        
        <abstract>
        
        <t>
        
        This specification defines a new Session Description Protocol (SDP) Grouping Framework extension, 'BUNDLE'. The extension can be used with the SDP Offer/Answer mechanism to negotiate the usage of a single transport (5-tuple) for sending and receiving media described by multiple SDP media descriptions ("m=" sections). Such transport is referred to as a BUNDLE transport, and the media is referred to as bundled media. The "m=" sections that use the BUNDLE transport form a BUNDLE group. This specification updates RFC 3264, to also allow assigning a zero port value to a "m=" section in cases where the media described by the "m=" section is not disabled or rejected. This specification updates RFC 5888, to also allow an SDP 'group' attribute to contain an identification-tag that identifies a "m=" section with the port set to zero. This specification defines a new RTP Control Protocol (RTCP) source description (SDES) item and a new RTP header extension that can be used to correlate bundled RTP/RTCP packets with their appropriate "m=" section. This specification updates RFC 7941, by adding an exception, for the MID RTP header extension, to the requirement regarding protection of an SDES RTP header extension carrying an SDES item for the MID RTP header extension.
        
        </t>
        
        </abstract>
        
        </front>
        
        </reference>
      - <reference anchor="I-D.ietf-rtcweb-gateways">
        
        <front>
        
        <title>
        
        WebRTC Gateways
        
        </title>
        
        <seriesInfo name="Work in Progress," value="draft-ietf-rtcweb-gateways-02"/>
        
        <author initials="H" surname="Alvestrand" fullname="Harald T. Alvestrand">
        
        <organization/>
        
        </author>
        
        <author initials="U" surname="Rauschenbach" fullname="Uwe Rauschenbach">
        
        <organization/>
        
        </author>
        
        <date month="January" year="2016"/>
        
        <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>
        
        </front>
        
        </reference>
      - <reference anchor="I-D.ietf-tsvwg-rtcweb-qos">
        
        <front>
        
        <title>
        
        DSCP Packet Markings for WebRTC QoS
        
        </title>
        
        <seriesInfo name="Work in Progress," value="draft-ietf-tsvwg-rtcweb-qos-18"/>
        
        <author initials="P" surname="Jones" fullname="Paul Jones">
        
        <organization/>
        
        </author>
        
        <author initials="S" surname="Dhesikan" fullname="Subha Dhesikan">
        
        <organization/>
        
        </author>
        
        <author initials="C" surname="Jennings" fullname="Cullen Jennings">
        
        <organization/>
        
        </author>
        
        <author initials="D" surname="Druta" fullname="Dan Druta">
        
        <organization/>
        
        </author>
        
        <date month="August" year="2016"/>
        
        <abstract>
        
        <t>
        
        Many networks, such as service provider and enterprise networks, can provide different forwarding treatments for individual packets based on Differentiated Services Code Point (DSCP) values on a per-hop basis. This document provides the recommended DSCP values for web browsers to use for various classes of WebRTC traffic.
        
        </t>
        
        </abstract>
        
        </front>
        
        </reference>
      - <reference anchor="XEP-0166" target="http://xmpp.org/extensions/xep-0166.html">
        
        <front>
        
        <title>
        
        Jingle
        
        </title>
        
        <seriesInfo name="XSF XEP" value="0166"/>
        
        <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 month="June" year="2007"/>
        
        </front>
        
        </reference>
      - <reference anchor="XEP-0124" target="http://xmpp.org/extensions/xep-0124.html">
        
        <front>
        
        <title>
        
        BOSH
        
        </title>
        
        <seriesInfo name="XSF XEP" value="0124"/>
        
        <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 month="November" year="2016"/>
        
        </front>
        
        </reference>
      - </references>
    - </references>
  - </back>
- </rfc>

<?xml version='1.0' encoding='utf-8'?>
<!DOCTYPE rfc SYSTEM "rfc2629-xhtml.ent">
<rfc xmlns:xi="http://www.w3.org/2001/XInclude" category="std" number="0000" ipr="trust200902" obsoletes="" updates="" consensus="true" submissionType="IETF" xml:lang="en" tocInclude="true" symRefs="true" sortRefs="true" version="3">
  <!-- xml2rfc v2v3 conversion 2.23.1 -->
  <front>
    <title abbrev="WebRTC Overview">Overview: Real Time Protocols for
    Browser-based Applications</title>
    <seriesInfo name="RFC" value="0000"/>
    <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 month="July" year="2019"/>
    <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 numbered="true" toc="default" anchor="Intro">
      <name>Introduction</name>
      <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" format="default"/>.</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  sub-protocols 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" format="default"/> 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" format="default"/> may serve
      as a suitable replacement for <xref target="RFC3361" format="default"/>. 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" format="default"/> 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 numbered="true" toc="default">
      <name>Principles and Terminology</name>
      <section numbered="true" toc="default">
        <name>Goals of this document</name>
        <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 numbered="true" toc="default">
        <name>Relationship between API and protocol</name>
        <t>The total WebRTC effort consists of two major parts, each
        consisting of multiple documents:</t>
        <ul spacing="normal">
          <li>A protocol specification, done in the IETF</li>
          <li>A JavaScript API specification, defined in a series of W3C
            documents <xref target="W3C.WD-webrtc-20120209" format="default"/><xref target="W3C.WD-mediacapture-streams-20120628" format="default"/></li>
        </ul>
        <t>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>

<dl newline="false" spacing="normal">
          <dt>Agent:</dt>
          <dd>Undefined term. See "SDP Agent" and "ICE
            Agent".</dd>
          <dt>Application Programming Interface (API):</dt>
          <dd>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.</dd>
          <dt>Browser:</dt>
          <dd>Used synonymously with "Interactive User
            Agent" as defined in the HTML specification <xref target="W3C.WD-html5-20110525" format="default"/>. See also "WebRTC User
            Agent".</dd>
          <dt>Data Channel:</dt>
          <dd>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.</dd>
          <dt>ICE Agent:</dt>
          <dd>An implementation of the Interactive
            Connectivity Establishment (ICE) <xref target="RFC5245" format="default"/> 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" format="default">
            </xref>).</dd>
          <dt>Interactive:</dt>
          <dd>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.</dd>
          <dt>Media:</dt>
          <dd>Audio and video content. Not to be confused
            with "transmission media" such as wires.</dd>
          <dt>Media Path:</dt>
          <dd>The path that media data follows from
            one WebRTC endpoint to another.</dd>
          <dt>Protocol:</dt>
          <dd>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.</dd>
          <dt>Real-time Media:</dt>
          <dd>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.</dd>
          <dt>SDP Agent:</dt>
          <dd>The protocol implementation involved in
            the Session Description Protocol (SDP) offer/answer exchange, as
            defined in <xref target="RFC3264" sectionFormat="comma" section="3"/></dd>
          <dt>Signaling:</dt>
          <dd>Communication that happens in order to
            establish, manage and control media paths and data paths.</dd>
          <dt>Signaling Path:</dt>
          <dd>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.</dd>
          <dt>WebRTC Browser:</dt>
          <dd>(also called a WebRTC User Agent
            or WebRTC UA) Something that conforms to both the protocol
            specification and the JavaScript API cited above.</dd>
          <dt>WebRTC non-Browser:</dt>
          <dd> 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".</dd>
          <dt>WebRTC Endpoint:</dt>
          <dd> Either a WebRTC browser or a
            WebRTC non-browser. It conforms to the protocol specification.</dd>
          <dt>WebRTC-compatible Endpoint:</dt>
          <dd> 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.</dd>
          <dt>WebRTC Gateway:</dt>
          <dd> A WebRTC-compatible endpoint that
            mediates media traffic to non-WebRTC entities.</dd>
        </dl>


        <t>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" format="default"/>.</t>
      </section>
      <section numbered="true" toc="default">
        <name>On interoperability and innovation</name>
        <t>The "Mission statement of the IETF" <xref target="RFC3935" format="default"/> 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>
        <ul spacing="normal">
          <li>Two parties communicate, through some mechanism, what
            functionality they both are able to support</li>
          <li>They use that shared communicative functionality to
            communicate, or, failing to find anything in common, give up on
            communication.</li>
        </ul>
        <t>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 numbered="true" toc="default">
        <name>Terminology</name>
        <t>The key words "<bcp14>MUST</bcp14>", "<bcp14>MUST NOT</bcp14>",
    "<bcp14>REQUIRED</bcp14>", "<bcp14>SHALL</bcp14>", "<bcp14>SHALL
    NOT</bcp14>", "<bcp14>SHOULD</bcp14>", "<bcp14>SHOULD NOT</bcp14>",
    "<bcp14>RECOMMENDED</bcp14>", "<bcp14>NOT RECOMMENDED</bcp14>",
    "<bcp14>MAY</bcp14>", and "<bcp14>OPTIONAL</bcp14>" in this document are
    to be interpreted as described in BCP 14 <xref target="RFC2119"/>
    <xref target="RFC8174"/> when, and only when, they appear in all capitals,
    as shown here. 
</t>
      </section>
    </section>
    <section numbered="true" toc="default">
      <name>Architecture and Functionality groups</name>
      <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">
        <name>Browser Model</name>
        <artwork name="" type="" align="left" alt="">

                                                                          
                     +------------------------+  On-the-wire              
                     |                        |  Protocols                
                     |      Servers           |---------&gt;                 
                     |                        |                           
                     |                        |                           
                     +------------------------+                           
                                 ^                                        
                                 |                                        
                                 |                                        
                                 | HTTPS/                                  
                                 | WebSockets                             
                                 |                                              
                                 |                                        
                   +----------------------------+                         
                   |    JavaScript/HTML/CSS     |                         
                   +----------------------------+                         
                Other  ^                 ^ RTC                             
                APIs   |                 | APIs                            
                   +---|-----------------|------+                         
                   |   |                 |      |                         
                   |                 +---------+|                         
                   |                 | Browser ||  On-the-wire            
                   | Browser         | RTC     ||  Protocols              
                   |                 | Function|-----------&gt;              
                   |                 |         ||                         
                   |                 |         ||                         
                   |                 +---------+|                         
                   +---------------------|------+                         
                                         |                                
                                         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 <xref target="fig-webtrapezoid"/> below, JS is JavaScript.</t>
      <figure anchor="fig-webtrapezoid">
        <name>Browser RTC Trapezoid</name>
        <artwork name="" type="" align="left" alt=""><![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" format="default"/> or XMPP <xref target="RFC6120" format="default"/>)
      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" format="default"/> 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
       signaling, 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>
      <ul spacing="normal">
        <li>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.</li>
        <li>Data framing: RTP, SCTP, DTLS, and other data formats that serve
          as containers, and their functions for data confidentiality and
          integrity.</li>
        <li>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.</li>
        <li>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.</li>
        <li>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.</li>
        <li>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.</li>
      </ul>
      <t>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" numbered="true" toc="default">
      <name>Data transport</name>
      <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 <bcp14>MUST</bcp14> implement the transport protocols described in
      <xref target="RFCZZZZ" format="default"/>.</t>
    </section>
    <section numbered="true" toc="default">
      <name>Data framing and securing</name>
      <t>The format for media transport is RTP <xref target="RFC3550" format="default"/>.
      Implementation of SRTP <xref target="RFC3711" format="default"/> is <bcp14>REQUIRED</bcp14> for all
      implementations.</t>
      <t>The detailed considerations for usage of functions from RTP and SRTP
      are given in <xref target="RFCAAAA" format="default"/>. The security
      considerations for the WebRTC use case are in <xref target="RFCYYYY" format="default"/>, and the resulting security
      functions are described in <xref target="RFCDDDD" format="default"/>.</t>
      <t>Considerations for the transfer of data that is not in RTP format is
      described in <xref target="RFCBBBB" format="default"/>, and a
      supporting protocol for establishing individual data channels is
      described in <xref target="RFCCCCC" format="default"/>. WebRTC
      endpoints <bcp14>MUST</bcp14> implement these two specifications.</t>
      <t>WebRTC endpoints <bcp14>MUST</bcp14> implement <xref target="RFCAAAA" format="default"/>, <xref target="RFCYYYY" format="default"/>, <xref target="RFCDDDD" format="default"/>, and the requirements they
      include.</t>
    </section>
    <section anchor="ch-data" numbered="true" toc="default">
      <name>Data formats</name>
      <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 <bcp14>MUST</bcp14> implement the
      codecs and profiles required in <xref target="RFC7874" format="default"/> and <xref target="RFC7742" format="default"/>.</t>
    </section>
    <section numbered="true" toc="default">
      <name>Connection management</name>
      <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>
      <ol spacing="normal" type="1">
        <li>The WebRTC media negotiations will be capable of representing the
          same SDP offer/answer semantics <xref target="RFC3264" format="default"/> 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.</li>
        <li>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.</li>
        <li>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.</li>
      </ol>
      <t>The particular choices made for WebRTC, and their implications
      for the API offered by a browser implementing WebRTC, are described in
      <xref target="RFCEEEE" format="default"/>.</t>
      <t>WebRTC browsers <bcp14>MUST</bcp14> implement <xref target="RFCEEEE" format="default"/>.</t>
      <t>WebRTC endpoints <bcp14>MUST</bcp14> 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" format="default"/>, RTCP-mux <xref target="RFC5761" format="default"/> and Trickle ICE <xref target="I-D.ietf-ice-trickle" format="default"/>), but do not need to support the API
      functionality described there.</t>
    </section>
    <section numbered="true" toc="default">
      <name>Presentation and control</name>
      <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" format="default"/>, and the media capture API <xref target="W3C.WD-mediacapture-streams-20120628" format="default"/>.</t>
      <t>WebRTC browsers <bcp14>MUST</bcp14> implement these two specifications.</t>
    </section>
    <section numbered="true" toc="default">
      <name>Local system support functions</name>
      <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>
      <ul spacing="normal">
        <li>Echo cancellation should be good enough to achieve the
          suppression of acoustical feedback loops below a perceptually
          noticeable level.</li>
        <li>Privacy concerns <bcp14>MUST</bcp14> 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.</li>
        <li>Automatic gain control, if present, should normalize a speaking
          voice into a reasonable dB range.</li>
      </ul>
      <t>The requirements on WebRTC systems with regard to audio
      processing are found in <xref target="RFC7874" format="default"/> 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" format="default"/>.</t>
      <t>WebRTC endpoints <bcp14>MUST</bcp14> implement the processing functions in <xref target="RFC7874" format="default"/>. (Together with the requirement in <xref target="ch-data" format="default"/>, this means that WebRTC endpoints <bcp14>MUST</bcp14> implement the
      whole document.)</t>
    </section>
    <section anchor="IANA" numbered="true" toc="default">
      <name>IANA Considerations</name>
      <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" numbered="true" toc="default">
      <name>Security Considerations</name>
      <t>Security of the web-enabled real time communications comes in several
      pieces:</t>
      <ul spacing="normal">
        <li>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.</li>
        <li>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.</li>
        <li>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).</li>
      </ul>
      <t>The security analysis, and the requirements derived from that
      analysis, is contained in <xref target="RFCYYYY" format="default"/>.</t>
      <t>It is also important to read the security sections of <xref target="W3C.WD-mediacapture-streams-20120628" format="default"/> and <xref target="W3C.WD-webrtc-20120209" format="default"/>.</t>
    </section>
    <section anchor="Acknowledgements" numbered="true" toc="default">
      <name>Acknowledgements</name>
      <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 <xref target="Intro"/>.</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>
      <name>References</name>
      <references>
        <name>Normative References</name>

<reference anchor="RFC2119" target="https://www.rfc-editor.org/info/rfc2119" xml:base="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.2119.xml">
<front>
<title>Key words for use in RFCs to Indicate Requirement Levels</title>
<author initials="S." surname="Bradner" fullname="S. Bradner"><organization/></author>
<date year="1997" month="March"/>
<abstract><t>In many standards track documents several words are used to signify the requirements in the specification.  These words are often capitalized. This document defines these words as they should be interpreted in IETF documents.  This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements.</t></abstract>
</front>
<seriesInfo name="BCP" value="14"/>
<seriesInfo name="RFC" value="2119"/>
<seriesInfo name="DOI" value="10.17487/RFC2119"/>
</reference>
<reference anchor="RFC3264" target="https://www.rfc-editor.org/info/rfc3264" xml:base="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.3264.xml">
<front>
<title>An Offer/Answer Model with Session Description Protocol (SDP)</title>
<author initials="J." surname="Rosenberg" fullname="J. Rosenberg"><organization/></author>
<author initials="H." surname="Schulzrinne" fullname="H. Schulzrinne"><organization/></author>
<date year="2002" month="June"/>
<abstract><t>This document defines a mechanism by which two entities can make use of the Session Description Protocol (SDP) to arrive at a common view of a multimedia session between them.  In the model, one participant offers the other a description of the desired session from their perspective, and the other participant answers with the desired session from their perspective.  This offer/answer model is most useful in unicast sessions where information from both participants is needed for the complete view of the session.  The offer/answer model is used by protocols like the Session Initiation Protocol (SIP).  [STANDARDS-TRACK]</t></abstract>
</front>
<seriesInfo name="RFC" value="3264"/>
<seriesInfo name="DOI" value="10.17487/RFC3264"/>
</reference>
<reference anchor="RFC3550" target="https://www.rfc-editor.org/info/rfc3550" xml:base="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.3550.xml">
<front>
<title>RTP: A Transport Protocol for Real-Time Applications</title>
<author initials="H." surname="Schulzrinne" fullname="H. Schulzrinne"><organization/></author>
<author initials="S." surname="Casner" fullname="S. Casner"><organization/></author>
<author initials="R." surname="Frederick" fullname="R. Frederick"><organization/></author>
<author initials="V." surname="Jacobson" fullname="V. Jacobson"><organization/></author>
<date year="2003" month="July"/>
<abstract><t>This memorandum describes RTP, the real-time transport protocol.  RTP provides end-to-end network transport functions suitable for applications transmitting real-time data, such as audio, video or simulation data, over multicast or unicast network services.  RTP does not address resource reservation and does not guarantee quality-of- service for real-time services.  The data transport is augmented by a control protocol (RTCP) to allow monitoring of the data delivery in a manner scalable to large multicast networks, and to provide minimal control and identification functionality.  RTP and RTCP are designed to be independent of the underlying transport and network layers.  The protocol supports the use of RTP-level translators and mixers. Most of the text in this memorandum is identical to RFC 1889 which it obsoletes.  There are no changes in the packet formats on the wire, only changes to the rules and algorithms governing how the protocol is used. The biggest change is an enhancement to the scalable timer algorithm for calculating when to send RTCP packets in order to minimize transmission in excess of the intended rate when many participants join a session simultaneously.  [STANDARDS-TRACK]</t></abstract>
</front>
<seriesInfo name="STD" value="64"/>
<seriesInfo name="RFC" value="3550"/>
<seriesInfo name="DOI" value="10.17487/RFC3550"/>
</reference>
<reference anchor="RFC3711" target="https://www.rfc-editor.org/info/rfc3711" xml:base="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.3711.xml">
<front>
<title>The Secure Real-time Transport Protocol (SRTP)</title>
<author initials="M." surname="Baugher" fullname="M. Baugher"><organization/></author>
<author initials="D." surname="McGrew" fullname="D. McGrew"><organization/></author>
<author initials="M." surname="Naslund" fullname="M. Naslund"><organization/></author>
<author initials="E." surname="Carrara" fullname="E. Carrara"><organization/></author>
<author initials="K." surname="Norrman" fullname="K. Norrman"><organization/></author>
<date year="2004" month="March"/>
<abstract><t>This document describes the Secure Real-time Transport Protocol (SRTP), a profile of the Real-time Transport Protocol (RTP), which can provide confidentiality, message authentication, and replay protection to the RTP traffic and to the control traffic for RTP, the Real-time Transport Control Protocol (RTCP).   [STANDARDS-TRACK]</t></abstract>
</front>
<seriesInfo name="RFC" value="3711"/>
<seriesInfo name="DOI" value="10.17487/RFC3711"/>
</reference>
<reference anchor="RFC5245" target="https://www.rfc-editor.org/info/rfc5245" xml:base="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.5245.xml">
<front>
<title>Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols</title>
<author initials="J." surname="Rosenberg" fullname="J. Rosenberg"><organization/></author>
<date year="2010" month="April"/>
<abstract><t>This document describes a protocol for Network Address Translator (NAT) traversal for UDP-based multimedia sessions established with the offer/answer model.  This protocol is called Interactive Connectivity Establishment (ICE).  ICE makes use of the Session Traversal Utilities for NAT (STUN) protocol and its extension, Traversal Using Relay NAT (TURN).  ICE can be used by any protocol utilizing the offer/answer model, such as the Session Initiation Protocol (SIP).  [STANDARDS-TRACK]</t></abstract>
</front>
<seriesInfo name="RFC" value="5245"/>
<seriesInfo name="DOI" value="10.17487/RFC5245"/>
</reference>
<reference anchor="RFC7742" target="https://www.rfc-editor.org/info/rfc7742" xml:base="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.7742.xml">
<front>
<title>WebRTC Video Processing and Codec Requirements</title>
<author initials="A.B." surname="Roach" fullname="A.B. Roach"><organization/></author>
<date year="2016" month="March"/>
<abstract><t>This specification provides the requirements and considerations for WebRTC applications to send and receive video across a network.  It specifies the video processing that is required as well as video codecs and their parameters.</t></abstract>
</front>
<seriesInfo name="RFC" value="7742"/>
<seriesInfo name="DOI" value="10.17487/RFC7742"/>
</reference>
<reference anchor="RFC7874" target="https://www.rfc-editor.org/info/rfc7874" xml:base="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.7874.xml">
<front>
<title>WebRTC Audio Codec and Processing Requirements</title>
<author initials="JM." surname="Valin" fullname="JM. Valin"><organization/></author>
<author initials="C." surname="Bran" fullname="C. Bran"><organization/></author>
<date year="2016" month="May"/>
<abstract><t>This document outlines the audio codec and processing requirements for WebRTC endpoints.</t></abstract>
</front>
<seriesInfo name="RFC" value="7874"/>
<seriesInfo name="DOI" value="10.17487/RFC7874"/>
</reference>

   
    
        <reference anchor="RFCYYYY" target="https://www.rfc-editor.org/info/rfcYYYY">
          <front>
            <title>Security Considerations for WebRTC</title>
            <seriesInfo name="RFC" value="YYYY"/>
            <seriesInfo name="DOI" value="10.17487/RFCYYYY"/>
            <author initials="E" surname="Rescorla" fullname="Eric Rescorla">
              <organization/>
            </author>
            <date month="August" year="2019"/>
            <abstract>
              <t>WebRTC is a protocol suite for use with real-time applications that can be deployed in browsers - "real time communication on the Web". This document defines the WebRTC threat model and analyzes the security threats of WebRTC in that model.</t>
            </abstract>
          </front>
        </reference>

        <reference anchor="RFCZZZZ" target="https://www.rfc-editor.org/info/rfcZZZZ">
          <front>
            <title>Transports for WebRTC</title>
            <seriesInfo name="RFC" value="ZZZZ"/>
            <seriesInfo name="DOI" value="10.17487/RFCZZZZ"/>
            <author initials="H" surname="Alvestrand" fullname="Harald Alvestrand">
              <organization/>
            </author>
            <date month="August" year="2019"/>
            <abstract>
              <t>This document describes the data transport protocols used by WebRTC, including the protocols used for interaction with intermediate boxes such as firewalls, relays and NAT boxes.</t>
            </abstract>
          </front>
        </reference>

        <reference anchor="RFCAAAA" target="https://www.rfc-editor.org/info/rfcAAAA">
          <front>
            <title>Web Real-Time Communication (WebRTC): Media Transport and Use of RTP</title>
            <seriesInfo name="RFC" value="AAAA"/>
            <seriesInfo name="DOI" value="10.17487/RFCAAAA"/>
            <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="Joerg Ott">
              <organization/>
            </author>
            <date month="August" year="2019"/>
            <abstract>
              <t>The Web Real-Time Communication (WebRTC) framework provides support for direct interactive rich communication using audio, video, text, collaboration, games, etc. between two peers' web-browsers.  This memo describes the media transport aspects of the WebRTC framework. It specifies how the Real-time Transport Protocol (RTP) is used in the WebRTC context, and gives requirements for which RTP features, profiles, and extensions need to be supported.</t>
            </abstract>
          </front>
        </reference>

        <reference anchor="RFCBBBB" target="https://www.rfc-editor.org/info/rfcBBBB">
          <front>
            <title>WebRTC Data Channels</title>
            <seriesInfo name="RFC" value="BBBB"/>
            <seriesInfo name="DOI" value="10.17487/RFCBBBB"/>
            <author initials="R" surname="Jesup" fullname="Randell Jesup">
              <organization/>
            </author>
            <author initials="S" surname="Loreto" fullname="Salvatore Loreto">
              <organization/>
            </author>
            <author initials="M" surname="Tuexen" fullname="Michael Tuexen">
              <organization/>
            </author>
            <date month="August" year="2019"/>
            <abstract>
              <t>The WebRTC framework specifies protocol support for direct interactive rich communication using audio, video, and data between two peers' web-browsers.  This document specifies the non-media data transport aspects of the WebRTC framework.  It provides an architectural overview of how the Stream Control Transmission Protocol (SCTP) is used in the WebRTC context as a generic transport service allowing WEB-browsers to exchange generic data from peer to peer.</t>
            </abstract>
          </front>
        </reference>

        <reference anchor="RFCCCCC" target="https://www.rfc-editor.org/info/rfcCCCC">
          <front>
            <title>WebRTC Data Channel Establishment Protocol</title>
             <seriesInfo name="RFC" value="CCCC"/>
            <seriesInfo name="DOI" value="10.17487/RFCCCCC"/>
            <author initials="R" surname="Jesup" fullname="Randell Jesup">
              <organization/>
            </author>
            <author initials="S" surname="Loreto" fullname="Salvatore Loreto">
              <organization/>
            </author>
            <author initials="M" surname="Tuexen" fullname="Michael Tuexen">
              <organization/>
            </author>
            <date month="August" year="2019"/>
            <abstract>
              <t>The WebRTC framework specifies protocol support for direct interactive rich communication using audio, video, and data between two peers' web-browsers.  This document specifies a simple protocol for establishing symmetric Data Channels between the peers.  It uses a two way handshake and allows sending of user data without waiting for the handshake to complete.</t>
            </abstract>
          </front>
        </reference>

        <reference anchor="RFCDDDD" target="https://www.rfc-editor.org/info/rfcDDDD">
          <front>
            <title>WebRTC Security Architecture</title>
            <seriesInfo name="RFC" value="DDDD"/>
            <seriesInfo name="DOI" value="10.17487/RFCDDDD"/>
            <author initials="E" surname="Rescorla" fullname="Eric Rescorla">
              <organization/>
            </author>
            <date month="August" year="2019"/>
            <abstract>
              <t>This document defines the security architecture for WebRTC, a protocol suite intended for use with real-time applications that can be deployed in browsers - "real time communication on the Web".</t>
            </abstract>
          </front>
        </reference>

        <reference anchor="RFCEEEE" target="https://www.rfc-editor.org/info/rfcEEEE">
          <front>
            <title>JavaScript Session Establishment Protocol</title>
            <seriesInfo name="RFC" value="EEEE"/>
            <seriesInfo name="DOI" value="10.17487/RFCEEEE"/>
            <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">
              <organization/>
            </author>
            <date month="August" year="2019"/>
            <abstract>
              <t>This document describes the mechanisms for allowing a JavaScript application to control the signaling plane of a multimedia session via the interface specified in the W3C RTCPeerConnection API, and discusses how this relates to existing signaling protocols.</t>
            </abstract>
          </front>
        </reference>



        <reference anchor="W3C.WD-webrtc-20120209" target="http://www.w3.org/TR/2012/WD-webrtc-20120209" xml:base="https://xml2rfc.tools.ietf.org/public/rfc/bibxml4/reference.W3C.WD-webrtc-20120209.xml">
          <front>
            <title>WebRTC 1.0: Real-time Communication Between Browsers</title>
            <seriesInfo name="World Wide Web Consortium WD" value="WD-webrtc-20120209"/>
            <author initials="A." surname="Bergkvist" fullname="Adam Bergkvist">
              <organization/>
            </author>
            <author initials="D." surname="Burnett" fullname="Daniel C. Burnett">
              <organization/>
            </author>
            <author initials="C." surname="Jennings" fullname="Cullen Jennings">
              <organization/>
            </author>
            <author initials="A." surname="Narayanan" fullname="Anant Narayanan">
              <organization/>
            </author>
            <date month="February" year="2012"/>
          </front>
        </reference>
        <reference anchor="W3C.WD-mediacapture-streams-20120628" target="http://www.w3.org/TR/2012/WD-mediacapture-streams-20120628" xml:base="https://xml2rfc.tools.ietf.org/public/rfc/bibxml4/reference.W3C.WD-mediacapture-streams-20120628.xml">
          <front>
            <title>Media Capture and Streams</title>
            <seriesInfo name="World Wide Web Consortium WD" value="WD-mediacapture-streams-20120628"/>
            <author initials="D." surname="Burnett" fullname="Daniel C. Burnett">
              <organization/>
            </author>
            <author initials="A." surname="Narayanan" fullname="Anant Narayanan">
              <organization/>
            </author>
            <date month="June" year="2012"/>
          </front>
        </reference>
      </references>
      <references>
        <name>Informative References</name>

<reference anchor="RFC3261" target="https://www.rfc-editor.org/info/rfc3261" xml:base="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.3261.xml">
<front>
<title>SIP: Session Initiation Protocol</title>
<author initials="J." surname="Rosenberg" fullname="J. Rosenberg"><organization/></author>
<author initials="H." surname="Schulzrinne" fullname="H. Schulzrinne"><organization/></author>
<author initials="G." surname="Camarillo" fullname="G. Camarillo"><organization/></author>
<author initials="A." surname="Johnston" fullname="A. Johnston"><organization/></author>
<author initials="J." surname="Peterson" fullname="J. Peterson"><organization/></author>
<author initials="R." surname="Sparks" fullname="R. Sparks"><organization/></author>
<author initials="M." surname="Handley" fullname="M. Handley"><organization/></author>
<author initials="E." surname="Schooler" fullname="E. Schooler"><organization/></author>
<date year="2002" month="June"/>
<abstract><t>This document describes Session Initiation Protocol (SIP), an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants.  These sessions include Internet telephone calls, multimedia distribution, and multimedia conferences.  [STANDARDS-TRACK]</t></abstract>
</front>
<seriesInfo name="RFC" value="3261"/>
<seriesInfo name="DOI" value="10.17487/RFC3261"/>
</reference>
<reference anchor="RFC3361" target="https://www.rfc-editor.org/info/rfc3361" xml:base="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.3361.xml">
<front>
<title>Dynamic Host Configuration Protocol (DHCP-for-IPv4) Option for Session Initiation Protocol (SIP) Servers</title>
<author initials="H." surname="Schulzrinne" fullname="H. Schulzrinne"><organization/></author>
<date year="2002" month="August"/>
</front>
<seriesInfo name="RFC" value="3361"/>
<seriesInfo name="DOI" value="10.17487/RFC3361"/>
</reference>
<reference anchor="RFC3935" target="https://www.rfc-editor.org/info/rfc3935" xml:base="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.3935.xml">
<front>
<title>A Mission Statement for the IETF</title>
<author initials="H." surname="Alvestrand" fullname="H. Alvestrand"><organization/></author>
<date year="2004" month="October"/>
<abstract><t>This memo gives a mission statement for the IETF, tries to define the terms used in the statement sufficiently to make the mission statement understandable and useful, argues why the IETF needs a mission statement, and tries to capture some of the debate that led to this point.  This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements.</t></abstract>
</front>
<seriesInfo name="BCP" value="95"/>
<seriesInfo name="RFC" value="3935"/>
<seriesInfo name="DOI" value="10.17487/RFC3935"/>
</reference>
<reference anchor="RFC5761" target="https://www.rfc-editor.org/info/rfc5761" xml:base="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.5761.xml">
<front>
<title>Multiplexing RTP Data and Control Packets on a Single Port</title>
<author initials="C." surname="Perkins" fullname="C. Perkins"><organization/></author>
<author initials="M." surname="Westerlund" fullname="M. Westerlund"><organization/></author>
<date year="2010" month="April"/>
<abstract><t>This memo discusses issues that arise when multiplexing RTP data packets and RTP Control Protocol (RTCP) packets on a single UDP port. It updates RFC 3550 and RFC 3551 to describe when such multiplexing is and is not appropriate, and it explains how the Session Description Protocol (SDP) can be used to signal multiplexed sessions.  [STANDARDS-TRACK]</t></abstract>
</front>
<seriesInfo name="RFC" value="5761"/>
<seriesInfo name="DOI" value="10.17487/RFC5761"/>
</reference>
<reference anchor="RFC6120" target="https://www.rfc-editor.org/info/rfc6120" xml:base="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.6120.xml">
<front>
<title>Extensible Messaging and Presence Protocol (XMPP): Core</title>
<author initials="P." surname="Saint-Andre" fullname="P. Saint-Andre"><organization/></author>
<date year="2011" month="March"/>
<abstract><t>The Extensible Messaging and Presence Protocol (XMPP) is an application profile of the Extensible Markup Language (XML) that enables the near-real-time exchange of structured yet extensible data between any two or more network entities.  This document defines XMPP's core protocol methods: setup and teardown of XML streams, channel encryption, authentication, error handling, and communication primitives for messaging, network availability ("presence"), and request-response interactions.  This document obsoletes RFC 3920.  [STANDARDS-TRACK]</t></abstract>
</front>
<seriesInfo name="RFC" value="6120"/>
<seriesInfo name="DOI" value="10.17487/RFC6120"/>
</reference>
<reference anchor="RFC7478" target="https://www.rfc-editor.org/info/rfc7478" xml:base="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.7478.xml">
<front>
<title>Web Real-Time Communication Use Cases and Requirements</title>
<author initials="C." surname="Holmberg" fullname="C. Holmberg"><organization/></author>
<author initials="S." surname="Hakansson" fullname="S. Hakansson"><organization/></author>
<author initials="G." surname="Eriksson" fullname="G. Eriksson"><organization/></author>
<date year="2015" month="March"/>
<abstract><t>This document describes web-based real-time communication use cases. Requirements on the browser functionality are derived from the use cases.</t><t>This document was developed in an initial phase of the work with rather minor updates at later stages.  It has not really served as a tool in deciding features or scope for the WG's efforts so far.  It is being published to record the early conclusions of the WG.  It will not be used as a set of rigid guidelines that specifications and implementations will be held to in the future.</t></abstract>
</front>
<seriesInfo name="RFC" value="7478"/>
<seriesInfo name="DOI" value="10.17487/RFC7478"/>
</reference>
<reference anchor="RFC8155" target="https://www.rfc-editor.org/info/rfc8155" xml:base="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.8155.xml">
<front>
<title>Traversal Using Relays around NAT (TURN) Server Auto Discovery</title>
<author initials="P." surname="Patil" fullname="P. Patil"><organization/></author>
<author initials="T." surname="Reddy" fullname="T. Reddy"><organization/></author>
<author initials="D." surname="Wing" fullname="D. Wing"><organization/></author>
<date year="2017" month="April"/>
<abstract><t>Current Traversal Using Relays around NAT (TURN) server discovery mechanisms are relatively static and limited to explicit configuration.  These are usually under the administrative control of the application or TURN service provider, and not the enterprise, ISP, or the network in which the client is located.  Enterprises and ISPs wishing to provide their own TURN servers need auto-discovery mechanisms that a TURN client could use with minimal or no configuration.  This document describes three such mechanisms for TURN server discovery.</t><t>This document updates RFC 5766 to relax the requirement for mutual authentication in certain cases.</t></abstract>
</front>
<seriesInfo name="RFC" value="8155"/>
<seriesInfo name="DOI" value="10.17487/RFC8155"/>
</reference>
<reference anchor="RFC8174" target="https://www.rfc-editor.org/info/rfc8174" xml:base="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.8174.xml">
<front>
<title>Ambiguity of Uppercase vs Lowercase in RFC 2119 Key Words</title>
<author initials="B." surname="Leiba" fullname="B. Leiba"><organization/></author>
<date year="2017" month="May"/>
<abstract><t>RFC 2119 specifies common key words that may be used in protocol  specifications.  This document aims to reduce the ambiguity by clarifying that only UPPERCASE usage of the key words have the  defined special meanings.</t></abstract>
</front>
<seriesInfo name="BCP" value="14"/>
<seriesInfo name="RFC" value="8174"/>
<seriesInfo name="DOI" value="10.17487/RFC8174"/>
</reference>

        
        <reference anchor="W3C.WD-html5-20110525" target="http://www.w3.org/TR/2011/WD-html5-20110525" xml:base="https://xml2rfc.tools.ietf.org/public/rfc/bibxml4/reference.W3C.WD-html5-20110525.xml">
          <front>
            <title>HTML5</title>
            <seriesInfo name="World Wide Web Consortium LastCall," value="WD-html5-20110525"/>
            <author initials="I." surname="Hickson" fullname="Ian Hickson">
              <organization/>
            </author>
            <date month="May" year="2011"/>
          </front>
        </reference>
        <reference anchor="I-D.ietf-ice-trickle">
          <front>
            <title>Trickle ICE: Incremental Provisioning of Candidates for the Interactive Connectivity Establishment (ICE) Protocol</title>
            <seriesInfo name="Work in Progress," value="draft-ietf-ice-trickle-21"/>
            <author initials="E" surname="Ivov" fullname="Emil Ivov">
              <organization/>
            </author>
            <author initials="E" surname="Rescorla" fullname="Eric Rescorla">
              <organization/>
            </author>
            <author initials="J" surname="Uberti" fullname="Justin Uberti">
              <organization/>
            </author>
            <author initials="P" surname="Saint-Andre" fullname="Peter Saint-Andre">
              <organization/>
            </author>
            <date month="April" year="2018"/>
            <abstract>
              <t>This document describes "Trickle ICE", an extension to the Interactive Connectivity Establishment (ICE) protocol that enables ICE agents to begin connectivity checks while they are still gathering candidates, by incrementally exchanging candidates over time instead of all at once.  This method can considerably accelerate the process of establishing a communication session.</t>
            </abstract>
          </front>
        </reference>

        <reference anchor="I-D.ietf-mmusic-sdp-bundle-negotiation">
          <front>
            <title>Negotiating Media Multiplexing Using the Session Description Protocol (SDP)</title>
            <seriesInfo name="Work in Progress," value="draft-ietf-mmusic-sdp-bundle-negotiation-54"/>
            <author initials="C" surname="Holmberg" fullname="Christer Holmberg">
              <organization/>
            </author>
            <author initials="H" surname="Alvestrand" fullname="Harald Alvestrand">
              <organization/>
            </author>
            <author initials="C" surname="Jennings" fullname="Cullen Jennings">
              <organization/>
            </author>
            <date month="December" year="2018"/>
            <abstract>
              <t>This specification defines a new Session Description Protocol (SDP) Grouping Framework extension, 'BUNDLE'.  The extension can be used with the SDP Offer/Answer mechanism to negotiate the usage of a single transport (5-tuple) for sending and receiving media described by multiple SDP media descriptions ("m=" sections).  Such transport is referred to as a BUNDLE transport, and the media is referred to as bundled media.  The "m=" sections that use the BUNDLE transport form a BUNDLE group.  This specification updates RFC 3264, to also allow assigning a zero port value to a "m=" section in cases where the media described by the "m=" section is not disabled or rejected.  This specification updates RFC 5888, to also allow an SDP 'group' attribute to contain an identification-tag that identifies a "m=" section with the port set to zero.  This specification defines a new RTP Control Protocol (RTCP) source description (SDES) item and a new RTP header extension that can be used to correlate bundled RTP/RTCP packets with their appropriate "m=" section.  This specification updates RFC 7941, by adding an exception, for the MID RTP header extension, to the requirement regarding protection of an SDES RTP header extension carrying an SDES item for the MID RTP header extension.</t>
            </abstract>
          </front>
        </reference>

        <reference anchor="I-D.ietf-rtcweb-gateways">
          <front>
            <title>WebRTC Gateways</title>
            <seriesInfo name="Work in Progress," value="draft-ietf-rtcweb-gateways-02"/>
            <author initials="H" surname="Alvestrand" fullname="Harald T. Alvestrand">
              <organization/>
            </author>
            <author initials="U" surname="Rauschenbach" fullname="Uwe Rauschenbach">
              <organization/>
            </author>
            <date month="January" year="2016"/>
            <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>
          </front>
        </reference>

        <reference anchor="I-D.ietf-tsvwg-rtcweb-qos">
          <front>
            <title>DSCP Packet Markings for WebRTC QoS</title>
            <seriesInfo name="Work in Progress," value="draft-ietf-tsvwg-rtcweb-qos-18"/>
            <author initials="P" surname="Jones" fullname="Paul Jones">
              <organization/>
            </author>
            <author initials="S" surname="Dhesikan" fullname="Subha Dhesikan">
              <organization/>
            </author>
            <author initials="C" surname="Jennings" fullname="Cullen Jennings">
              <organization/>
            </author>
            <author initials="D" surname="Druta" fullname="Dan Druta">
              <organization/>
            </author>
            <date month="August" year="2016"/>
            <abstract>
              <t>Many networks, such as service provider and enterprise networks, can provide different forwarding treatments for individual packets based on Differentiated Services Code Point (DSCP) values on a per-hop basis.  This document provides the recommended DSCP values for web browsers to use for various classes of WebRTC traffic.</t>
            </abstract>
          </front>
        </reference>

        <reference anchor="XEP-0166" target="http://xmpp.org/extensions/xep-0166.html">
          <front>
            <title>Jingle</title>
            <seriesInfo name="XSF XEP" value="0166"/>
            <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 month="June" year="2007"/>
          </front>
        </reference>

        <reference anchor="XEP-0124" target="http://xmpp.org/extensions/xep-0124.html">
          <front>
            <title>BOSH</title>
            <seriesInfo name="XSF XEP" value="0124"/>
            <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 month="November" year="2016"/>
          </front>
        </reference>
      </references>
    </references>
  </back>
</rfc>

<reference anchor="RFC2119" target="https://www.rfc-editor.org/info/rfc2119" xml:base="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.2119.xml">
<front>
<title>Key words for use in RFCs to Indicate Requirement Levels</title>
<author initials="S." surname="Bradner" fullname="S. Bradner"><organization/></author>
<date year="1997" month="March"/>
<abstract><t>In many standards track documents several words are used to signify the requirements in the specification.  These words are often capitalized. This document defines these words as they should be interpreted in IETF documents.  This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements.</t></abstract>
</front>
<seriesInfo name="BCP" value="14"/>
<seriesInfo name="RFC" value="2119"/>
<seriesInfo name="DOI" value="10.17487/RFC2119"/>
</reference>

<reference anchor="RFC3264" target="https://www.rfc-editor.org/info/rfc3264" xml:base="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.3264.xml">
<front>
<title>An Offer/Answer Model with Session Description Protocol (SDP)</title>
<author initials="J." surname="Rosenberg" fullname="J. Rosenberg"><organization/></author>
<author initials="H." surname="Schulzrinne" fullname="H. Schulzrinne"><organization/></author>
<date year="2002" month="June"/>
<abstract><t>This document defines a mechanism by which two entities can make use of the Session Description Protocol (SDP) to arrive at a common view of a multimedia session between them.  In the model, one participant offers the other a description of the desired session from their perspective, and the other participant answers with the desired session from their perspective.  This offer/answer model is most useful in unicast sessions where information from both participants is needed for the complete view of the session.  The offer/answer model is used by protocols like the Session Initiation Protocol (SIP).  [STANDARDS-TRACK]</t></abstract>
</front>
<seriesInfo name="RFC" value="3264"/>
<seriesInfo name="DOI" value="10.17487/RFC3264"/>
</reference>

<reference anchor="RFC3550" target="https://www.rfc-editor.org/info/rfc3550" xml:base="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.3550.xml">
<front>
<title>RTP: A Transport Protocol for Real-Time Applications</title>
<author initials="H." surname="Schulzrinne" fullname="H. Schulzrinne"><organization/></author>
<author initials="S." surname="Casner" fullname="S. Casner"><organization/></author>
<author initials="R." surname="Frederick" fullname="R. Frederick"><organization/></author>
<author initials="V." surname="Jacobson" fullname="V. Jacobson"><organization/></author>
<date year="2003" month="July"/>
<abstract><t>This memorandum describes RTP, the real-time transport protocol.  RTP provides end-to-end network transport functions suitable for applications transmitting real-time data, such as audio, video or simulation data, over multicast or unicast network services.  RTP does not address resource reservation and does not guarantee quality-of- service for real-time services.  The data transport is augmented by a control protocol (RTCP) to allow monitoring of the data delivery in a manner scalable to large multicast networks, and to provide minimal control and identification functionality.  RTP and RTCP are designed to be independent of the underlying transport and network layers.  The protocol supports the use of RTP-level translators and mixers. Most of the text in this memorandum is identical to RFC 1889 which it obsoletes.  There are no changes in the packet formats on the wire, only changes to the rules and algorithms governing how the protocol is used. The biggest change is an enhancement to the scalable timer algorithm for calculating when to send RTCP packets in order to minimize transmission in excess of the intended rate when many participants join a session simultaneously.  [STANDARDS-TRACK]</t></abstract>
</front>
<seriesInfo name="STD" value="64"/>
<seriesInfo name="RFC" value="3550"/>
<seriesInfo name="DOI" value="10.17487/RFC3550"/>
</reference>

<reference anchor="RFC3711" target="https://www.rfc-editor.org/info/rfc3711" xml:base="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.3711.xml">
<front>
<title>The Secure Real-time Transport Protocol (SRTP)</title>
<author initials="M." surname="Baugher" fullname="M. Baugher"><organization/></author>
<author initials="D." surname="McGrew" fullname="D. McGrew"><organization/></author>
<author initials="M." surname="Naslund" fullname="M. Naslund"><organization/></author>
<author initials="E." surname="Carrara" fullname="E. Carrara"><organization/></author>
<author initials="K." surname="Norrman" fullname="K. Norrman"><organization/></author>
<date year="2004" month="March"/>
<abstract><t>This document describes the Secure Real-time Transport Protocol (SRTP), a profile of the Real-time Transport Protocol (RTP), which can provide confidentiality, message authentication, and replay protection to the RTP traffic and to the control traffic for RTP, the Real-time Transport Control Protocol (RTCP).   [STANDARDS-TRACK]</t></abstract>
</front>
<seriesInfo name="RFC" value="3711"/>
<seriesInfo name="DOI" value="10.17487/RFC3711"/>
</reference>

<?xml version='1.0' encoding='UTF-8'?>

<reference  anchor='RFC5245' target='https://www.rfc-editor.org/info/rfc5245'>
<front>
<title>Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols</title>
<author initials='J.' surname='Rosenberg' fullname='J. Rosenberg'><organization /></author>
<date year='2010' month='April' />
<abstract><t>This document describes a protocol for Network Address Translator (NAT) traversal for UDP-based multimedia sessions established with the offer/answer model.  This protocol is called Interactive Connectivity Establishment (ICE).  ICE makes use of the Session Traversal Utilities for NAT (STUN) protocol and its extension, Traversal Using Relay NAT (TURN).  ICE can be used by any protocol utilizing the offer/answer model, such as the Session Initiation Protocol (SIP).  [STANDARDS-TRACK]</t></abstract>
</front>
<seriesInfo name='RFC' value='5245'/>
<seriesInfo name='DOI' value='10.17487/RFC5245'/>
</reference>

<?xml version='1.0' encoding='UTF-8'?>

<reference  anchor='RFC7742' target='https://www.rfc-editor.org/info/rfc7742'>
<front>
<title>WebRTC Video Processing and Codec Requirements</title>
<author initials='A.B.' surname='Roach' fullname='A.B. Roach'><organization /></author>
<date year='2016' month='March' />
<abstract><t>This specification provides the requirements and considerations for WebRTC applications to send and receive video across a network.  It specifies the video processing that is required as well as video codecs and their parameters.</t></abstract>
</front>
<seriesInfo name='RFC' value='7742'/>
<seriesInfo name='DOI' value='10.17487/RFC7742'/>
</reference>

<?xml version='1.0' encoding='UTF-8'?>

<reference  anchor='RFC7874' target='https://www.rfc-editor.org/info/rfc7874'>
<front>
<title>WebRTC Audio Codec and Processing Requirements</title>
<author initials='JM.' surname='Valin' fullname='JM. Valin'><organization /></author>
<author initials='C.' surname='Bran' fullname='C. Bran'><organization /></author>
<date year='2016' month='May' />
<abstract><t>This document outlines the audio codec and processing requirements for WebRTC endpoints.</t></abstract>
</front>
<seriesInfo name='RFC' value='7874'/>
<seriesInfo name='DOI' value='10.17487/RFC7874'/>
</reference>

<?xml version='1.0' encoding='UTF-8'?>

<reference anchor='W3C.WD-webrtc-20120209'
           target='http://www.w3.org/TR/2012/WD-webrtc-20120209'>
<front>
<title>WebRTC 1.0: Real-time Communication Between Browsers</title>

<author initials='A.' surname='Bergkvist' fullname='Adam Bergkvist'>
    <organization />
</author>

<author initials='D.' surname='Burnett' fullname='Daniel C. Burnett'>
    <organization />
</author>

<author initials='C.' surname='Jennings' fullname='Cullen Jennings'>
    <organization />
</author>

<author initials='A.' surname='Narayanan' fullname='Anant Narayanan'>
    <organization />
</author>

<date month='February' day='9' year='2012' />
</front>

<seriesInfo name='World Wide Web Consortium WD' value='WD-webrtc-20120209' />
<format type='HTML' target='http://www.w3.org/TR/2012/WD-webrtc-20120209' />
</reference>

<?xml version='1.0' encoding='UTF-8'?>

<reference anchor='W3C.WD-mediacapture-streams-20120628'
           target='http://www.w3.org/TR/2012/WD-mediacapture-streams-20120628'>
<front>
<title>Media Capture and Streams</title>

<author initials='D.' surname='Burnett' fullname='Daniel C. Burnett'>
    <organization />
</author>

<author initials='A.' surname='Narayanan' fullname='Anant Narayanan'>
    <organization />
</author>

<date month='June' day='28' year='2012' />
</front>

<seriesInfo name='World Wide Web Consortium WD' value='WD-mediacapture-streams-20120628' />
<format type='HTML' target='http://www.w3.org/TR/2012/WD-mediacapture-streams-20120628' />
</reference>

<?xml version='1.0' encoding='UTF-8'?>

<reference  anchor='RFC3261' target='https://www.rfc-editor.org/info/rfc3261'>
<front>
<title>SIP: Session Initiation Protocol</title>
<author initials='J.' surname='Rosenberg' fullname='J. Rosenberg'><organization /></author>
<author initials='H.' surname='Schulzrinne' fullname='H. Schulzrinne'><organization /></author>
<author initials='G.' surname='Camarillo' fullname='G. Camarillo'><organization /></author>
<author initials='A.' surname='Johnston' fullname='A. Johnston'><organization /></author>
<author initials='J.' surname='Peterson' fullname='J. Peterson'><organization /></author>
<author initials='R.' surname='Sparks' fullname='R. Sparks'><organization /></author>
<author initials='M.' surname='Handley' fullname='M. Handley'><organization /></author>
<author initials='E.' surname='Schooler' fullname='E. Schooler'><organization /></author>
<date year='2002' month='June' />
<abstract><t>This document describes Session Initiation Protocol (SIP), an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants.  These sessions include Internet telephone calls, multimedia distribution, and multimedia conferences.  [STANDARDS-TRACK]</t></abstract>
</front>
<seriesInfo name='RFC' value='3261'/>
<seriesInfo name='DOI' value='10.17487/RFC3261'/>
</reference>

<?xml version='1.0' encoding='UTF-8'?>

<reference  anchor='RFC3361' target='https://www.rfc-editor.org/info/rfc3361'>
<front>
<title>Dynamic Host Configuration Protocol (DHCP-for-IPv4) Option for Session Initiation Protocol (SIP) Servers</title>
<author initials='H.' surname='Schulzrinne' fullname='H. Schulzrinne'><organization /></author>
<date year='2002' month='August' />
</front>
<seriesInfo name='RFC' value='3361'/>
<seriesInfo name='DOI' value='10.17487/RFC3361'/>
</reference>

<?xml version='1.0' encoding='UTF-8'?>

<reference  anchor='RFC3935' target='https://www.rfc-editor.org/info/rfc3935'>
<front>
<title>A Mission Statement for the IETF</title>
<author initials='H.' surname='Alvestrand' fullname='H. Alvestrand'><organization /></author>
<date year='2004' month='October' />
<abstract><t>This memo gives a mission statement for the IETF, tries to define the terms used in the statement sufficiently to make the mission statement understandable and useful, argues why the IETF needs a mission statement, and tries to capture some of the debate that led to this point.  This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements.</t></abstract>
</front>
<seriesInfo name='BCP' value='95'/>
<seriesInfo name='RFC' value='3935'/>
<seriesInfo name='DOI' value='10.17487/RFC3935'/>
</reference>

<?xml version='1.0' encoding='UTF-8'?>

<reference  anchor='RFC5761' target='https://www.rfc-editor.org/info/rfc5761'>
<front>
<title>Multiplexing RTP Data and Control Packets on a Single Port</title>
<author initials='C.' surname='Perkins' fullname='C. Perkins'><organization /></author>
<author initials='M.' surname='Westerlund' fullname='M. Westerlund'><organization /></author>
<date year='2010' month='April' />
<abstract><t>This memo discusses issues that arise when multiplexing RTP data packets and RTP Control Protocol (RTCP) packets on a single UDP port. It updates RFC 3550 and RFC 3551 to describe when such multiplexing is and is not appropriate, and it explains how the Session Description Protocol (SDP) can be used to signal multiplexed sessions.  [STANDARDS-TRACK]</t></abstract>
</front>
<seriesInfo name='RFC' value='5761'/>
<seriesInfo name='DOI' value='10.17487/RFC5761'/>
</reference>

<?xml version='1.0' encoding='UTF-8'?>

<reference  anchor='RFC6120' target='https://www.rfc-editor.org/info/rfc6120'>
<front>
<title>Extensible Messaging and Presence Protocol (XMPP): Core</title>
<author initials='P.' surname='Saint-Andre' fullname='P. Saint-Andre'><organization /></author>
<date year='2011' month='March' />
<abstract><t>The Extensible Messaging and Presence Protocol (XMPP) is an application profile of the Extensible Markup Language (XML) that enables the near-real-time exchange of structured yet extensible data between any two or more network entities.  This document defines XMPP's core protocol methods: setup and teardown of XML streams, channel encryption, authentication, error handling, and communication primitives for messaging, network availability (&quot;presence&quot;), and request-response interactions.  This document obsoletes RFC 3920.  [STANDARDS-TRACK]</t></abstract>
</front>
<seriesInfo name='RFC' value='6120'/>
<seriesInfo name='DOI' value='10.17487/RFC6120'/>
</reference>

<?xml version='1.0' encoding='UTF-8'?>

<reference  anchor='RFC7478' target='https://www.rfc-editor.org/info/rfc7478'>
<front>
<title>Web Real-Time Communication Use Cases and Requirements</title>
<author initials='C.' surname='Holmberg' fullname='C. Holmberg'><organization /></author>
<author initials='S.' surname='Hakansson' fullname='S. Hakansson'><organization /></author>
<author initials='G.' surname='Eriksson' fullname='G. Eriksson'><organization /></author>
<date year='2015' month='March' />
<abstract><t>This document describes web-based real-time communication use cases. Requirements on the browser functionality are derived from the use cases.</t><t>This document was developed in an initial phase of the work with rather minor updates at later stages.  It has not really served as a tool in deciding features or scope for the WG's efforts so far.  It is being published to record the early conclusions of the WG.  It will not be used as a set of rigid guidelines that specifications and implementations will be held to in the future.</t></abstract>
</front>
<seriesInfo name='RFC' value='7478'/>
<seriesInfo name='DOI' value='10.17487/RFC7478'/>
</reference>

<?xml version='1.0' encoding='UTF-8'?>

<reference  anchor='RFC8155' target='https://www.rfc-editor.org/info/rfc8155'>
<front>
<title>Traversal Using Relays around NAT (TURN) Server Auto Discovery</title>
<author initials='P.' surname='Patil' fullname='P. Patil'><organization /></author>
<author initials='T.' surname='Reddy' fullname='T. Reddy'><organization /></author>
<author initials='D.' surname='Wing' fullname='D. Wing'><organization /></author>
<date year='2017' month='April' />
<abstract><t>Current Traversal Using Relays around NAT (TURN) server discovery mechanisms are relatively static and limited to explicit configuration.  These are usually under the administrative control of the application or TURN service provider, and not the enterprise, ISP, or the network in which the client is located.  Enterprises and ISPs wishing to provide their own TURN servers need auto-discovery mechanisms that a TURN client could use with minimal or no configuration.  This document describes three such mechanisms for TURN server discovery.</t><t>This document updates RFC 5766 to relax the requirement for mutual authentication in certain cases.</t></abstract>
</front>
<seriesInfo name='RFC' value='8155'/>
<seriesInfo name='DOI' value='10.17487/RFC8155'/>
</reference>

<reference anchor="RFC8174" target="https://www.rfc-editor.org/info/rfc8174" xml:base="https://xml2rfc.tools.ietf.org/public/rfc/bibxml/reference.RFC.8174.xml">
<front>
<title>Ambiguity of Uppercase vs Lowercase in RFC 2119 Key Words</title>
<author initials="B." surname="Leiba" fullname="B. Leiba"><organization/></author>
<date year="2017" month="May"/>
<abstract><t>RFC 2119 specifies common key words that may be used in protocol  specifications.  This document aims to reduce the ambiguity by clarifying that only UPPERCASE usage of the key words have the  defined special meanings.</t></abstract>
</front>
<seriesInfo name="BCP" value="14"/>
<seriesInfo name="RFC" value="8174"/>
<seriesInfo name="DOI" value="10.17487/RFC8174"/>
</reference>

<?xml version='1.0' encoding='UTF-8'?>

<reference anchor='W3C.WD-html5-20110525'
           target='http://www.w3.org/TR/2011/WD-html5-20110525'>
<front>
<title>HTML5</title>

<author initials='I.' surname='Hickson' fullname='Ian Hickson'>
    <organization />
</author>

<date month='May' day='25' year='2011' />
</front>

<seriesInfo name='World Wide Web Consortium LastCall' value='WD-html5-20110525' />
<format type='HTML' target='http://www.w3.org/TR/2011/WD-html5-20110525' />
</reference>