Internet Engineering Task Force (IETF)                     H. Alvestrand
Request for Comments: 8825                                        Google
Category: Standards Track                                   January 2021
ISSN: 2070-1721

      Overview: Real-Time Protocols for Browser-Based Applications


   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".

   It intends to serve as a starting and coordination point to make sure
   that (1) all the parts that are needed to achieve this goal are
   findable and (2) the parts that belong in the Internet protocol suite
   are fully specified and on the right publication track.

   This document is an applicability statement -- it does not itself
   specify any protocol, but it specifies which other specifications
   implementations are supposed to follow to be compliant with Web Real-
   Time Communication (WebRTC).

Status of This Memo

   This is an Internet Standards Track document.

   This document is a product of the Internet Engineering Task Force
   (IETF).  It represents the consensus of the IETF community.  It has
   received public review and has been approved for publication by the
   Internet Engineering Steering Group (IESG).  Further information on
   Internet Standards is available in Section 2 of RFC 7841.

   Information about the current status of this document, any errata,
   and how to provide feedback on it may be obtained at

Copyright Notice

   Copyright (c) 2021 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

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   to this document.  Code Components extracted from this document must
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   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   1.  Introduction
   2.  Principles and Terminology
     2.1.  Goals of This Document
     2.2.  Relationship between API and Protocol
     2.3.  On Interoperability and Innovation
     2.4.  Terminology
   3.  Architecture and Functionality Groups
   4.  Data Transport
   5.  Data Framing and Securing
   6.  Data Formats
   7.  Connection Management
   8.  Presentation and Control
   9.  Local System Support Functions
   10. IANA Considerations
   11. Security Considerations
   12. References
     12.1.  Normative References
     12.2.  Informative References
   Author's Address

1.  Introduction

   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.

   The first attempts to build such applications were dependent on
   special networks, special hardware, and custom-built software, often
   at very high prices or of low quality, placing great demands on the

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

   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).

   Development of "The Universal Solution" has, however, proved hard.

   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.

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

   This memo describes a set of building blocks that (1) can be made
   accessible and controllable through a JavaScript API in a browser and
   (2) 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 WebRTC "use cases" document [RFC7478].

   Other efforts -- for instance, the W3C Web Real-Time Communications,
   Web Applications Security, and Devices and Sensors 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.

   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 the Session Description
   Protocol (SDP) -- through either (1) the endpoint asking the network
   for a SIP server [RFC3361] or (2) the transparent insertion of SIP
   Application Layer Gateways (ALGs) -- will not work with such
   signaling.  In the case of networks using cooperative endpoints, the
   approaches defined in [RFC8155] may serve as a suitable replacement
   for [RFC3361].  The increase in browser-based communications may also
   lead to a shift away from dedicated real-time-communications
   hardware, such as SIP desk phones.  This will diminish the efficacy
   of operational techniques that place dedicated real-time devices on
   their own network segment, address range, or VLAN for purposes such
   as applying traffic filtering and QoS.  Applying the markings
   described in [RFC8837] may be appropriate replacements for such

   While this document formally relies on [RFC8445], at the time of its
   publication, the majority of WebRTC implementations support the
   version of Interactive Connectivity Establishment (ICE) that is
   described in [RFC5245] and use a pre-standard version of the Trickle
   ICE mechanism described in [RFC8838].  The "ice2" attribute defined
   in [RFC8445] can be used to detect the version in use by a remote
   endpoint and to provide a smooth transition from the older
   specification to the newer one.

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

2.  Principles and Terminology

2.1.  Goals of This Document

   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

   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.

   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.

2.2.  Relationship between API and Protocol

   The total WebRTC effort consists of two major parts, each consisting
   of multiple documents:

   *  A protocol specification, done in the IETF

   *  A JavaScript API specification, defined in a series of W3C
      documents [W3C.WD-webrtc] [W3C.WD-mediacapture-streams]

   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 these specifications.
   The browser environment does not constrain the types of application
   in which this functionality can be used.

   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 the protocol

   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 are subject to constraints of the implementation, of

   The following terms are used across the documents specifying the
   WebRTC suite, with the specific meanings given here.  Not all terms
   are used in this document.  Other terms are used per their commonly
   used meanings.

   Agent:  Undefined term.  See "SDP Agent" and "ICE Agent".

   Application Programming Interface (API):  A specification of a set of
      calls and events, usually tied to a programming language or an
      abstract formal specification such as WebIDL, with its defined

   Browser:  Used synonymously with "interactive user agent" as defined
      in [HTML5].  See also the "WebRTC Browser" (aka "WebRTC User
      Agent") definition below.

   Data Channel:  An abstraction that allows data to be sent between
      WebRTC endpoints in the form of messages.  Two endpoints can have
      multiple data channels between them.

   ICE Agent:  An implementation of the Interactive Connectivity
      Establishment (ICE) protocol [RFC8445].  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 [XEP-0166]).

   Interactive:  Communication between multiple parties, where the
      expectation is that an action from one party can cause a reaction
      by another party, and the reaction can be observed by the first
      party, where the total time required for the action/reaction/
      observation is on the order of no more than hundreds of

   Media:  Audio and video content.  Not to be confused with
      "transmission media" such as wires.

   Media Path:  The path that media data follows from one WebRTC
      endpoint to another.

   Protocol:  A specification of a set of data units, their
      representation, and rules for their transmission, with their
      defined semantics.  A protocol is usually thought of as going
      between systems.

   Real-Time Media:  Media where the generation 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.

   SDP Agent:  The protocol implementation involved in the Session
      Description Protocol (SDP) offer/answer exchange, as defined in
      [RFC3264], Section 3.

   Signaling:  Communication that happens in order to establish, manage,
      and control media paths and data paths.

   Signaling Path:  The communication channels used between entities
      participating in signaling to transfer signaling.  There may be
      more entities in the signaling path than in the media path.

   WebRTC Browser (also called a "WebRTC User Agent" or "WebRTC UA"):
      Something that conforms to both the protocol specification and the
      JavaScript API cited above.

   WebRTC Non-Browser:  Something that conforms to the protocol
      specification but does not claim to implement the JavaScript API.
      This can also be called a "WebRTC device" or "WebRTC native

   WebRTC Endpoint:  Either a WebRTC browser or a WebRTC non-browser.
      It conforms to the protocol specification.

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

   WebRTC Gateway:  A WebRTC-compatible endpoint that mediates media
      traffic to non-WebRTC entities.

   All WebRTC browsers are WebRTC endpoints, so any requirement on a
   WebRTC endpoint also applies to a WebRTC browser.

   A WebRTC non-browser may be capable of hosting applications in a way
   that is similar 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, security
   considerations similar to those 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.

   WebRTC gateways are described in a separate document

2.3.  On Interoperability and Innovation

   The "Mission statement for the IETF" [RFC3935] states that "The
   benefit of a standard to the Internet is in interoperability - that
   multiple products implementing a standard are able to work together
   in order to deliver valuable functions to the Internet's users."

   Communication on the Internet frequently occurs in two phases:

   *  Two parties communicate, through some mechanism, what
      functionality they are both able to support.

   *  They use that shared communicative functionality to communicate
      or, failing to find anything in common, give up on communication.

   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.

   The goal of having a mandatory-to-implement function set is to
   prevent negotiation failure, not to preempt or prevent negotiation.

   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 you can communicate successfully as long as (1) you
   conform to a specification and (2) the other party is willing to
   accept communication at the base level of that specification.

   The alternative (that is, not having a mandatory-to-implement
   function) 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.

2.4.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "OPTIONAL" in this document are to be interpreted as described in
   BCP 14 [RFC2119] [RFC8174] when, and only when, they appear in all
   capitals, as shown here.

3.  Architecture and Functionality Groups

   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.

   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.

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

                          Figure 1: Browser Model

   Note that HTTPS and WebSockets are also offered to the JavaScript
   application through browser APIs.

   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.

   A commonly imagined model of deployment is depicted in Figure 2.
   ("JS" stands for JavaScript.)

           +-----------+                  +-----------+
           |   Web     |                  |   Web     |
           |           |                  |           |
           |           |------------------|           |
           |  Server   |  Signaling Path  |  Server   |
           |           |                  |           |
           +-----------+                  +-----------+
                /                                \
               /                                  \ Application-defined
              /                                    \ over
             /                                      \ HTTPS/WebSockets
            /  Application-defined over              \
           /   HTTPS/WebSockets                       \
          /                                            \
    +-----------+                                +-----------+
    |JS/HTML/CSS|                                |JS/HTML/CSS|
    +-----------+                                +-----------+
    +-----------+                                +-----------+
    |           |                                |           |
    |           |                                |           |
    |  Browser  |--------------------------------|  Browser  |
    |           |          Media Path            |           |
    |           |                                |           |
    +-----------+                                +-----------+

                      Figure 2: Browser RTC Trapezoid

   In 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.

   If the two web servers are operated by different entities, the inter-
   server signaling mechanism needs to be agreed upon, by either
   standardization or other means of agreement.  Existing protocols
   (e.g., SIP [RFC3261] or the Extensible Messaging and Presence
   Protocol (XMPP) [RFC6120]) could be used between servers, while
   either a standards-based or proprietary protocol could be used
   between the browser and the web server.

   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 XMPP, XMPP could be used for communication between
   XMPP servers, with either a standardized signaling mechanism (e.g.,
   XMPP over WebSockets or Bidirectional-streams Over Synchronous HTTP
   (BOSH) [XEP-0124]) or a proprietary signaling mechanism used between
   the application running in the browser and the web server.

   The choice of protocols for client-server and inter-server signaling,
   and the definition of the translation between them, are outside the
   scope of the WebRTC protocol suite described in this document.

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

   Data transport:  For example, TCP and 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.

   Data framing:  RTP, the Stream Control Transmission Protocol (SCTP),
      DTLS, and other data formats that serve as containers, and their
      functions for data confidentiality and integrity.

   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 (e.g., a session description) is

   Connection management:  For example, 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.

   Presentation and control:  What needs to happen in order to ensure
      that interactions behave in an unsurprising manner.  This can
      include floor control, screen layout, voice-activated image
      switching, and other such functions, where part of the system
      requires cooperation between parties.  Centralized Conferencing
      (XCON) [RFC6501] and Cisco/Tandberg's Telepresence
      Interoperability Protocol (TIP) were some attempts at specifying
      this kind of functionality; many applications have been built
      without standardized interfaces to these functions.

   Local system support functions:  Functions that need not be specified
      uniformly, because each participant may implement these functions
      as they choose, 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.

   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.
   The ability to communicate globally is helped by both (1) having core
   specifications be unencumbered by IPR issues and (2) having the
   formats and protocols be fully enough specified to allow for
   independent implementation.

   One can think of the first three groups as forming a "media transport
   infrastructure" and of the last three 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.

4.  Data Transport

   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 Traversal
   Using Relays around NAT (TURN) relays).

   It includes necessary functions for congestion control,
   retransmission, and in-order delivery.

   WebRTC endpoints MUST implement the transport protocols described in

5.  Data Framing and Securing

   The format for media transport is RTP [RFC3550].  Implementation of
   the Secure Real-time Transport Protocol (SRTP) [RFC3711] is REQUIRED
   for all implementations.

   The detailed considerations for usage of functions from RTP and SRTP
   are given in [RFC8834].  The security considerations for the WebRTC
   use case are provided in [RFC8826], and the resulting security
   functions are described in [RFC8827].

   Considerations for the transfer of data that is not in RTP format are
   described in [RFC8831], and a supporting protocol for establishing
   individual data channels is described in [RFC8832].  WebRTC endpoints
   MUST implement these two specifications.

   WebRTC endpoints MUST implement [RFC8834], [RFC8826], [RFC8827], and
   the requirements they include.

6.  Data Formats

   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 implementer.

   WebRTC endpoints that support audio and/or video MUST implement the
   codecs and profiles required in [RFC7874] and [RFC7742].

7.  Connection Management

   The methods, mechanisms, and requirements for setting up,
   negotiating, and tearing down connections comprise a large subject,
   and one where it is desirable to have both interoperability and
   freedom to innovate.

   The following principles apply:

   1.  The WebRTC media negotiations will be capable of representing the
       same SDP offer/answer semantics [RFC3264] that are used in SIP,
       in such a way that it is possible to build a signaling gateway
       between SIP and the WebRTC media negotiation.

   2.  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 and the
       SIP signaling may be needed.

   3.  When an SDP for a new codec is specified, no other
       standardization should be required for it to be possible to use
       that codec in the web browsers.  Adding new codecs that might
       have new SDP parameters should not change the APIs between the
       browser and the 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 JavaScript

   The particular choices made for WebRTC, and their implications for
   the API offered by a browser implementing WebRTC, are described in

   WebRTC browsers MUST implement [RFC8829].

   WebRTC endpoints MUST implement those functions described in
   [RFC8829] that relate to the network layer (e.g., BUNDLE [RFC8843],
   "rtcp-mux" [RFC5761], and Trickle ICE [RFC8838]), but these endpoints
   do not need to support the API functionality described in [RFC8829].

8.  Presentation and Control

   The most important part of control is the users' control over the
   browser's interaction with input/output devices and communications
   channels.  It is important that the users have some way of figuring
   out where their 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
   [W3C.WD-webrtc] and the media capture API

   WebRTC browsers MUST implement these two specifications.

9.  Local System Support Functions

   These functions are characterized by the fact that the quality of an
   implementation strongly influences 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.

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

   One would want to see certain parts of the system conform to certain
   properties; for instance:

   *  Echo cancellation should be good enough to achieve the suppression
      of acoustical feedback loops below a perceptually noticeable

   *  Privacy concerns MUST be satisfied; for instance, if remote
      control of a 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.

   *  Automatic Gain Control (AGC), if present, should normalize a
      speaking voice into a reasonable dB range.

   The requirements on WebRTC systems with regard to audio processing
   are found in [RFC7874], and that document includes more guidance
   about echo cancellation and AGC; the APIs for control of local
   devices are found in [W3C.WD-mediacapture-streams].

   WebRTC endpoints MUST implement the processing functions in
   [RFC7874].  (Together with the requirement in Section 6, this means
   that WebRTC endpoints MUST implement the whole document.)

10.  IANA Considerations

   This document has no IANA actions.

11.  Security Considerations

   Security of the web-enabled real-time communications comes in several

   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.

   Security of the communication channels:  It should be easy for
      participants to reassure themselves of the security of their
      communication -- by verifying the crypto parameters of the links
      that they participate in, and to get reassurances from the other
      parties to the communication that those parties promise that
      appropriate measures are taken.

   Security of the partners' identities:  Verifying that the
      participants are who they say they are (when positive
      identification is appropriate) or that their identities cannot be
      uncovered (when anonymity is a goal of the application).

   The security analysis, and the requirements derived from that
   analysis, are contained in [RFC8826].

   It is also important to read the security sections of
   [W3C.WD-mediacapture-streams] and [W3C.WD-webrtc].

12.  References

12.1.  Normative References

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119,
              DOI 10.17487/RFC2119, March 1997,

   [RFC3264]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
              with Session Description Protocol (SDP)", RFC 3264,
              DOI 10.17487/RFC3264, June 2002,

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
              July 2003, <https://www.rfc-editor.org/info/rfc3550>.

   [RFC3711]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
              Norrman, "The Secure Real-time Transport Protocol (SRTP)",
              RFC 3711, DOI 10.17487/RFC3711, March 2004,

   [RFC7742]  Roach, A.B., "WebRTC Video Processing and Codec
              Requirements", RFC 7742, DOI 10.17487/RFC7742, March 2016,

   [RFC7874]  Valin, JM. and C. Bran, "WebRTC Audio Codec and Processing
              Requirements", RFC 7874, DOI 10.17487/RFC7874, May 2016,

   [RFC8174]  Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
              2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
              May 2017, <https://www.rfc-editor.org/info/rfc8174>.

   [RFC8445]  Keranen, A., Holmberg, C., and J. Rosenberg, "Interactive
              Connectivity Establishment (ICE): A Protocol for Network
              Address Translator (NAT) Traversal", RFC 8445,
              DOI 10.17487/RFC8445, July 2018,

   [RFC8826]  Rescorla, E., "Security Considerations for WebRTC",
              RFC 8826, DOI 10.17487/RFC8826, January 2021,

   [RFC8827]  Rescorla, E., "WebRTC Security Architecture", RFC 8827,
              DOI 10.17487/RFC8827, January 2021,

   [RFC8829]  Uberti, J., Jennings, C., and E. Rescorla, Ed.,
              "JavaScript Session Establishment Protocol (JSEP)",
              RFC 8829, DOI 10.17487/RFC8829, January 2021,

   [RFC8831]  Jesup, R., Loreto, S., and M. Tüxen, "WebRTC Data
              Channels", RFC 8831, DOI 10.17487/RFC8831, January 2021,

   [RFC8832]  Jesup, R., Loreto, S., and M. Tüxen, "WebRTC Data Channel
              Establishment Protocol", RFC 8832, DOI 10.17487/RFC8832,
              January 2021, <https://www.rfc-editor.org/info/rfc8832>.

   [RFC8834]  Perkins, C., Westerlund, M., and J. Ott, "Media Transport
              and Use of RTP in WebRTC", RFC 8834, DOI 10.17487/RFC8834,
              January 2021, <https://www.rfc-editor.org/info/rfc8834>.

   [RFC8835]  Alvestrand, H., "Transports for WebRTC", RFC 8835,
              DOI 10.17487/RFC8835, January 2021,

              Jennings, C., Aboba, B., Bruaroey, J-I., and H. Boström,
              "Media Capture and Streams", W3C Candidate Recommendation,

              Jennings, C., Boström, H., and J-I. Bruaroey, "WebRTC 1.0:
              Real-time Communication Between Browsers", W3C Proposed
              Recommendation, <https://www.w3.org/TR/webrtc/>.

12.2.  Informative References

   [HTML5]    WHATWG, "HTML - Living Standard", January 2021,

   [RFC3261]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
              A., Peterson, J., Sparks, R., Handley, M., and E.
              Schooler, "SIP: Session Initiation Protocol", RFC 3261,
              DOI 10.17487/RFC3261, June 2002,

   [RFC3361]  Schulzrinne, H., "Dynamic Host Configuration Protocol
              (DHCP-for-IPv4) Option for Session Initiation Protocol
              (SIP) Servers", RFC 3361, DOI 10.17487/RFC3361, August
              2002, <https://www.rfc-editor.org/info/rfc3361>.

   [RFC3935]  Alvestrand, H., "A Mission Statement for the IETF",
              BCP 95, RFC 3935, DOI 10.17487/RFC3935, October 2004,

   [RFC5245]  Rosenberg, J., "Interactive Connectivity Establishment
              (ICE): A Protocol for Network Address Translator (NAT)
              Traversal for Offer/Answer Protocols", RFC 5245,
              DOI 10.17487/RFC5245, April 2010,

   [RFC5761]  Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
              Control Packets on a Single Port", RFC 5761,
              DOI 10.17487/RFC5761, April 2010,

   [RFC6120]  Saint-Andre, P., "Extensible Messaging and Presence
              Protocol (XMPP): Core", RFC 6120, DOI 10.17487/RFC6120,
              March 2011, <https://www.rfc-editor.org/info/rfc6120>.

   [RFC6501]  Novo, O., Camarillo, G., Morgan, D., and J. Urpalainen,
              "Conference Information Data Model for Centralized
              Conferencing (XCON)", RFC 6501, DOI 10.17487/RFC6501,
              March 2012, <https://www.rfc-editor.org/info/rfc6501>.

   [RFC7478]  Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
              Time Communication Use Cases and Requirements", RFC 7478,
              DOI 10.17487/RFC7478, March 2015,

   [RFC8155]  Patil, P., Reddy, T., and D. Wing, "Traversal Using Relays
              around NAT (TURN) Server Auto Discovery", RFC 8155,
              DOI 10.17487/RFC8155, April 2017,

   [RFC8837]  Jones, P., Dhesikan, S., Jennings, C., and D. Druta,
              "Differentiated Services Code Point (DSCP) Packet Markings
              for WebRTC QoS", RFC 8837, DOI 10.17487/RFC8837, January
              2021, <https://www.rfc-editor.org/info/rfc8837>.

   [RFC8838]  Ivov, E., Uberti, J., and P. Saint-Andre, "Trickle ICE:
              Incremental Provisioning of Candidates for the Interactive
              Connectivity Establishment (ICE) Protocol", RFC 8838,
              DOI 10.17487/RFC8838, January 2021,

   [RFC8843]  Holmberg, C., Alvestrand, H., and C. Jennings,
              "Negotiating Media Multiplexing Using the Session
              Description Protocol (SDP)", RFC 8843,
              DOI 10.17487/RFC8843, January 2021,

              Alvestrand, H. and U. Rauschenbach, "WebRTC Gateways",
              Work in Progress, Internet-Draft, draft-ietf-rtcweb-
              gateways-02, 21 January 2016,

   [XEP-0124] Paterson, I., Smith, D., Saint-Andre, P., Moffitt, J.,
              Stout, L., and W. Tilanus, "Bidirectional-streams Over
              Synchronous HTTP (BOSH)", XSF XEP 0124, November 2016,

   [XEP-0166] Ludwig, S., Beda, J., Saint-Andre, P., McQueen, R., Egan,
              S., and J. Hildebrand, "Jingle", XSF XEP 0166, September
              2018, <https://xmpp.org/extensions/xep-0166.html>.


   The number of people who have taken part in the discussions
   surrounding this document are too numerous to list, or even to
   identify.  The people listed below have made special, identifiable
   contributions; this does not mean that others' contributions are less

   Thanks to Cary Bran, Cullen Jennings, Colin Perkins, Magnus
   Westerlund, and Jörg Ott, who offered technical contributions to
   various draft versions of this document.

   Thanks to Jonathan Rosenberg, Matthew Kaufman, and others at Skype
   for the ASCII drawings in Section 3.

   Thanks to Alissa Cooper, Björn Höhrmann, 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.

Author's Address

   Harald T. Alvestrand
   Kungsbron 2
   SE-11122 Stockholm

   Email: harald@alvestrand.no