draft-ietf-rtcweb-overview-19.txt   rfc8825.txt 
Network Working Group H. Alvestrand Internet Engineering Task Force (IETF) H. Alvestrand
Internet-Draft Google Request for Comments: 8825 Google
Intended status: Standards Track November 12, 2017 Category: Standards Track January 2021
Expires: May 16, 2018 ISSN: 2070-1721
Overview: Real Time Protocols for Browser-based Applications Overview: Real-Time Protocols for Browser-Based Applications
draft-ietf-rtcweb-overview-19
Abstract Abstract
This document gives an overview and context of a protocol suite This document gives an overview and context of a protocol suite
intended for use with real-time applications that can be deployed in intended for use with real-time applications that can be deployed in
browsers - "real time communication on the Web". browsers -- "real-time communication on the Web".
It intends to serve as a starting and coordination point to make sure 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 (1) all the parts that are needed to achieve this goal are
that the parts that belong in the Internet protocol suite are fully findable and (2) the parts that belong in the Internet protocol suite
specified and on the right publication track. are fully specified and on the right publication track.
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.
This document is a work item of the RTCWEB working group. 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 Status of This Memo
This Internet-Draft is submitted in full conformance with the This is an Internet Standards Track document.
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at https://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months This document is a product of the Internet Engineering Task Force
and may be updated, replaced, or obsoleted by other documents at any (IETF). It represents the consensus of the IETF community. It has
time. It is inappropriate to use Internet-Drafts as reference received public review and has been approved for publication by the
material or to cite them other than as "work in progress." Internet Engineering Steering Group (IESG). Further information on
Internet Standards is available in Section 2 of RFC 7841.
This Internet-Draft will expire on May 16, 2018. Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
https://www.rfc-editor.org/info/rfc8825.
Copyright Notice Copyright Notice
Copyright (c) 2017 IETF Trust and the persons identified as the Copyright (c) 2021 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
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described in the Simplified BSD License. described in the Simplified BSD License.
Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3 1. Introduction
2. Principles and Terminology . . . . . . . . . . . . . . . . . 4 2. Principles and Terminology
2.1. Goals of this document . . . . . . . . . . . . . . . . . 4 2.1. Goals of This Document
2.2. Relationship between API and protocol . . . . . . . . . . 5 2.2. Relationship between API and Protocol
2.3. On interoperability and innovation . . . . . . . . . . . 7 2.3. On Interoperability and Innovation
2.4. Terminology . . . . . . . . . . . . . . . . . . . . . . . 8 2.4. Terminology
3. Architecture and Functionality groups . . . . . . . . . . . . 8 3. Architecture and Functionality Groups
4. Data transport . . . . . . . . . . . . . . . . . . . . . . . 12 4. Data Transport
5. Data framing and securing . . . . . . . . . . . . . . . . . . 13 5. Data Framing and Securing
6. Data formats . . . . . . . . . . . . . . . . . . . . . . . . 13 6. Data Formats
7. Connection management . . . . . . . . . . . . . . . . . . . . 13 7. Connection Management
8. Presentation and control . . . . . . . . . . . . . . . . . . 14 8. Presentation and Control
9. Local system support functions . . . . . . . . . . . . . . . 14 9. Local System Support Functions
10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 15 10. IANA Considerations
11. Security Considerations . . . . . . . . . . . . . . . . . . . 15 11. Security Considerations
12. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 16 12. References
13. References . . . . . . . . . . . . . . . . . . . . . . . . . 16 12.1. Normative References
13.1. Normative References . . . . . . . . . . . . . . . . . . 16 12.2. Informative References
13.2. Informative References . . . . . . . . . . . . . . . . . 18 Acknowledgements
Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 20 Author's Address
A.1. Changes from draft-alvestrand-dispatch-rtcweb-datagram-00
to -01 . . . . . . . . . . . . . . . . . . . . . . . . . 20
A.2. Changes from draft-alvestrand-dispatch-01 to draft-
alvestrand-rtcweb-overview-00 . . . . . . . . . . . . . . 20
A.3. Changes from draft-alvestrand-rtcweb-00 to -01 . . . . . 20
A.4. Changes from draft-alvestrand-rtcweb-overview-01 to
draft-ietf-rtcweb-overview-00 . . . . . . . . . . . . . . 21
A.5. Changes from -00 to -01 of draft-ietf-rtcweb-overview . . 21
A.6. Changes from -01 to -02 of draft-ietf-rtcweb-overview . . 21
A.7. Changes from -02 to -03 of draft-ietf-rtcweb-overview . . 21
A.8. Changes from -03 to -04 of draft-ietf-rtcweb-overview . . 22
A.9. Changes from -04 to -05 of draft-ietf-rtcweb-overview . . 22
A.10. Changes from -05 to -06 . . . . . . . . . . . . . . . . . 22
A.11. Changes from -06 to -07 . . . . . . . . . . . . . . . . . 22
A.12. Changes from -07 to -08 . . . . . . . . . . . . . . . . . 22
A.13. Changes from -08 to -09 . . . . . . . . . . . . . . . . . 22
A.14. Changes from -09 to -10 . . . . . . . . . . . . . . . . . 22
A.15. Changes from -10 to -11 . . . . . . . . . . . . . . . . . 23
A.16. Changes from -11 to -12 . . . . . . . . . . . . . . . . . 23
A.17. Changes from -12 to -13 . . . . . . . . . . . . . . . . . 23
A.18. Changes from -13 to -14 . . . . . . . . . . . . . . . . . 23
A.19. Changes from -14 to -15 . . . . . . . . . . . . . . . . . 23
A.20. Changes from -15 to -16 . . . . . . . . . . . . . . . . . 23
A.21. Changes from -16 to -17 . . . . . . . . . . . . . . . . . 24
A.22. Changes from -17 to -18 . . . . . . . . . . . . . . . . . 24
A.23. Changes from -18 to -19 . . . . . . . . . . . . . . . . . 24
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 24
1. Introduction 1. Introduction
The Internet was, from very early in its lifetime, considered a The Internet was, from very early in its lifetime, considered a
possible vehicle for the deployment of real-time, interactive possible vehicle for the deployment of real-time, interactive
applications - with the most easily imaginable being audio applications -- with the most easily imaginable being audio
conversations (aka "Internet telephony") and video conferencing. conversations (aka "Internet telephony") and video conferencing.
The first attempts to build this were dependent on special networks, The first attempts to build such applications were dependent on
special hardware and custom-built software, often at very high prices special networks, special hardware, and custom-built software, often
or at low quality, placing great demands on the infrastructure. at very high prices or of low quality, placing great demands on the
infrastructure.
As the available bandwidth has increased, and as processors and other As the available bandwidth has increased, and as processors and other
hardware has become ever faster, the barriers to participation have hardware have become ever faster, the barriers to participation have
decreased, and it has become possible to deliver a satisfactory decreased, and it has become possible to deliver a satisfactory
experience on commonly available computing hardware. experience on commonly available computing hardware.
Still, there are a number of barriers to the ability to communicate 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 universally -- one of these is that there is, as of yet, no single
of communication protocols that all agree should be made available set of communication protocols that all agree should be made
for communication; another is the sheer lack of universal available for communication; another is the sheer lack of universal
identification systems (such as is served by telephone numbers or identification systems (such as is served by telephone numbers or
email addresses in other communications systems). email addresses in other communications systems).
Development of The Universal Solution has, however, proved hard. Development of "The Universal Solution" has, however, proved hard.
The last few years have also seen a new platform rise for deployment The last few years have also seen a new platform rise for deployment
of services: The browser-embedded application, or "Web application". of services: the browser-embedded application, or "web application".
It turns out that as long as the browser platform has the necessary 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 interfaces, it is possible to deliver almost any kind of service
it. on it.
Traditionally, these interfaces have been delivered by plugins, which Traditionally, these interfaces have been delivered by plugins, which
had to be downloaded and installed separately from the browser; in had to be downloaded and installed separately from the browser; in
the development of HTML5, application developers see much promise in the development of HTML5 [HTML5], application developers see much
the possibility of making those interfaces available in a promise in the possibility of making those interfaces available in a
standardized way within the browser. standardized way within the browser.
This memo describes a set of building blocks that can be made This memo describes a set of building blocks that (1) can be made
accessible and controllable through a Javascript API in a browser, accessible and controllable through a JavaScript API in a browser and
and which together form a sufficient set of functions to allow the (2) together form a sufficient set of functions to allow the use of
use of interactive audio and video in applications that communicate interactive audio and video in applications that communicate directly
directly between browsers across the Internet. The resulting between browsers across the Internet. The resulting protocol suite
protocol suite is intended to enable all the applications that are is intended to enable all the applications that are described as
described as required scenarios in the use cases document [RFC7478]. required scenarios in the WebRTC "use cases" document [RFC7478].
Other efforts, for instance the W3C Web Real-Time Communications, Web Other efforts -- for instance, the W3C Web Real-Time Communications,
Applications Security, and Device and Sensor working groups, focus on Web Applications Security, and Devices and Sensors Working Groups --
making standardized APIs and interfaces available, within or focus on making standardized APIs and interfaces available, within or
alongside the HTML5 effort, for those functions. This memo alongside the HTML5 effort, for those functions. This memo
concentrates on specifying the protocols and subprotocols that are concentrates on specifying the protocols and subprotocols that are
needed to specify the interactions over the network. needed to specify the interactions over the network.
Operators should note that deployment of WebRTC will result in a Operators should note that deployment of WebRTC will result in a
change in the nature of signaling for real time media on the network, 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 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 consume such media. In the case of signaling, WebRTC session setup
will typically occur over TLS-secured web technologies using will typically occur over TLS-secured web technologies using
application-specific protocols. Operational techniques that involve application-specific protocols. Operational techniques that involve
inserting network elements to interpret SDP -- either through inserting network elements to interpret the Session Description
endpoint cooperation [RFC3361] or through the transparent insertion Protocol (SDP) -- through either (1) the endpoint asking the network
of SIP Application Level Gateways (ALGs) -- will not work with such 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 signaling. In the case of networks using cooperative endpoints, the
approaches defined in [RFC8155] may serve as a suitable replacement approaches defined in [RFC8155] may serve as a suitable replacement
for [RFC3361]. The increase in browser-based communications may also for [RFC3361]. The increase in browser-based communications may also
lead to a shift away from dedicated real-time-communications lead to a shift away from dedicated real-time-communications
hardware, such as SIP desk phones. This will diminish the efficacy hardware, such as SIP desk phones. This will diminish the efficacy
of operational techniques that place dedicated real-time devices on of operational techniques that place dedicated real-time devices on
their own network segment, address range, or VLAN for purposes such their own network segment, address range, or VLAN for purposes such
as applying traffic filtering and QoS. Applying the markings as applying traffic filtering and QoS. Applying the markings
described in [I-D.ietf-tsvwg-rtcweb-qos] may be appropriate described in [RFC8837] may be appropriate replacements for such
replacements for such techniques. techniques.
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 This memo uses the term "WebRTC" (note the case used) to refer to the
overall effort consisting of both IETF and W3C efforts. overall effort consisting of both IETF and W3C efforts.
2. Principles and Terminology 2. Principles and Terminology
2.1. Goals of this document 2.1. Goals of This Document
The goal of the WebRTC protocol specification is to specify a set of The goal of the WebRTC protocol specification is to specify a set of
protocols that, if all are implemented, will allow an implementation protocols that, if all are implemented, will allow an implementation
to communicate with another implementation using audio, video and to communicate with another implementation using audio, video, and
data sent along the most direct possible path between the data sent along the most direct possible path between the
participants. participants.
This document is intended to serve as the roadmap to the WebRTC This document is intended to serve as the roadmap to the WebRTC
specifications. It defines terms used by other parts of the WebRTC specifications. It defines terms used by other parts of the WebRTC
protocol specifications, lists references to other specifications protocol specifications, lists references to other specifications
that don't need further elaboration in the WebRTC context, and gives that don't need further elaboration in the WebRTC context, and gives
pointers to other documents that form part of the WebRTC suite. pointers to other documents that form part of the WebRTC suite.
By reading this document and the documents it refers to, it should be By reading this document and the documents it refers to, it should be
possible to have all information needed to implement a WebRTC possible to have all information needed to implement a WebRTC-
compatible implementation. compatible implementation.
2.2. Relationship between API and protocol 2.2. Relationship between API and Protocol
The total WebRTC effort consists of two major parts, each consisting The total WebRTC effort consists of two major parts, each consisting
of multiple documents: of multiple documents:
o A protocol specification, done in the IETF * A protocol specification, done in the IETF
o A Javascript API specification, defined in a series of W3C * A JavaScript API specification, defined in a series of W3C
documents documents [W3C.WD-webrtc] [W3C.WD-mediacapture-streams]
[W3C.WD-webrtc-20120209][W3C.WD-mediacapture-streams-20120628]
Together, these two specifications aim to provide an environment Together, these two specifications aim to provide an environment
where Javascript embedded in any page, when suitably authorized by where JavaScript embedded in any page, when suitably authorized by
its user, is able to set up communication using audio, video and its user, is able to set up communication using audio, video, and
auxiliary data, as long as the browser supports this specification. auxiliary data, as long as the browser supports these specifications.
The browser environment does not constrain the types of application The browser environment does not constrain the types of application
in which this functionality can be used. in which this functionality can be used.
The protocol specification does not assume that all implementations The protocol specification does not assume that all implementations
implement this API; it is not intended to be necessary for implement this API; it is not intended to be necessary for
interoperation to know whether the entity one is communicating with interoperation to know whether the entity one is communicating with
is a browser or another device implementing this specification. is a browser or another device implementing the protocol
specification.
The goal of cooperation between the protocol specification and the The goal of cooperation between the protocol specification and the
API specification is that for all options and features of the API specification is that for all options and features of the
protocol specification, it should be clear which API calls to make to protocol specification, it should be clear which API calls to make to
exercise that option or feature; similarly, for any sequence of API exercise that option or feature; similarly, for any sequence of API
calls, it should be clear which protocol options and features will be calls, it should be clear which protocol options and features will be
invoked. Both subject to constraints of the implementation, of invoked. Both are subject to constraints of the implementation, of
course. course.
The following terms are used across the documents specifying the The following terms are used across the documents specifying the
WebRTC suite, in the specific meanings given here. Not all terms are WebRTC suite, with the specific meanings given here. Not all terms
used in this document. Other terms are used in their commonly used are used in this document. Other terms are used per their commonly
meaning. used meanings.
Agent: Undefined term. See "SDP Agent" and "ICE Agent". Agent: Undefined term. See "SDP Agent" and "ICE Agent".
Application Programming Interface (API): A specification of a set of Application Programming Interface (API): A specification of a set of
calls and events, usually tied to a programming language or an calls and events, usually tied to a programming language or an
abstract formal specification such as WebIDL, with its defined abstract formal specification such as WebIDL, with its defined
semantics. semantics.
Browser: Used synonymously with "Interactive User Agent" as defined Browser: Used synonymously with "interactive user agent" as defined
in the HTML specification [W3C.WD-html5-20110525]. See also in [HTML5]. See also the "WebRTC Browser" (aka "WebRTC User
"WebRTC User Agent". Agent") definition below.
Data Channel: An abstraction that allows data to be sent between Data Channel: An abstraction that allows data to be sent between
WebRTC endpoints in the form of messages. Two endpoints can have WebRTC endpoints in the form of messages. Two endpoints can have
multiple data channels between them. multiple data channels between them.
ICE Agent: An implementation of the Interactive Connectivity ICE Agent: An implementation of the Interactive Connectivity
Establishment (ICE) [RFC5245] protocol. An ICE Agent may also be Establishment (ICE) protocol [RFC8445]. An ICE Agent may also be
an SDP Agent, but there exist ICE Agents that do not use SDP (for an SDP Agent, but there exist ICE Agents that do not use SDP (for
instance those that use Jingle [XEP-0166]). instance, those that use Jingle [XEP-0166]).
Interactive: Communication between multiple parties, where the Interactive: Communication between multiple parties, where the
expectation is that an action from one party can cause a reaction expectation is that an action from one party can cause a reaction
by another party, and the reaction can be observed by the first by another party, and the reaction can be observed by the first
party, with the total time required for the action/reaction/ party, where the total time required for the action/reaction/
observation is on the order of no more than hundreds of observation is on the order of no more than hundreds of
milliseconds. milliseconds.
Media: Audio and video content. Not to be confused with Media: Audio and video content. Not to be confused with
"transmission media" such as wires. "transmission media" such as wires.
Media Path: The path that media data follows from one WebRTC Media Path: The path that media data follows from one WebRTC
endpoint to another. endpoint to another.
Protocol: A specification of a set of data units, their Protocol: A specification of a set of data units, their
representation, and rules for their transmission, with their representation, and rules for their transmission, with their
defined semantics. A protocol is usually thought of as going defined semantics. A protocol is usually thought of as going
between systems. between systems.
Real-time Media: Media where generation of content and display of Real-Time Media: Media where the generation and display of content
content are intended to occur closely together in time (on the are intended to occur closely together in time (on the order of no
order of no more than hundreds of milliseconds). Real-time media more than hundreds of milliseconds). Real-time media can be used
can be used to support interactive communication. to support interactive communication.
SDP Agent: The protocol implementation involved in the Session SDP Agent: The protocol implementation involved in the Session
Description Protocol (SDP) offer/answer exchange, as defined in Description Protocol (SDP) offer/answer exchange, as defined in
[RFC3264] section 3. [RFC3264], Section 3.
Signaling: Communication that happens in order to establish, manage Signaling: Communication that happens in order to establish, manage,
and control media paths and data paths. and control media paths and data paths.
Signaling Path: The communication channels used between entities Signaling Path: The communication channels used between entities
participating in signaling to transfer signaling. There may be participating in signaling to transfer signaling. There may be
more entities in the signaling path than in the media path. more entities in the signaling path than in the media path.
WebRTC Browser: (also called a WebRTC User Agent or WebRTC UA) WebRTC Browser (also called a "WebRTC User Agent" or "WebRTC UA"):
Something that conforms to both the protocol specification and the Something that conforms to both the protocol specification and the
Javascript API cited above. JavaScript API cited above.
WebRTC non-Browser: Something that conforms to the protocol WebRTC Non-Browser: Something that conforms to the protocol
specification, but does not claim to implement the Javascript API. specification but does not claim to implement the JavaScript API.
This can also be called a "WebRTC device" or "WebRTC native This can also be called a "WebRTC device" or "WebRTC native
application". application".
WebRTC Endpoint: Either a WebRTC browser or a WebRTC non-browser. WebRTC Endpoint: Either a WebRTC browser or a WebRTC non-browser.
It conforms to the protocol specification. It conforms to the protocol specification.
WebRTC-compatible Endpoint: An endpoint that is able to successfully WebRTC-Compatible Endpoint: An endpoint that is able to successfully
communicate with a WebRTC endpoint, but may fail to meet some communicate with a WebRTC endpoint but may fail to meet some
requirements of a WebRTC endpoint. This may limit where in the requirements of a WebRTC endpoint. This may limit where in the
network such an endpoint can be attached, or may limit the network such an endpoint can be attached or may limit the security
security guarantees that it offers to others. It is not guarantees that it offers to others. It is not constrained by
constrained by this specification; when it is mentioned at all, it this specification; when it is mentioned at all, it is to note the
is to note the implications on WebRTC-compatible endpoints of the implications on WebRTC-compatible endpoints of the requirements
requirements placed on WebRTC endpoints. placed on WebRTC endpoints.
WebRTC Gateway: A WebRTC-compatible endpoint that mediates media WebRTC Gateway: A WebRTC-compatible endpoint that mediates media
traffic to non-WebRTC entities. traffic to non-WebRTC entities.
All WebRTC browsers are WebRTC endpoints, so any requirement on a All WebRTC browsers are WebRTC endpoints, so any requirement on a
WebRTC endpoint also applies to a WebRTC browser. WebRTC endpoint also applies to a WebRTC browser.
A WebRTC non-browser may be capable of hosting applications in a A WebRTC non-browser may be capable of hosting applications in a way
similar way to the way in which a browser can host Javascript that is similar to the way in which a browser can host JavaScript
applications, typically by offering APIs in other languages. For applications, typically by offering APIs in other languages. For
instance it may be implemented as a library that offers a C++ API instance, it may be implemented as a library that offers a C++ API
intended to be loaded into applications. In this case, similar intended to be loaded into applications. In this case, security
security considerations as for Javascript may be needed; however, considerations similar to those for JavaScript may be needed;
since such APIs are not defined or referenced here, this document however, since such APIs are not defined or referenced here, this
cannot give any specific rules for those interfaces. document cannot give any specific rules for those interfaces.
WebRTC gateways are described in a separate document, WebRTC gateways are described in a separate document
[I-D.ietf-rtcweb-gateways]. [WebRTC-Gateways].
2.3. On interoperability and innovation 2.3. On Interoperability and Innovation
The "Mission statement of the IETF" [RFC3935] states that "The The "Mission statement for the IETF" [RFC3935] states that "The
benefit of a standard to the Internet is in interoperability - that benefit of a standard to the Internet is in interoperability - that
multiple products implementing a standard are able to work together multiple products implementing a standard are able to work together
in order to deliver valuable functions to the Internet's users." in order to deliver valuable functions to the Internet's users."
Communication on the Internet frequently occurs in two phases: Communication on the Internet frequently occurs in two phases:
o Two parties communicate, through some mechanism, what * Two parties communicate, through some mechanism, what
functionality they both are able to support functionality they are both able to support.
o They use that shared communicative functionality to communicate, * They use that shared communicative functionality to communicate
or, failing to find anything in common, give up on communication. or, failing to find anything in common, give up on communication.
There are often many choices that can be made for communicative There are often many choices that can be made for communicative
functionality; the history of the Internet is rife with the proposal, functionality; the history of the Internet is rife with the proposal,
standardization, implementation, and success or failure of many types standardization, implementation, and success or failure of many types
of options, in all sorts of protocols. of options, in all sorts of protocols.
The goal of having a mandatory to implement function set is to The goal of having a mandatory-to-implement function set is to
prevent negotiation failure, not to preempt or prevent negotiation. prevent negotiation failure, not to preempt or prevent negotiation.
The presence of a mandatory to implement function set serves as a The presence of a mandatory-to-implement function set serves as a
strong changer of the marketplace of deployment - in that it gives 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 guarantee that you can communicate successfully as long as (1) you
other party is willing to accept communication at the base level of conform to a specification and (2) the other party is willing to
that specification, you can communicate successfully. accept communication at the base level of that specification.
The alternative, that is having no mandatory to implement, does not The alternative (that is, not having a mandatory-to-implement
mean that you cannot communicate, it merely means that in order to be function) does not mean that you cannot communicate; it merely means
part of the communications partnership, you have to implement the that in order to be part of the communications partnership, you have
standard "and then some". The "and then some" is usually called a to implement the standard "and then some". The "and then some" is
profile of some sort; in the version most antithetical to the usually called a profile of some sort; in the version most
Internet ethos, that "and then some" consists of having to use a antithetical to the Internet ethos, that "and then some" consists of
specific vendor's product only. having to use a specific vendor's product only.
2.4. Terminology 2.4. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
document are to be interpreted as described in [RFC2119]. "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 3. Architecture and Functionality Groups
For browser-based applications, the model for real-time support does For browser-based applications, the model for real-time support does
not assume that the browser will contain all the functions needed for not assume that the browser will contain all the functions needed for
an application such as a telephone or a video conference. The vision an application such as a telephone or a video conference. The vision
is that the browser will have the functions needed for a Web is that the browser will have the functions needed for a web
application, working in conjunction with its backend servers, to application, working in conjunction with its backend servers, to
implement these functions. implement these functions.
This means that two vital interfaces need specification: The This means that two vital interfaces need specification: the
protocols that browsers use to talk to each other, without any protocols that browsers use to talk to each other, without any
intervening servers, and the APIs that are offered for a Javascript intervening servers; and the APIs that are offered for a JavaScript
application to take advantage of the browser's functionality. application to take advantage of the browser's functionality.
+------------------------+ On-the-wire +------------------------+ On-the-wire
| | Protocols | | Protocols
| Servers |---------> | Servers |--------->
| | | |
| | | |
+------------------------+ +------------------------+
^ ^
| |
| |
| HTTPS/ | HTTPS/
| WebSockets | WebSockets
| |
| |
+----------------------------+ +----------------------------+
| Javascript/HTML/CSS | | JavaScript/HTML/CSS |
+----------------------------+ +----------------------------+
Other ^ ^ RTC Other ^ ^ RTC
APIs | | APIs APIs | | APIs
+---|-----------------|------+ +---|-----------------|------+
| | | | | | | |
| +---------+| | +---------+|
| | Browser || On-the-wire | | Browser || On-the-wire
| Browser | RTC || Protocols | Browser | RTC || Protocols
| | Function|-----------> | | Function|----------->
| | || | | ||
| | || | | ||
| +---------+| | +---------+|
+---------------------|------+ +---------------------|------+
| |
V V
Native OS Services Native OS Services
Figure 1: Browser Model Figure 1: Browser Model
Note that HTTPS and WebSockets are also offered to the Javascript Note that HTTPS and WebSockets are also offered to the JavaScript
application through browser APIs. application through browser APIs.
As for all protocol and API specifications, there is no restriction As for all protocol and API specifications, there is no restriction
that the protocols can only be used to talk to another browser; since that the protocols can only be used to talk to another browser; since
they are fully specified, any endpoint that implements the protocols they are fully specified, any endpoint that implements the protocols
faithfully should be able to interoperate with the application faithfully should be able to interoperate with the application
running in the browser. running in the browser.
A commonly imagined model of deployment is the one depicted below. A commonly imagined model of deployment is depicted in Figure 2.
In the figure below JS is Javascript. ("JS" stands for JavaScript.)
+-----------+ +-----------+ +-----------+ +-----------+
| Web | | Web | | Web | | Web |
| | Signaling | | | | | |
| |-------------| | | |------------------| |
| Server | path | Server | | Server | Signaling Path | Server |
| | | | | | | |
+-----------+ +-----------+ +-----------+ +-----------+
/ \ / \
/ \ Application-defined / \ Application-defined
/ \ over / \ over
/ \ HTTPS/WebSockets / \ HTTPS/WebSockets
/ Application-defined over \ / Application-defined over \
/ HTTPS/WebSockets \ / HTTPS/WebSockets \
/ \ / \
+-----------+ +-----------+ +-----------+ +-----------+
|JS/HTML/CSS| |JS/HTML/CSS| |JS/HTML/CSS| |JS/HTML/CSS|
+-----------+ +-----------+ +-----------+ +-----------+
+-----------+ +-----------+ +-----------+ +-----------+
| | | | | | | |
| | | | | | | |
| Browser | ------------------------- | Browser | | Browser |--------------------------------| Browser |
| | Media path | | | | Media Path | |
| | | | | | | |
+-----------+ +-----------+ +-----------+ +-----------+
Figure 2: Browser RTC Trapezoid Figure 2: Browser RTC Trapezoid
On this drawing, the critical part to note is that the media path 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 ("low path") goes directly between the browsers, so it has to be
conformant to the specifications of the WebRTC protocol suite; the conformant to the specifications of the WebRTC protocol suite; the
signaling path ("high path") goes via servers that can modify, signaling path ("high path") goes via servers that can modify,
translate or manipulate the signals as needed. translate, or manipulate the signals as needed.
If the two Web servers are operated by different entities, the inter- If the two web servers are operated by different entities, the inter-
server signaling mechanism needs to be agreed upon, either by server signaling mechanism needs to be agreed upon, by either
standardization or by other means of agreement. Existing protocols standardization or other means of agreement. Existing protocols
(e.g. SIP [RFC3261] or XMPP [RFC6120]) could be used between (e.g., SIP [RFC3261] or the Extensible Messaging and Presence
servers, while either a standards-based or proprietary protocol could Protocol (XMPP) [RFC6120]) could be used between servers, while
be used between the browser and the web server. 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 For example, if both operators' servers implement SIP, SIP could be
used for communication between servers, along with either a used for communication between servers, along with either a
standardized signaling mechanism (e.g. SIP over WebSockets) or a standardized signaling mechanism (e.g., SIP over WebSockets) or a
proprietary signaling mechanism used between the application running proprietary signaling mechanism used between the application running
in the browser and the web server. Similarly, if both operators' in the browser and the web server. Similarly, if both operators'
servers implement Extensible Messaging and Presence Protocol (XMPP), servers implement XMPP, XMPP could be used for communication between
XMPP could be used for communication between XMPP servers, with XMPP servers, with either a standardized signaling mechanism (e.g.,
either a standardized signaling mechanism (e.g. XMPP over WebSockets XMPP over WebSockets or Bidirectional-streams Over Synchronous HTTP
or BOSH [XEP-0124] or a proprietary signaling mechanism used between (BOSH) [XEP-0124]) or a proprietary signaling mechanism used between
the application running in the browser and the web server. the application running in the browser and the web server.
The choice of protocols for client-server and inter-server The choice of protocols for client-server and inter-server signaling,
signalling, and definition of the translation between them, is and the definition of the translation between them, are outside the
outside the scope of the WebRTC protocol suite described in the scope of the WebRTC protocol suite described in this document.
document.
The functionality groups that are needed in the browser can be The functionality groups that are needed in the browser can be
specified, more or less from the bottom up, as: specified, more or less from the bottom up, as:
o Data transport: such as TCP, UDP and the means to securely set up Data transport: For example, TCP and UDP, and the means to securely
connections between entities, as well as the functions for set up connections between entities, as well as the functions for
deciding when to send data: congestion management, bandwidth deciding when to send data: congestion management, bandwidth
estimation and so on. estimation, and so on.
o Data framing: RTP, SCTP, DTLS, and other data formats that serve Data framing: RTP, the Stream Control Transmission Protocol (SCTP),
as containers, and their functions for data confidentiality and DTLS, and other data formats that serve as containers, and their
integrity. functions for data confidentiality and integrity.
o Data formats: Codec specifications, format specifications and Data formats: Codec specifications, format specifications, and
functionality specifications for the data passed between systems. functionality specifications for the data passed between systems.
Audio and video codecs, as well as formats for data and document Audio and video codecs, as well as formats for data and document
sharing, belong in this category. In order to make use of data sharing, belong in this category. In order to make use of data
formats, a way to describe them, a session description, is needed. formats, a way to describe them (e.g., a session description) is
needed.
o Connection management: Setting up connections, agreeing on data Connection management: For example, setting up connections, agreeing
formats, changing data formats during the duration of a call; SDP, on data formats, changing data formats during the duration of a
SIP, and Jingle/XMPP belong in this category. call. SDP, SIP, and Jingle/XMPP belong in this category.
o Presentation and control: What needs to happen in order to ensure Presentation and control: What needs to happen in order to ensure
that interactions behave in a non-surprising manner. This can that interactions behave in an unsurprising manner. This can
include floor control, screen layout, voice activated image include floor control, screen layout, voice-activated image
switching and other such functions - where part of the system switching, and other such functions, where part of the system
require the cooperation between parties. XCON and Cisco/ requires cooperation between parties. Centralized Conferencing
Tandberg's TIP were some attempts at specifying this kind of (XCON) [RFC6501] and Cisco/Tandberg's Telepresence
functionality; many applications have been built without Interoperability Protocol (TIP) were some attempts at specifying
standardized interfaces to these functions. this kind of functionality; many applications have been built
without standardized interfaces to these functions.
o Local system support functions: These are things that need not be Local system support functions: Functions that need not be specified
specified uniformly, because each participant may choose to do uniformly, because each participant may implement these functions
these in a way of the participant's choosing, without affecting as they choose, without affecting the bits on the wire in a way
the bits on the wire in a way that others have to be cognizant of. that others have to be cognizant of. Examples in this category
Examples in this category include echo cancellation (some forms of include echo cancellation (some forms of it), local authentication
it), local authentication and authorization mechanisms, OS access and authorization mechanisms, OS access control, and the ability
control and the ability to do local recording of conversations. to do local recording of conversations.
Within each functionality group, it is important to preserve both Within each functionality group, it is important to preserve both
freedom to innovate and the ability for global communication. freedom to innovate and the ability for global communication.
Freedom to innovate is helped by doing the specification in terms of Freedom to innovate is helped by doing the specification in terms of
interfaces, not implementation; any implementation able to interfaces, not implementation; any implementation able to
communicate according to the interfaces is a valid implementation. communicate according to the interfaces is a valid implementation.
Ability to communicate globally is helped both by having core The ability to communicate globally is helped by both (1) having core
specifications be unencumbered by IPR issues and by having the specifications be unencumbered by IPR issues and (2) having the
formats and protocols be fully enough specified to allow for formats and protocols be fully enough specified to allow for
independent implementation. independent implementation.
One can think of the three first groups as forming a "media transport One can think of the first three groups as forming a "media transport
infrastructure", and of the three last groups as forming a "media infrastructure" and of the last three groups as forming a "media
service". In many contexts, it makes sense to use a common service". In many contexts, it makes sense to use a common
specification for the media transport infrastructure, which can be specification for the media transport infrastructure, which can be
embedded in browsers and accessed using standard interfaces, and "let embedded in browsers and accessed using standard interfaces, and "let
a thousand flowers bloom" in the "media service" layer; to achieve a thousand flowers bloom" in the "media service" layer; to achieve
interoperable services, however, at least the first five of the six interoperable services, however, at least the first five of the six
groups need to be specified. groups need to be specified.
4. Data transport 4. Data Transport
Data transport refers to the sending and receiving of data over the Data transport refers to the sending and receiving of data over the
network interfaces, the choice of network-layer addresses at each end network interfaces, the choice of network-layer addresses at each end
of the communication, and the interaction with any intermediate of the communication, and the interaction with any intermediate
entities that handle the data, but do not modify it (such as TURN entities that handle the data but do not modify it (such as Traversal
relays). Using Relays around NAT (TURN) relays).
It includes necessary functions for congestion control, It includes necessary functions for congestion control,
retransmission, and in-order delivery. retransmission, and in-order delivery.
WebRTC endpoints MUST implement the transport protocols described in WebRTC endpoints MUST implement the transport protocols described in
[I-D.ietf-rtcweb-transports]. [RFC8835].
5. Data framing and securing 5. Data Framing and Securing
The format for media transport is RTP [RFC3550]. Implementation of The format for media transport is RTP [RFC3550]. Implementation of
SRTP [RFC3711] is REQUIRED for all implementations. the Secure Real-time Transport Protocol (SRTP) [RFC3711] is REQUIRED
for all implementations.
The detailed considerations for usage of functions from RTP and SRTP The detailed considerations for usage of functions from RTP and SRTP
are given in [I-D.ietf-rtcweb-rtp-usage]. The security are given in [RFC8834]. The security considerations for the WebRTC
considerations for the WebRTC use case are in use case are provided in [RFC8826], and the resulting security
[I-D.ietf-rtcweb-security], and the resulting security functions are functions are described in [RFC8827].
described in [I-D.ietf-rtcweb-security-arch].
Considerations for the transfer of data that is not in RTP format is Considerations for the transfer of data that is not in RTP format are
described in [I-D.ietf-rtcweb-data-channel], and a supporting described in [RFC8831], and a supporting protocol for establishing
protocol for establishing individual data channels is described in individual data channels is described in [RFC8832]. WebRTC endpoints
[I-D.ietf-rtcweb-data-protocol]. WebRTC endpoints MUST implement MUST implement these two specifications.
these two specifications.
WebRTC endpoints MUST implement [I-D.ietf-rtcweb-rtp-usage], WebRTC endpoints MUST implement [RFC8834], [RFC8826], [RFC8827], and
[I-D.ietf-rtcweb-security], [I-D.ietf-rtcweb-security-arch], and the the requirements they include.
requirements they include.
6. Data formats 6. Data Formats
The intent of this specification is to allow each communications The intent of this specification is to allow each communications
event to use the data formats that are best suited for that event to use the data formats that are best suited for that
particular instance, where a format is supported by both sides of the particular instance, where a format is supported by both sides of the
connection. However, a minimum standard is greatly helpful in order connection. However, a minimum standard is greatly helpful in order
to ensure that communication can be achieved. This document to ensure that communication can be achieved. This document
specifies a minimum baseline that will be supported by all specifies a minimum baseline that will be supported by all
implementations of this specification, and leaves further codecs to implementations of this specification and leaves further codecs to be
be included at the will of the implementor. included at the will of the implementer.
WebRTC endpoints that support audio and/or video MUST implement the WebRTC endpoints that support audio and/or video MUST implement the
codecs and profiles required in [RFC7874] and [RFC7742]. codecs and profiles required in [RFC7874] and [RFC7742].
7. Connection management 7. Connection Management
The methods, mechanisms and requirements for setting up, negotiating The methods, mechanisms, and requirements for setting up,
and tearing down connections is a large subject, and one where it is negotiating, and tearing down connections comprise a large subject,
desirable to have both interoperability and freedom to innovate. and one where it is desirable to have both interoperability and
freedom to innovate.
The following principles apply: The following principles apply:
1. The WebRTC media negotiations will be capable of representing the 1. The WebRTC media negotiations will be capable of representing the
same SDP offer/answer semantics [RFC3264] that are used in SIP, same SDP offer/answer semantics [RFC3264] that are used in SIP,
in such a way that it is possible to build a signaling gateway in such a way that it is possible to build a signaling gateway
between SIP and the WebRTC media negotiation. between SIP and the WebRTC media negotiation.
2. It will be possible to gateway between legacy SIP devices that 2. It will be possible to gateway between legacy SIP devices that
support ICE and appropriate RTP / SDP mechanisms, codecs and support ICE and appropriate RTP/SDP mechanisms, codecs, and
security mechanisms without using a media gateway. A signaling security mechanisms without using a media gateway. A signaling
gateway to convert between the signaling on the web side to the gateway to convert between the signaling on the web side and the
SIP signaling may be needed. SIP signaling may be needed.
3. When an SDP for a new codec is specified, no other 3. When an SDP for a new codec is specified, no other
standardization should be required for it to be possible to use standardization should be required for it to be possible to use
that in the web browsers. Adding new codecs which might have new that codec in the web browsers. Adding new codecs that might
SDP parameters should not change the APIs between the browser and have new SDP parameters should not change the APIs between the
Javascript application. As soon as the browsers support the new browser and the JavaScript application. As soon as the browsers
codecs, old applications written before the codecs were specified support the new codecs, old applications written before the
should automatically be able to use the new codecs where codecs were specified should automatically be able to use the new
appropriate with no changes to the JS applications. codecs where appropriate, with no changes to the JavaScript
applications.
The particular choices made for WebRTC, and their implications for The particular choices made for WebRTC, and their implications for
the API offered by a browser implementing WebRTC, are described in the API offered by a browser implementing WebRTC, are described in
[I-D.ietf-rtcweb-jsep]. [RFC8829].
WebRTC browsers MUST implement [I-D.ietf-rtcweb-jsep]. WebRTC browsers MUST implement [RFC8829].
WebRTC endpoints MUST implement the functions described in that WebRTC endpoints MUST implement those functions described in
document that relate to the network layer (e.g. Bundle [RFC8829] that relate to the network layer (e.g., BUNDLE [RFC8843],
[I-D.ietf-mmusic-sdp-bundle-negotiation], RTCP-mux [RFC5761] and "rtcp-mux" [RFC5761], and Trickle ICE [RFC8838]), but these endpoints
Trickle ICE [I-D.ietf-ice-trickle]), but do not need to support the do not need to support the API functionality described in [RFC8829].
API functionality described there.
8. Presentation and control 8. Presentation and Control
The most important part of control is the user's control over the The most important part of control is the users' control over the
browser's interaction with input/output devices and communications browser's interaction with input/output devices and communications
channels. It is important that the user have some way of figuring channels. It is important that the users have some way of figuring
out where his audio, video or texting is being sent, for what out where their audio, video, or texting is being sent; for what
purported reason, and what guarantees are made by the parties that purported reason; and what guarantees are made by the parties that
form part of this control channel. This is largely a local function form part of this control channel. This is largely a local function
between the browser, the underlying operating system and the user between the browser, the underlying operating system, and the user
interface; this is specified in the peer connection API interface; this is specified in the peer connection API
[W3C.WD-webrtc-20120209], and the media capture API [W3C.WD-webrtc] and the media capture API
[W3C.WD-mediacapture-streams-20120628]. [W3C.WD-mediacapture-streams].
WebRTC browsers MUST implement these two specifications. WebRTC browsers MUST implement these two specifications.
9. Local system support functions 9. Local System Support Functions
These are characterized by the fact that the quality of these These functions are characterized by the fact that the quality of an
functions strongly influence the user experience, but the exact implementation strongly influences the user experience, but the exact
algorithm does not need coordination. In some cases (for instance algorithm does not need coordination. In some cases (for instance,
echo cancellation, as described below), the overall system definition echo cancellation, as described below), the overall system definition
may need to specify that the overall system needs to have some may need to specify that the overall system needs to have some
characteristics for which these facilities are useful, without characteristics for which these facilities are useful, without
requiring them to be implemented a certain way. requiring them to be implemented a certain way.
Local functions include echo cancellation, volume control, camera Local functions include echo cancellation; volume control; camera
management including focus, zoom, pan/tilt controls (if available), management, including focus, zoom, and pan/tilt controls (if
and more. available); and more.
One would want to see certain parts of the system conform to certain One would want to see certain parts of the system conform to certain
properties, for instance: properties; for instance:
o Echo cancellation should be good enough to achieve the suppression * Echo cancellation should be good enough to achieve the suppression
of acoustical feedback loops below a perceptually noticeable of acoustical feedback loops below a perceptually noticeable
level. level.
o Privacy concerns MUST be satisfied; for instance, if remote * Privacy concerns MUST be satisfied; for instance, if remote
control of camera is offered, the APIs should be available to let control of a camera is offered, the APIs should be available to
the local participant figure out who's controlling the camera, and let the local participant figure out who's controlling the camera
possibly decide to revoke the permission for camera usage. and possibly decide to revoke the permission for camera usage.
o Automatic gain control, if present, should normalize a speaking * Automatic Gain Control (AGC), if present, should normalize a
voice into a reasonable dB range. speaking voice into a reasonable dB range.
The requirements on WebRTC systems with regard to audio processing The requirements on WebRTC systems with regard to audio processing
are found in [RFC7874] and includes more guidance about echo are found in [RFC7874], and that document includes more guidance
cancellation and AGC; the proposed API for control of local devices about echo cancellation and AGC; the APIs for control of local
are found in [W3C.WD-mediacapture-streams-20120628]. devices are found in [W3C.WD-mediacapture-streams].
WebRTC endpoints MUST implement the processing functions in WebRTC endpoints MUST implement the processing functions in
[RFC7874]. (Together with the requirement in Section 6, this means [RFC7874]. (Together with the requirement in Section 6, this means
that WebRTC endpoints MUST implement the whole document.) that WebRTC endpoints MUST implement the whole document.)
10. IANA Considerations 10. IANA Considerations
This document makes no request of IANA. This document has no IANA actions.
Note to RFC Editor: this section may be removed on publication as an
RFC.
11. Security Considerations 11. Security Considerations
Security of the web-enabled real time communications comes in several Security of the web-enabled real-time communications comes in several
pieces: pieces:
o Security of the components: The browsers, and other servers Security of the components: The browsers, and other servers
involved. The most target-rich environment here is probably the involved. The most target-rich environment here is probably the
browser; the aim here should be that the introduction of these browser; the aim here should be that the introduction of these
components introduces no additional vulnerability. components introduces no additional vulnerability.
o Security of the communication channels: It should be easy for a Security of the communication channels: It should be easy for
participant to reassure himself of the security of his participants to reassure themselves of the security of their
communication - by verifying the crypto parameters of the links he communication -- by verifying the crypto parameters of the links
himself participates in, and to get reassurances from the other that they participate in, and to get reassurances from the other
parties to the communication that they promise that appropriate parties to the communication that those parties promise that
measures are taken. appropriate measures are taken.
o Security of the partners' identity: verifying that the Security of the partners' identities: Verifying that the
participants are who they say they are (when positive participants are who they say they are (when positive
identification is appropriate), or that their identity cannot be identification is appropriate) or that their identities cannot be
uncovered (when anonymity is a goal of the application). uncovered (when anonymity is a goal of the application).
The security analysis, and the requirements derived from that The security analysis, and the requirements derived from that
analysis, is contained in [I-D.ietf-rtcweb-security]. analysis, are contained in [RFC8826].
It is also important to read the security sections of It is also important to read the security sections of
[W3C.WD-mediacapture-streams-20120628] and [W3C.WD-webrtc-20120209]. [W3C.WD-mediacapture-streams] and [W3C.WD-webrtc].
12. Acknowledgements
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.
Thanks to Cary Bran, Cullen Jennings, Colin Perkins, Magnus
Westerlund and Joerg Ott, who offered technical contributions on
various versions of the draft.
Thanks to Jonathan Rosenberg, Matthew Kaufman and others at Skype for
the ASCII drawings in section 1.
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.
13. References
13.1. Normative References
[I-D.ietf-rtcweb-data-channel]
Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data
Channels", draft-ietf-rtcweb-data-channel-13 (work in
progress), January 2015.
[I-D.ietf-rtcweb-data-protocol]
Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel
Establishment Protocol", draft-ietf-rtcweb-data-
protocol-09 (work in progress), January 2015.
[I-D.ietf-rtcweb-jsep]
Uberti, J., Jennings, C., and E. Rescorla, "JavaScript
Session Establishment Protocol", draft-ietf-rtcweb-jsep-24
(work in progress), October 2017.
[I-D.ietf-rtcweb-rtp-usage]
Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time
Communication (WebRTC): Media Transport and Use of RTP",
draft-ietf-rtcweb-rtp-usage-26 (work in progress), March
2016.
[I-D.ietf-rtcweb-security]
Rescorla, E., "Security Considerations for WebRTC", draft-
ietf-rtcweb-security-09 (work in progress), October 2017.
[I-D.ietf-rtcweb-security-arch] 12. References
Rescorla, E., "WebRTC Security Architecture", draft-ietf-
rtcweb-security-arch-13 (work in progress), October 2017.
[I-D.ietf-rtcweb-transports] 12.1. Normative References
Alvestrand, H., "Transports for WebRTC", draft-ietf-
rtcweb-transports-17 (work in progress), October 2016.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997, DOI 10.17487/RFC2119, March 1997,
<https://www.rfc-editor.org/info/rfc2119>. <https://www.rfc-editor.org/info/rfc2119>.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model [RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264, with Session Description Protocol (SDP)", RFC 3264,
DOI 10.17487/RFC3264, June 2002, DOI 10.17487/RFC3264, June 2002,
<https://www.rfc-editor.org/info/rfc3264>. <https://www.rfc-editor.org/info/rfc3264>.
skipping to change at page 18, line 5 skipping to change at line 730
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
July 2003, <https://www.rfc-editor.org/info/rfc3550>. July 2003, <https://www.rfc-editor.org/info/rfc3550>.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)", Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, DOI 10.17487/RFC3711, March 2004, RFC 3711, DOI 10.17487/RFC3711, March 2004,
<https://www.rfc-editor.org/info/rfc3711>. <https://www.rfc-editor.org/info/rfc3711>.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment [RFC7742] Roach, A.B., "WebRTC Video Processing and Codec
(ICE): A Protocol for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols", RFC 5245,
DOI 10.17487/RFC5245, April 2010,
<https://www.rfc-editor.org/info/rfc5245>.
[RFC7742] Roach, A., "WebRTC Video Processing and Codec
Requirements", RFC 7742, DOI 10.17487/RFC7742, March 2016, Requirements", RFC 7742, DOI 10.17487/RFC7742, March 2016,
<https://www.rfc-editor.org/info/rfc7742>. <https://www.rfc-editor.org/info/rfc7742>.
[RFC7874] Valin, JM. and C. Bran, "WebRTC Audio Codec and Processing [RFC7874] Valin, JM. and C. Bran, "WebRTC Audio Codec and Processing
Requirements", RFC 7874, DOI 10.17487/RFC7874, May 2016, Requirements", RFC 7874, DOI 10.17487/RFC7874, May 2016,
<https://www.rfc-editor.org/info/rfc7874>. <https://www.rfc-editor.org/info/rfc7874>.
[W3C.WD-mediacapture-streams-20120628] [RFC8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
Burnett, D. and A. Narayanan, "Media Capture and Streams", 2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
World Wide Web Consortium WD WD-mediacapture-streams- May 2017, <https://www.rfc-editor.org/info/rfc8174>.
20120628, June 2012, <http://www.w3.org/TR/2012/
WD-mediacapture-streams-20120628>.
[W3C.WD-webrtc-20120209] [RFC8445] Keranen, A., Holmberg, C., and J. Rosenberg, "Interactive
Bergkvist, A., Burnett, D., Jennings, C., and A. Connectivity Establishment (ICE): A Protocol for Network
Narayanan, "WebRTC 1.0: Real-time Communication Between Address Translator (NAT) Traversal", RFC 8445,
Browsers", World Wide Web Consortium WD WD-webrtc- DOI 10.17487/RFC8445, July 2018,
20120209, February 2012, <https://www.rfc-editor.org/info/rfc8445>.
<http://www.w3.org/TR/2012/WD-webrtc-20120209>.
13.2. Informative References [RFC8826] Rescorla, E., "Security Considerations for WebRTC",
RFC 8826, DOI 10.17487/RFC8826, January 2021,
<https://www.rfc-editor.org/info/rfc8826>.
[I-D.ietf-ice-trickle] [RFC8827] Rescorla, E., "WebRTC Security Architecture", RFC 8827,
Ivov, E., Rescorla, E., Uberti, J., and P. Saint-Andre, DOI 10.17487/RFC8827, January 2021,
"Trickle ICE: Incremental Provisioning of Candidates for <https://www.rfc-editor.org/info/rfc8827>.
the Interactive Connectivity Establishment (ICE)
Protocol", draft-ietf-ice-trickle-14 (work in progress),
September 2017.
[I-D.ietf-mmusic-sdp-bundle-negotiation] [RFC8829] Uberti, J., Jennings, C., and E. Rescorla, Ed.,
Holmberg, C., Alvestrand, H., and C. Jennings, "JavaScript Session Establishment Protocol (JSEP)",
"Negotiating Media Multiplexing Using the Session RFC 8829, DOI 10.17487/RFC8829, January 2021,
Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle- <https://www.rfc-editor.org/info/rfc8829>.
negotiation-39 (work in progress), August 2017.
[I-D.ietf-rtcweb-gateways] [RFC8831] Jesup, R., Loreto, S., and M. Tüxen, "WebRTC Data
Alvestrand, H. and U. Rauschenbach, "WebRTC Gateways", Channels", RFC 8831, DOI 10.17487/RFC8831, January 2021,
draft-ietf-rtcweb-gateways-02 (work in progress), January <https://www.rfc-editor.org/info/rfc8831>.
2016.
[I-D.ietf-tsvwg-rtcweb-qos] [RFC8832] Jesup, R., Loreto, S., and M. Tüxen, "WebRTC Data Channel
Jones, P., Dhesikan, S., Jennings, C., and D. Druta, "DSCP Establishment Protocol", RFC 8832, DOI 10.17487/RFC8832,
Packet Markings for WebRTC QoS", draft-ietf-tsvwg-rtcweb- January 2021, <https://www.rfc-editor.org/info/rfc8832>.
qos-18 (work in progress), August 2016.
[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,
<https://www.rfc-editor.org/info/rfc8835>.
[W3C.WD-mediacapture-streams]
Jennings, C., Aboba, B., Bruaroey, J-I., and H. Boström,
"Media Capture and Streams", W3C Candidate Recommendation,
<https://www.w3.org/TR/mediacapture-streams/>.
[W3C.WD-webrtc]
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,
<https://html.spec.whatwg.org/>.
[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, [RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
A., Peterson, J., Sparks, R., Handley, M., and E. A., Peterson, J., Sparks, R., Handley, M., and E.
Schooler, "SIP: Session Initiation Protocol", RFC 3261, Schooler, "SIP: Session Initiation Protocol", RFC 3261,
DOI 10.17487/RFC3261, June 2002, DOI 10.17487/RFC3261, June 2002,
<https://www.rfc-editor.org/info/rfc3261>. <https://www.rfc-editor.org/info/rfc3261>.
[RFC3361] Schulzrinne, H., "Dynamic Host Configuration Protocol [RFC3361] Schulzrinne, H., "Dynamic Host Configuration Protocol
(DHCP-for-IPv4) Option for Session Initiation Protocol (DHCP-for-IPv4) Option for Session Initiation Protocol
(SIP) Servers", RFC 3361, DOI 10.17487/RFC3361, August (SIP) Servers", RFC 3361, DOI 10.17487/RFC3361, August
2002, <https://www.rfc-editor.org/info/rfc3361>. 2002, <https://www.rfc-editor.org/info/rfc3361>.
[RFC3935] Alvestrand, H., "A Mission Statement for the IETF", [RFC3935] Alvestrand, H., "A Mission Statement for the IETF",
BCP 95, RFC 3935, DOI 10.17487/RFC3935, October 2004, BCP 95, RFC 3935, DOI 10.17487/RFC3935, October 2004,
<https://www.rfc-editor.org/info/rfc3935>. <https://www.rfc-editor.org/info/rfc3935>.
[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,
<https://www.rfc-editor.org/info/rfc5245>.
[RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and [RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
Control Packets on a Single Port", RFC 5761, Control Packets on a Single Port", RFC 5761,
DOI 10.17487/RFC5761, April 2010, DOI 10.17487/RFC5761, April 2010,
<https://www.rfc-editor.org/info/rfc5761>. <https://www.rfc-editor.org/info/rfc5761>.
[RFC6120] Saint-Andre, P., "Extensible Messaging and Presence [RFC6120] Saint-Andre, P., "Extensible Messaging and Presence
Protocol (XMPP): Core", RFC 6120, DOI 10.17487/RFC6120, Protocol (XMPP): Core", RFC 6120, DOI 10.17487/RFC6120,
March 2011, <https://www.rfc-editor.org/info/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- [RFC7478] Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
Time Communication Use Cases and Requirements", RFC 7478, Time Communication Use Cases and Requirements", RFC 7478,
DOI 10.17487/RFC7478, March 2015, DOI 10.17487/RFC7478, March 2015,
<https://www.rfc-editor.org/info/rfc7478>. <https://www.rfc-editor.org/info/rfc7478>.
[RFC8155] Patil, P., Reddy, T., and D. Wing, "Traversal Using Relays [RFC8155] Patil, P., Reddy, T., and D. Wing, "Traversal Using Relays
around NAT (TURN) Server Auto Discovery", RFC 8155, around NAT (TURN) Server Auto Discovery", RFC 8155,
DOI 10.17487/RFC8155, April 2017, DOI 10.17487/RFC8155, April 2017,
<https://www.rfc-editor.org/info/rfc8155>. <https://www.rfc-editor.org/info/rfc8155>.
[W3C.WD-html5-20110525] [RFC8837] Jones, P., Dhesikan, S., Jennings, C., and D. Druta,
Hickson, I., "HTML5", World Wide Web Consortium LastCall "Differentiated Services Code Point (DSCP) Packet Markings
WD-html5-20110525, May 2011, for WebRTC QoS", RFC 8837, DOI 10.17487/RFC8837, January
<http://www.w3.org/TR/2011/WD-html5-20110525>. 2021, <https://www.rfc-editor.org/info/rfc8837>.
[XEP-0124]
Paterson, I., Smith, D., Saint-Andre, P., Moffitt, J.,
Stout, L., and W. Tilanus, "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, June 2007.
Appendix A. Change log
This section may be deleted by the RFC Editor when preparing for
publication.
A.1. Changes from draft-alvestrand-dispatch-rtcweb-datagram-00 to -01
Added section "On interoperability and innovation"
Added data confidentiality and integrity to the "data framing" layer
Added congestion management requirements in the "data transport"
layer section
Changed need for non-media data from "question: do we need this?" to
"Open issue: How do we do this?"
Strengthened disclaimer that listed codecs are placeholders, not
decisions.
More details on why the "local system support functions" section is
there.
A.2. Changes from draft-alvestrand-dispatch-01 to draft-alvestrand-
rtcweb-overview-00
Added section on "Relationship between API and protocol"
Added terminology section
Mentioned congestion management as part of the "data transport" layer
in the layer list
A.3. Changes from draft-alvestrand-rtcweb-00 to -01
Removed most technical content, and replaced with pointers to drafts
as requested and identified by the RTCWEB WG chairs.
Added content to acknowledgments section.
Added change log.
Spell-checked document.
A.4. Changes from draft-alvestrand-rtcweb-overview-01 to draft-ietf-
rtcweb-overview-00
Changed draft name and document date.
Removed unused references
A.5. Changes from -00 to -01 of draft-ietf-rtcweb-overview
Added architecture figures to section 2.
Changed the description of "echo cancellation" under "local system
support functions".
Added a few more definitions.
A.6. Changes from -01 to -02 of draft-ietf-rtcweb-overview
Added pointers to use cases, security and rtp-usage drafts (now WG
drafts).
Changed description of SRTP from mandatory-to-use to mandatory-to-
implement.
Added the "3 principles of negotiation" to the connection management
section.
Added an explicit statement that ICE is required for both NAT and
consent-to-receive.
A.7. Changes from -02 to -03 of draft-ietf-rtcweb-overview
Added references to a number of new drafts.
Expanded the description text under the "trapezoid" drawing with some
more text discussed on the list.
Changed the "Connection management" sentence from "will be done using
SDP offer/answer" to "will be capable of representing SDP offer/
answer" - this seems more consistent with JSEP.
Added "security mechanisms" to the things a non-gatewayed SIP devices
must support in order to not need a media gateway.
Added a definition for "browser".
A.8. Changes from -03 to -04 of draft-ietf-rtcweb-overview
Made introduction more normative.
Several wording changes in response to review comments from EKR
Added an appendix to hold references and notes that are not yet in a
separate document.
A.9. Changes from -04 to -05 of draft-ietf-rtcweb-overview
Minor grammatical fixes. This is mainly a "keepalive" refresh.
A.10. Changes from -05 to -06
Clarifications in response to Last Call review comments. Inserted
reference to draft-ietf-rtcweb-audio.
A.11. Changes from -06 to -07
Added a reference to the "unified plan" draft, and updated some
references.
Otherwise, it's a "keepalive" draft.
A.12. Changes from -07 to -08
Removed the appendix that detailed transports, and replaced it with a
reference to draft-ietf-rtcweb-transports. Removed now-unused
references.
A.13. Changes from -08 to -09
Added text to the Abstract indicating that the intended status is an
Applicability Statement.
A.14. Changes from -09 to -10
Defined "WebRTC Browser" and "WebRTC device" as things that do, or
don't, conform to the API.
Updated reference to data-protocol draft
Updated data formats to reference -rtcweb-audio- and not the expired
-cbran draft.
Deleted references to -unified-plan
Deleted reference to -generic-idp (draft expired)
Added notes on which referenced documents WebRTC browsers or devices
MUST conform to.
Added pointer to the security section of the API drafts.
A.15. Changes from -10 to -11
Added "WebRTC Gateway" as a third class of device, and referenced the
doc describing them.
Made a number of text clarifications in response to document reviews.
A.16. Changes from -11 to -12
Refined entity definitions to define "WebRTC endpoint" and "WebRTC-
compatible endpoint".
Changed remaining usage of the term "RTCWEB" to "WebRTC", including
in the page header.
A.17. Changes from -12 to -13
Changed "WebRTC device" to be "WebRTC non-browser", per decision at
IETF 91. This led to the need for "WebRTC endpoint" as the common
label for both, and the usage of that term in the rest of the
document.
Added words about WebRTC APIs in languages other than Javascript.
Referenced draft-ietf-rtcweb-video for video codecs to support.
A.18. Changes from -13 to -14
None. This is a "keepalive" update.
A.19. Changes from -14 to -15
Changed "gateways" reference to point to the WG document.
A.20. Changes from -15 to -16
None. This is a "keepalive" publication.
A.21. Changes from -16 to -17 [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,
<https://www.rfc-editor.org/info/rfc8838>.
Addressed review comments by Olle E. Johansson and Magnus Westerlund [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,
<https://www.rfc-editor.org/info/rfc8843>.
A.22. Changes from -17 to -18 [WebRTC-Gateways]
Alvestrand, H. and U. Rauschenbach, "WebRTC Gateways",
Work in Progress, Internet-Draft, draft-ietf-rtcweb-
gateways-02, 21 January 2016,
<https://tools.ietf.org/html/draft-ietf-rtcweb-gateways-
02>.
Addressed review comments from Sean Turner and Alissa Cooper [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,
<https://xmpp.org/extensions/xep-0124.html>.
A.23. Changes from -18 to -19 [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>.
A number of grammatical issues were fixed. Acknowledgements
Added note on operational impact of WebRTC. 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
important.
Unified all definitions into the definitions list. Thanks to Cary Bran, Cullen Jennings, Colin Perkins, Magnus
Westerlund, and Jörg Ott, who offered technical contributions to
various draft versions of this document.
Added a reference for BOSH. Thanks to Jonathan Rosenberg, Matthew Kaufman, and others at Skype
for the ASCII drawings in Section 3.
Changed ICE reference from 5245bis to RFC 5245. 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 Author's Address
Harald T. Alvestrand Harald T. Alvestrand
Google Google
Kungsbron 2 Kungsbron 2
Stockholm 11122 SE-11122 Stockholm
Sweden Sweden
Email: harald@alvestrand.no Email: harald@alvestrand.no
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