draft-ietf-rtcweb-overview-18.txt   draft-ietf-rtcweb-overview-19.txt 
Network Working Group H. Alvestrand Network Working Group H. Alvestrand
Internet-Draft Google Internet-Draft Google
Intended status: Standards Track March 3, 2017 Intended status: Standards Track November 12, 2017
Expires: September 4, 2017 Expires: May 16, 2018
Overview: Real Time Protocols for Browser-based Applications Overview: Real Time Protocols for Browser-based Applications
draft-ietf-rtcweb-overview-18 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 all the parts that are needed to achieve this goal are findable, and
that the parts that belong in the Internet protocol suite are fully that the parts that belong in the Internet protocol suite are fully
skipping to change at page 1, line 36 skipping to change at page 1, line 36
This document is a work item of the RTCWEB working group. This document is a work item of the RTCWEB working group.
Status of This Memo Status of This Memo
This Internet-Draft is submitted in full conformance with the This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79. provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet- working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/. Drafts is at https://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress." material or to cite them other than as "work in progress."
This Internet-Draft will expire on September 4, 2017. This Internet-Draft will expire on May 16, 2018.
Copyright Notice Copyright Notice
Copyright (c) 2017 IETF Trust and the persons identified as the Copyright (c) 2017 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 . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Principles and Terminology . . . . . . . . . . . . . . . . . 4 2. Principles and Terminology . . . . . . . . . . . . . . . . . 4
2.1. Goals of this document . . . . . . . . . . . . . . . . . 4 2.1. Goals of this document . . . . . . . . . . . . . . . . . 4
2.2. Relationship between API and protocol . . . . . . . . . . 4 2.2. Relationship between API and protocol . . . . . . . . . . 5
2.3. On interoperability and innovation . . . . . . . . . . . 6 2.3. On interoperability and innovation . . . . . . . . . . . 7
2.4. Terminology . . . . . . . . . . . . . . . . . . . . . . . 7 2.4. Terminology . . . . . . . . . . . . . . . . . . . . . . . 8
3. Architecture and Functionality groups . . . . . . . . . . . . 8 3. Architecture and Functionality groups . . . . . . . . . . . . 8
4. Data transport . . . . . . . . . . . . . . . . . . . . . . . 12 4. Data transport . . . . . . . . . . . . . . . . . . . . . . . 12
5. Data framing and securing . . . . . . . . . . . . . . . . . . 12 5. Data framing and securing . . . . . . . . . . . . . . . . . . 13
6. Data formats . . . . . . . . . . . . . . . . . . . . . . . . 13 6. Data formats . . . . . . . . . . . . . . . . . . . . . . . . 13
7. Connection management . . . . . . . . . . . . . . . . . . . . 13 7. Connection management . . . . . . . . . . . . . . . . . . . . 13
8. Presentation and control . . . . . . . . . . . . . . . . . . 14 8. Presentation and control . . . . . . . . . . . . . . . . . . 14
9. Local system support functions . . . . . . . . . . . . . . . 14 9. Local system support functions . . . . . . . . . . . . . . . 14
10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 15 10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 15
11. Security Considerations . . . . . . . . . . . . . . . . . . . 15 11. Security Considerations . . . . . . . . . . . . . . . . . . . 15
12. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 16 12. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 16
13. References . . . . . . . . . . . . . . . . . . . . . . . . . 16 13. References . . . . . . . . . . . . . . . . . . . . . . . . . 16
13.1. Normative References . . . . . . . . . . . . . . . . . . 16 13.1. Normative References . . . . . . . . . . . . . . . . . . 16
13.2. Informative References . . . . . . . . . . . . . . . . . 18 13.2. Informative References . . . . . . . . . . . . . . . . . 18
Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 19 Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 20
A.1. Changes from draft-alvestrand-dispatch-rtcweb-datagram-00 A.1. Changes from draft-alvestrand-dispatch-rtcweb-datagram-00
to -01 . . . . . . . . . . . . . . . . . . . . . . . . . 19 to -01 . . . . . . . . . . . . . . . . . . . . . . . . . 20
A.2. Changes from draft-alvestrand-dispatch-01 to draft- A.2. Changes from draft-alvestrand-dispatch-01 to draft-
alvestrand-rtcweb-overview-00 . . . . . . . . . . . . . . 19 alvestrand-rtcweb-overview-00 . . . . . . . . . . . . . . 20
A.3. Changes from draft-alvestrand-rtcweb-00 to -01 . . . . . 20 A.3. Changes from draft-alvestrand-rtcweb-00 to -01 . . . . . 20
A.4. Changes from draft-alvestrand-rtcweb-overview-01 to A.4. Changes from draft-alvestrand-rtcweb-overview-01 to
draft-ietf-rtcweb-overview-00 . . . . . . . . . . . . . . 20 draft-ietf-rtcweb-overview-00 . . . . . . . . . . . . . . 21
A.5. Changes from -00 to -01 of draft-ietf-rtcweb-overview . . 20 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 . . 20 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 . . 20 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 . . 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 . . 21 A.9. Changes from -04 to -05 of draft-ietf-rtcweb-overview . . 22
A.10. Changes from -05 to -06 . . . . . . . . . . . . . . . . . 21 A.10. Changes from -05 to -06 . . . . . . . . . . . . . . . . . 22
A.11. Changes from -06 to -07 . . . . . . . . . . . . . . . . . 21 A.11. Changes from -06 to -07 . . . . . . . . . . . . . . . . . 22
A.12. Changes from -07 to -08 . . . . . . . . . . . . . . . . . 21 A.12. Changes from -07 to -08 . . . . . . . . . . . . . . . . . 22
A.13. Changes from -08 to -09 . . . . . . . . . . . . . . . . . 21 A.13. Changes from -08 to -09 . . . . . . . . . . . . . . . . . 22
A.14. Changes from -09 to -10 . . . . . . . . . . . . . . . . . 22 A.14. Changes from -09 to -10 . . . . . . . . . . . . . . . . . 22
A.15. Changes from -10 to -11 . . . . . . . . . . . . . . . . . 22 A.15. Changes from -10 to -11 . . . . . . . . . . . . . . . . . 23
A.16. Changes from -11 to -12 . . . . . . . . . . . . . . . . . 22 A.16. Changes from -11 to -12 . . . . . . . . . . . . . . . . . 23
A.17. Changes from -12 to -13 . . . . . . . . . . . . . . . . . 22 A.17. Changes from -12 to -13 . . . . . . . . . . . . . . . . . 23
A.18. Changes from -13 to -14 . . . . . . . . . . . . . . . . . 23 A.18. Changes from -13 to -14 . . . . . . . . . . . . . . . . . 23
A.19. Changes from -14 to -15 . . . . . . . . . . . . . . . . . 23 A.19. Changes from -14 to -15 . . . . . . . . . . . . . . . . . 23
A.20. Changes from -15 to -16 . . . . . . . . . . . . . . . . . 23 A.20. Changes from -15 to -16 . . . . . . . . . . . . . . . . . 23
A.21. Changes from -16 to -17 . . . . . . . . . . . . . . . . . 23 A.21. Changes from -16 to -17 . . . . . . . . . . . . . . . . . 24
A.22. Changes from -17 to -18 . . . . . . . . . . . . . . . . . 23 A.22. Changes from -17 to -18 . . . . . . . . . . . . . . . . . 24
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 23 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 this were dependent on special networks,
special hardware and custom-built software, often at very high prices special hardware and custom-built software, often at very high prices
or at low quality, placing great demands on the infrastructure. or at low quality, placing great demands on the infrastructure.
As the available bandwidth has increased, and as processors an other As the available bandwidth has increased, and as processors and other
hardware has become ever faster, the barriers to participation have hardware has 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 set
of communication protocols that all agree should be made available of communication protocols that all agree should be made available
for communication; another is the sheer lack of universal 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 proved hard, however, for Development of The Universal Solution has, however, proved hard.
all the usual reasons.
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 on
it. 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, application developers see much promise in
skipping to change at page 4, line 13 skipping to change at page 4, line 13
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 can be made
accessible and controllable through a Javascript API in a browser, accessible and controllable through a Javascript API in a browser,
and which together form a sufficient set of functions to allow the and which together form a sufficient set of functions to allow the
use of interactive audio and video in applications that communicate use of interactive audio and video in applications that communicate
directly between browsers across the Internet. The resulting directly between browsers across the Internet. The resulting
protocol suite is intended to enable all the applications that are protocol suite is intended to enable all the applications that are
described as required scenarios in the use cases document [RFC7478]. described as required scenarios in the use cases document [RFC7478].
Other efforts, for instance the W3C WEBRTC, Web Applications and Other efforts, for instance the W3C Web Real-Time Communications, Web
Device API working groups, focus on making standardized APIs and Applications Security, and Device and Sensor working groups, focus on
interfaces available, within or alongside the HTML5 effort, for those making standardized APIs and interfaces available, within or
functions; this memo concentrates on specifying the protocols and alongside the HTML5 effort, for those functions. This memo
subprotocols that are needed to specify the interactions that happen concentrates on specifying the protocols and subprotocols that are
across the network. 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 SDP -- either through
endpoint cooperation [RFC3361] or through the transparent insertion
of SIP Application Level Gateways (ALGs) -- will not work with such
signaling. In the case of networks using cooperative endpoints, the
approaches defined in [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 [I-D.ietf-tsvwg-rtcweb-qos] may be appropriate
replacements for such techniques.
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
skipping to change at page 4, line 40 skipping to change at page 5, line 12
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 an 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 o A protocol specification, done in the IETF
o A Javascript API specification, defined in a series of W3C o A Javascript API specification, defined in a series of W3C
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is a browser or another device implementing this specification. is a browser or another device implementing this 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 subject to constraints of the implementation, of
course. course.
For the purpose of this document, we define the following terminology The following terms are used across the documents specifying the
to talk about WebRTC things: WebRTC suite, in the specific meanings given here. Not all terms are
used in this document. Other terms are used in their commonly used
meaning.
o A WebRTC browser (also called a WebRTC User Agent or WebRTC UA) is Agent: Undefined term. See "SDP Agent" and "ICE Agent".
something that conforms to both the protocol specification and the
Application Programming Interface (API): A specification of a set of
calls and events, usually tied to a programming language or an
abstract formal specification such as WebIDL, with its defined
semantics.
Browser: Used synonymously with "Interactive User Agent" as defined
in the HTML specification [W3C.WD-html5-20110525]. See also
"WebRTC User Agent".
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) [RFC5245] protocol. An ICE Agent may also be
an SDP Agent, but there exist ICE Agents that do not use SDP (for
instance those that use Jingle [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, with the total time required for the action/reaction/
observation is on the order of no more than hundreds of
milliseconds.
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 generation of content and display of
content are intended to occur closely together in time (on the
order of no more than hundreds of milliseconds). Real-time media
can be used to support interactive communication.
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. Javascript API cited above.
o A WebRTC non-browser is 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".
o A WebRTC endpoint is either a WebRTC browser or a WebRTC non- WebRTC Endpoint: Either a WebRTC browser or a WebRTC non-browser.
browser. It conforms to the protocol specification. It conforms to the protocol specification.
o A WebRTC-compatible endpoint is an endpoint that is able to WebRTC-compatible Endpoint: An endpoint that is able to successfully
successfully communicate with a WebRTC endpoint, but may fail to communicate with a WebRTC endpoint, but may fail to meet some
meet some requirements of a WebRTC endpoint. This may limit where requirements of a WebRTC endpoint. This may limit where in the
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 guarantees that it offers to others. It is not security guarantees that it offers to others. It is not
constrained by this specification; when it is mentioned at all, it constrained by this specification; when it is mentioned at all, it
is to note the implications on WebRTC-compatible endpoints of the is to note the implications on WebRTC-compatible endpoints of the
requirements placed on WebRTC endpoints. requirements placed on WebRTC endpoints.
o A WebRTC gateway is a WebRTC-compatible endpoint that mediates WebRTC Gateway: A WebRTC-compatible endpoint that mediates media
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
similar way to the way in which a browser can host Javascript similar way 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, similar
security considerations as for Javascript may be needed; however, security considerations as for Javascript may be needed; however,
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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, as long as you conform to a specification, and the
other party is willing to accept communication at the base level of other party is willing to accept communication at the base level of
that specification, you can communicate successfully. that specification, you can communicate successfully.
The alternative - that of having no mandatory to implement - does not The alternative, that is having no mandatory to implement, does not
mean that you cannot communicate, it merely means that in order to be mean that you cannot communicate, it merely means that in order to be
part of the communications partnership, you have to implement the part of the communications partnership, you have to implement the
standard "and then some" - that "and then some" usually being called standard "and then some". The "and then some" is usually called a
a profile of some sort; in the version most antithetical to the profile of some sort; in the version most antithetical to the
Internet ethos, that "and then some" consists of having to use a Internet ethos, that "and then some" consists of having to use a
specific vendor's product only. 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", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119]. document are to be interpreted as described in [RFC2119].
The following terms are used across the documents specifying the
WebRTC suite, in the specific meanings given here. Not all terms are
used in this document. Other terms are used in their commonly used
meaning.
The list is in alphabetical order.
Agent: Undefined term. See "SDP Agent" and "ICE Agent".
API: Application Programming Interface - a specification of a set of
calls and events, usually tied to a programming language or an
abstract formal specification such as WebIDL, with its defined
semantics.
Browser: Used synonymously with "Interactive User Agent" as defined
in the HTML specification [W3C.WD-html5-20110525]. See also
"WebRTC User Agent".
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 Connectivty
Establishment (ICE) [I-D.ietf-ice-rfc5245bis] protocol. An ICE
Agent may also be an SDP Agent, but there exist ICE Agents that do
not use SDP (for instance those that use Jingle [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, with the total time required for the action/reaction/
observation is on the order of no more than hundreds of
milliseconds.
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 generation of content and display of
content are intended to occur closely together in time (on the
order of no more than hundreds of milliseconds). Real-time media
can be used to support interactive communication.
SDP Agent: The protocol implementation involved in the 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.
3. Architecture and Functionality groups 3. Architecture and Functionality groups
The model of real-time support for browser-based applications does For browser-based applications, the model for real-time support does
not assume that the browser will contain all the functions that need not assume that the browser will contain all the functions needed for
to be performed in order to have a function such as a telephone or a an application such as a telephone or a video conference. The vision
video conferencing unit; the vision is that the browser will have the is that the browser will have the functions needed for a Web
functions that are needed for a Web application, working in application, working in conjunction with its backend servers, to
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 |--------->
| | | |
| | | |
+------------------------+ +------------------------+
^ ^
| |
| |
| HTTP/ | 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 HTTP 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 the one depicted below.
In the figure below JS is Javascript.
+-----------+ +-----------+ +-----------+ +-----------+
| Web | | Web | | Web | | Web |
| | Signaling | | | | Signaling | |
| |-------------| | | |-------------| |
| Server | path | Server | | Server | path | Server |
| | | | | | | |
+-----------+ +-----------+ +-----------+ +-----------+
/ \ / \
/ \ Application-defined / \ Application-defined
/ \ over / \ over
/ \ HTTP/WebSockets / \ HTTPS/WebSockets
/ Application-defined over \ / Application-defined over \
/ HTTP/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 On this drawing, the critical part to note is that the media path
("low path") goes directly between the browsers, so it has to be ("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 massage 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, either by
standardization or by other means of agreement. Existing protocols standardization or by other means of agreement. Existing protocols
(for example SIP [RFC3261] or XMPP [RFC6120]) could be used between (e.g. SIP [RFC3261] or XMPP [RFC6120]) could be used between
servers, while either a standards-based or proprietary protocol could servers, while either a standards-based or proprietary protocol could
be used between the browser and the web server. 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 XMPP, XMPP could be used for communication between servers implement Extensible Messaging and Presence Protocol (XMPP),
XMPP servers, with either a standardized signaling mechanism (e.g. XMPP could be used for communication between XMPP servers, with
XMPP over WebSockets or BOSH) or a proprietary signaling mechanism either a standardized signaling mechanism (e.g. XMPP over WebSockets
used between the application running in the browser and the web or BOSH [XEP-0124] or a proprietary signaling mechanism used between
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
signalling, and definition of the translation between them, is signalling, and definition of the translation between them, is
outside the scope of the WebRTC protocol suite described in the outside the scope of the WebRTC protocol suite described in the
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: TCP, UDP and the means to securely set up o Data transport: such as TCP, UDP and the means to securely set up
connections between entities, as well as the functions for 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 and other data formats that serve as o Data framing: RTP, SCTP, DTLS, and other data formats that serve
containers, and their functions for data confidentiality and as containers, and their functions for data confidentiality and
integrity. integrity.
o Data formats: Codec specifications, format specifications and o 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, a session description, is needed.
o Connection management: Setting up connections, agreeing on data o Connection management: Setting up connections, agreeing on data
formats, changing data formats during the duration of a call; SIP formats, changing data formats during the duration of a call; SDP,
and Jingle/XMPP belong in this category. SIP, and Jingle/XMPP belong in this category.
o Presentation and control: What needs to happen in order to ensure o 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 a non-surprising 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/ require the cooperation between parties. XCON and Cisco/
Tandberg's TIP were some attempts at specifying this kind of Tandberg's TIP were some attempts at specifying this kind of
functionality; many applications have been built without functionality; many applications have been built without
standardized interfaces to these functions. standardized interfaces to these functions.
skipping to change at page 12, line 36 skipping to change at page 12, line 43
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 TURN
relays). relays).
It includes necessary functions for congestion control: When not to It includes necessary functions for congestion control,
send data. 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]. [I-D.ietf-rtcweb-transports].
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. 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
skipping to change at page 13, line 38 skipping to change at page 13, line 49
7. Connection management 7. Connection management
The methods, mechanisms and requirements for setting up, negotiating The methods, mechanisms and requirements for setting up, negotiating
and tearing down connections is a large subject, and one where it is and tearing down connections is a large subject, and one where it is
desirable to have both interoperability and freedom to innovate. 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 that are used in SIP [RFC3264], 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 to the
SIP signaling may be needed. SIP signaling may be needed.
3. When a new codec is specified, and the SDP for the new codec is 3. When an SDP for a new codec is specified, no other
specified in the MMUSIC WG, no other standardization should be standardization should be required for it to be possible to use
required for it to be possible to use that in the web browsers. that in the web browsers. Adding new codecs which might have new
Adding new codecs which might have new SDP parameters should not SDP parameters should not change the APIs between the browser and
change the APIs between the browser and Javascript application. Javascript application. As soon as the browsers support the new
codecs, old applications written before the codecs were specified
As soon as the browsers support the new codecs, old applications should automatically be able to use the new codecs where
written before the codecs were specified should automatically be appropriate with no changes to the JS applications.
able to use the new codecs where appropriate with no changes to
the JS 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]. [I-D.ietf-rtcweb-jsep].
WebRTC browsers MUST implement [I-D.ietf-rtcweb-jsep]. WebRTC browsers MUST implement [I-D.ietf-rtcweb-jsep].
WebRTC endpoints MUST implement the functions described in that WebRTC endpoints MUST implement the functions described in that
document that relate to the network layer (for example Bundle document that relate to the network layer (e.g. Bundle
[I-D.ietf-mmusic-sdp-bundle-negotiation], RTCP-mux [RFC5761] and [I-D.ietf-mmusic-sdp-bundle-negotiation], RTCP-mux [RFC5761] and
Trickle ICE [I-D.ietf-ice-trickle]), but do not need to support the Trickle ICE [I-D.ietf-ice-trickle]), but do not need to support the
API functionality described there. 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 user's 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 user have some way of figuring
out where his audio, video or texting is being sent, for what out where his audio, video or texting is being sent, for what
skipping to change at page 16, line 36 skipping to change at page 16, line 46
Thanks to Alissa Cooper, Bjoern Hoehrmann, Colin Perkins, Colton Thanks to Alissa Cooper, Bjoern Hoehrmann, Colin Perkins, Colton
Shields, Eric Rescorla, Heath Matlock, Henry Sinnreich, Justin Shields, Eric Rescorla, Heath Matlock, Henry Sinnreich, Justin
Uberti, Keith Drage, Magnus Westerlund, Olle E. Johansson, Sean Uberti, Keith Drage, Magnus Westerlund, Olle E. Johansson, Sean
Turner and Simon Leinen for document review. Turner and Simon Leinen for document review.
13. References 13. References
13.1. Normative References 13.1. Normative References
[I-D.ietf-ice-rfc5245bis]
Keranen, A., Holmberg, C., and J. Rosenberg, "Interactive
Connectivity Establishment (ICE): A Protocol for Network
Address Translator (NAT) Traversal", draft-ietf-ice-
rfc5245bis-08 (work in progress), December 2016.
[I-D.ietf-rtcweb-data-channel] [I-D.ietf-rtcweb-data-channel]
Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data
Channels", draft-ietf-rtcweb-data-channel-11 (work in Channels", draft-ietf-rtcweb-data-channel-13 (work in
progress), July 2014. progress), January 2015.
[I-D.ietf-rtcweb-data-protocol] [I-D.ietf-rtcweb-data-protocol]
Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel
Establishment Protocol", draft-ietf-rtcweb-data- Establishment Protocol", draft-ietf-rtcweb-data-
protocol-07 (work in progress), July 2014. protocol-09 (work in progress), January 2015.
[I-D.ietf-rtcweb-jsep] [I-D.ietf-rtcweb-jsep]
Uberti, J., Jennings, C., and E. Rescorla, "Javascript Uberti, J., Jennings, C., and E. Rescorla, "JavaScript
Session Establishment Protocol", draft-ietf-rtcweb-jsep-07 Session Establishment Protocol", draft-ietf-rtcweb-jsep-24
(work in progress), July 2014. (work in progress), October 2017.
[I-D.ietf-rtcweb-rtp-usage] [I-D.ietf-rtcweb-rtp-usage]
Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time
Communication (WebRTC): Media Transport and Use of RTP", Communication (WebRTC): Media Transport and Use of RTP",
draft-ietf-rtcweb-rtp-usage-16 (work in progress), July draft-ietf-rtcweb-rtp-usage-26 (work in progress), March
2014. 2016.
[I-D.ietf-rtcweb-security] [I-D.ietf-rtcweb-security]
Rescorla, E., "Security Considerations for WebRTC", draft- Rescorla, E., "Security Considerations for WebRTC", draft-
ietf-rtcweb-security-07 (work in progress), July 2014. ietf-rtcweb-security-09 (work in progress), October 2017.
[I-D.ietf-rtcweb-security-arch] [I-D.ietf-rtcweb-security-arch]
Rescorla, E., "WebRTC Security Architecture", draft-ietf- Rescorla, E., "WebRTC Security Architecture", draft-ietf-
rtcweb-security-arch-10 (work in progress), July 2014. rtcweb-security-arch-13 (work in progress), October 2017.
[I-D.ietf-rtcweb-transports] [I-D.ietf-rtcweb-transports]
Alvestrand, H., "Transports for WebRTC", draft-ietf- Alvestrand, H., "Transports for WebRTC", draft-ietf-
rtcweb-transports-06 (work in progress), August 2014. 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, March 1997. Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997,
<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, June with Session Description Protocol (SDP)", RFC 3264,
2002. DOI 10.17487/RFC3264, June 2002,
<https://www.rfc-editor.org/info/rfc3264>.
[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, July 2003. 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. [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, March 2004. RFC 3711, DOI 10.17487/RFC3711, March 2004,
<https://www.rfc-editor.org/info/rfc3711>.
[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>.
[RFC7742] Roach, A., "WebRTC Video Processing and Codec [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,
<http://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,
<http://www.rfc-editor.org/info/rfc7874>. <https://www.rfc-editor.org/info/rfc7874>.
[W3C.WD-mediacapture-streams-20120628] [W3C.WD-mediacapture-streams-20120628]
Burnett, D. and A. Narayanan, "Media Capture and Streams", Burnett, D. and A. Narayanan, "Media Capture and Streams",
World Wide Web Consortium WD WD-mediacapture-streams- World Wide Web Consortium WD WD-mediacapture-streams-
20120628, June 2012, <http://www.w3.org/TR/2012/ 20120628, June 2012, <http://www.w3.org/TR/2012/
WD-mediacapture-streams-20120628>. WD-mediacapture-streams-20120628>.
[W3C.WD-webrtc-20120209] [W3C.WD-webrtc-20120209]
Bergkvist, A., Burnett, D., Jennings, C., and A. Bergkvist, A., Burnett, D., Jennings, C., and A.
Narayanan, "WebRTC 1.0: Real-time Communication Between Narayanan, "WebRTC 1.0: Real-time Communication Between
Browsers", World Wide Web Consortium WD WD-webrtc- Browsers", World Wide Web Consortium WD WD-webrtc-
20120209, February 2012, 20120209, February 2012,
<http://www.w3.org/TR/2012/WD-webrtc-20120209>. <http://www.w3.org/TR/2012/WD-webrtc-20120209>.
13.2. Informative References 13.2. Informative References
[I-D.ietf-ice-trickle] [I-D.ietf-ice-trickle]
Ivov, E., Rescorla, E., Uberti, J., and P. Saint-Andre, Ivov, E., Rescorla, E., Uberti, J., and P. Saint-Andre,
"Trickle ICE: Incremental Provisioning of Candidates for "Trickle ICE: Incremental Provisioning of Candidates for
the Interactive Connectivity Establishment (ICE) the Interactive Connectivity Establishment (ICE)
Protocol", draft-ietf-ice-trickle-07 (work in progress), Protocol", draft-ietf-ice-trickle-14 (work in progress),
February 2017. September 2017.
[I-D.ietf-mmusic-sdp-bundle-negotiation] [I-D.ietf-mmusic-sdp-bundle-negotiation]
Holmberg, C., Alvestrand, H., and C. Jennings, Holmberg, C., Alvestrand, H., and C. Jennings,
"Negotiating Media Multiplexing Using the Session "Negotiating Media Multiplexing Using the Session
Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle- Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle-
negotiation-07 (work in progress), April 2014. negotiation-39 (work in progress), August 2017.
[I-D.ietf-rtcweb-gateways] [I-D.ietf-rtcweb-gateways]
Alvestrand, H. and U. Rauschenbach, "WebRTC Gateways", Alvestrand, H. and U. Rauschenbach, "WebRTC Gateways",
draft-ietf-rtcweb-gateways-02 (work in progress), January draft-ietf-rtcweb-gateways-02 (work in progress), January
2016. 2016.
[I-D.ietf-tsvwg-rtcweb-qos]
Jones, P., Dhesikan, S., Jennings, C., and D. Druta, "DSCP
Packet Markings for WebRTC QoS", draft-ietf-tsvwg-rtcweb-
qos-18 (work in progress), August 2016.
[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,
June 2002. DOI 10.17487/RFC3261, June 2002,
<https://www.rfc-editor.org/info/rfc3261>.
[RFC3935] Alvestrand, H., "A Mission Statement for the IETF", BCP [RFC3361] Schulzrinne, H., "Dynamic Host Configuration Protocol
95, RFC 3935, October 2004. (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,
<https://www.rfc-editor.org/info/rfc3935>.
[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, April 2010. Control Packets on a Single Port", RFC 5761,
DOI 10.17487/RFC5761, April 2010,
<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, March 2011. Protocol (XMPP): Core", RFC 6120, DOI 10.17487/RFC6120,
March 2011, <https://www.rfc-editor.org/info/rfc6120>.
[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,
<http://www.rfc-editor.org/info/rfc7478>. <https://www.rfc-editor.org/info/rfc7478>.
[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,
<https://www.rfc-editor.org/info/rfc8155>.
[W3C.WD-html5-20110525] [W3C.WD-html5-20110525]
Hickson, I., "HTML5", World Wide Web Consortium LastCall Hickson, I., "HTML5", World Wide Web Consortium LastCall
WD-html5-20110525, May 2011, WD-html5-20110525, May 2011,
<http://www.w3.org/TR/2011/WD-html5-20110525>. <http://www.w3.org/TR/2011/WD-html5-20110525>.
[XEP-0124]
Paterson, I., Smith, D., Saint-Andre, P., Moffitt, J.,
Stout, L., and W. Tilanus, "BOSH", XSF XEP 0124, November
2016.
[XEP-0166] [XEP-0166]
Ludwig, S., Beda, J., Saint-Andre, P., McQueen, R., Egan, Ludwig, S., Beda, J., Saint-Andre, P., McQueen, R., Egan,
S., and J. Hildebrand, "Jingle", XSF XEP 0166, June 2007. S., and J. Hildebrand, "Jingle", XSF XEP 0166, June 2007.
Appendix A. Change log Appendix A. Change log
This section may be deleted by the RFC Editor when preparing for This section may be deleted by the RFC Editor when preparing for
publication. publication.
A.1. Changes from draft-alvestrand-dispatch-rtcweb-datagram-00 to -01 A.1. Changes from draft-alvestrand-dispatch-rtcweb-datagram-00 to -01
skipping to change at page 23, line 25 skipping to change at page 24, line 13
None. This is a "keepalive" publication. None. This is a "keepalive" publication.
A.21. Changes from -16 to -17 A.21. Changes from -16 to -17
Addressed review comments by Olle E. Johansson and Magnus Westerlund Addressed review comments by Olle E. Johansson and Magnus Westerlund
A.22. Changes from -17 to -18 A.22. Changes from -17 to -18
Addressed review comments from Sean Turner and Alissa Cooper Addressed review comments from Sean Turner and Alissa Cooper
A.23. Changes from -18 to -19
A number of grammatical issues were fixed.
Added note on operational impact of WebRTC.
Unified all definitions into the definitions list.
Added a reference for BOSH.
Changed ICE reference from 5245bis to RFC 5245.
Author's Address Author's Address
Harald T. Alvestrand Harald T. Alvestrand
Google Google
Kungsbron 2 Kungsbron 2
Stockholm 11122 Stockholm 11122
Sweden Sweden
Email: harald@alvestrand.no Email: harald@alvestrand.no
 End of changes. 68 change blocks. 
187 lines changed or deleted 244 lines changed or added

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