draft-ietf-rtcweb-overview-09.txt   draft-ietf-rtcweb-overview-10.txt 
Network Working Group H. Alvestrand Network Working Group H. Alvestrand
Internet-Draft Google Internet-Draft Google
Intended status: Standards Track February 14, 2014 Intended status: Standards Track June 17, 2014
Expires: August 18, 2014 Expires: December 19, 2014
Overview: Real Time Protocols for Brower-based Applications Overview: Real Time Protocols for Browser-based Applications
draft-ietf-rtcweb-overview-09 draft-ietf-rtcweb-overview-10
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
specified and on the right publication track. specified and on the right publication track.
The document will be publishd as an Applicability Statement - it does This document is an Applicability Statement - it does not itself
not itself specify any protocol, but specifies which other specify any protocol, but specifies which other specifications RTCWEB
specifications RTCWEB compliant implementations are supposed to compliant implementations are supposed to follow.
follow.
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 http://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 August 18, 2014. This Internet-Draft will expire on December 19, 2014.
Copyright Notice Copyright Notice
Copyright (c) 2014 IETF Trust and the persons identified as the Copyright (c) 2014 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal This document is subject to BCP 78 and the IETF Trust's Legal
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to this document. Code Components extracted from this document must to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of include Simplified BSD License text as described in Section 4.e of
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described in the Simplified BSD License. described in the Simplified BSD License.
Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Principles and Terminology . . . . . . . . . . . . . . . . . . 5 2. Principles and Terminology . . . . . . . . . . . . . . . . . 4
2.1. Goals of this document . . . . . . . . . . . . . . . . . . 5 2.1. Goals of this document . . . . . . . . . . . . . . . . . 4
2.2. Relationship between API and protocol . . . . . . . . . . 5 2.2. Relationship between API and protocol . . . . . . . . . . 4
2.3. On interoperability and innovation . . . . . . . . . . . . 6 2.3. On interoperability and innovation . . . . . . . . . . . 5
2.4. Terminology . . . . . . . . . . . . . . . . . . . . . . . 7 2.4. Terminology . . . . . . . . . . . . . . . . . . . . . . . 6
3. Architecture and Functionality groups . . . . . . . . . . . . 8 3. Architecture and Functionality groups . . . . . . . . . . . . 7
4. Data transport . . . . . . . . . . . . . . . . . . . . . . . . 12 4. Data transport . . . . . . . . . . . . . . . . . . . . . . . 11
5. Data framing and securing . . . . . . . . . . . . . . . . . . 12 5. Data framing and securing . . . . . . . . . . . . . . . . . . 11
6. Data formats . . . . . . . . . . . . . . . . . . . . . . . . . 13 6. Data formats . . . . . . . . . . . . . . . . . . . . . . . . 12
7. Connection management . . . . . . . . . . . . . . . . . . . . 13 7. Connection management . . . . . . . . . . . . . . . . . . . . 12
8. Presentation and control . . . . . . . . . . . . . . . . . . . 14 8. Presentation and control . . . . . . . . . . . . . . . . . . 13
9. Local system support functions . . . . . . . . . . . . . . . . 14 9. Local system support functions . . . . . . . . . . . . . . . 13
10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 15 10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 14
11. Security Considerations . . . . . . . . . . . . . . . . . . . 15 11. Security Considerations . . . . . . . . . . . . . . . . . . . 14
12. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 16 12. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 15
13. References . . . . . . . . . . . . . . . . . . . . . . . . . . 16 13. References . . . . . . . . . . . . . . . . . . . . . . . . . 15
13.1. Normative References . . . . . . . . . . . . . . . . . . . 16 13.1. Normative References . . . . . . . . . . . . . . . . . . 15
13.2. Informative References . . . . . . . . . . . . . . . . . . 17 13.2. Informative References . . . . . . . . . . . . . . . . . 17
Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 18 Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 17
A.1. Changes from A.1. Changes from draft-alvestrand-dispatch-rtcweb-datagram-00
draft-alvestrand-dispatch-rtcweb-datagram-00 to -01 . . . 18 to -01 . . . . . . . . . . . . . . . . . . . . . . . . . 17
A.2. Changes from draft-alvestrand-dispatch-01 to A.2. Changes from draft-alvestrand-dispatch-01 to draft-
draft-alvestrand-rtcweb-overview-00 . . . . . . . . . . . 19 alvestrand-rtcweb-overview-00 . . . . . . . . . . . . . . 18
A.3. Changes from draft-alvestrand-rtcweb-00 to -01 . . . . . . 19 A.3. Changes from draft-alvestrand-rtcweb-00 to -01 . . . . . 18
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 . . . . . . . . . . . . . . 19 draft-ietf-rtcweb-overview-00 . . . . . . . . . . . . . . 18
A.5. Changes from -00 to -01 of draft-ietf-rtcweb-overview . . 19 A.5. Changes from -00 to -01 of draft-ietf-rtcweb-overview . . 18
A.6. Changes from -01 to -02 of draft-ietf-rtcweb-overview . . 19 A.6. Changes from -01 to -02 of draft-ietf-rtcweb-overview . . 18
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 . . 19
A.8. Changes from -03 to -04 of draft-ietf-rtcweb-overview . . 20 A.8. Changes from -03 to -04 of draft-ietf-rtcweb-overview . . 19
A.9. Changes from -04 to -05 of draft-ietf-rtcweb-overview . . 20 A.9. Changes from -04 to -05 of draft-ietf-rtcweb-overview . . 19
A.10. Changes from -05 to -06 . . . . . . . . . . . . . . . . . 20 A.10. Changes from -05 to -06 . . . . . . . . . . . . . . . . . 19
A.11. Changes from -06 to -07 . . . . . . . . . . . . . . . . . 20 A.11. Changes from -06 to -07 . . . . . . . . . . . . . . . . . 19
A.12. Changes from -07 to -08 . . . . . . . . . . . . . . . . . 21 A.12. Changes from -07 to -08 . . . . . . . . . . . . . . . . . 20
A.13. Changes from -08 to -09 . . . . . . . . . . . . . . . . . 21 A.13. Changes from -08 to -09 . . . . . . . . . . . . . . . . . 20
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 21 A.14. Changes from -09 to -10 . . . . . . . . . . . . . . . . . 20
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 20
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
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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 RTCWEB use cases document described as required scenarios in the RTCWEB use cases document
[I-D.ietf-rtcweb-use-cases-and-requirements]. [I-D.ietf-rtcweb-use-cases-and-requirements].
Other efforts, for instance the W3C WebRTC, Web Applications and Other efforts, for instance the W3C WEBRTC, Web Applications and
Device API working groups, focus on making standardized APIs and Device API working groups, focus on making standardized APIs and
interfaces available, within or alongside the HTML5 effort, for those interfaces available, within or alongside the HTML5 effort, for those
functions; this memo concentrates on specifying the protocols and functions; this memo concentrates on specifying the protocols and
subprotocols that are needed to specify the interactions that happen subprotocols that are needed to specify the interactions that happen
across the network. across the network.
This memo uses the term "WebRTC" (note the case used) to refer to the
overall effort consisting of both IETF and W3C efforts.
2. Principles and Terminology 2. Principles and Terminology
2.1. Goals of this document 2.1. Goals of this document
The goal of the RTCWEB protocol specification is to specify a set of The goal of the RTCWEB 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.
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specification, lists references to other specifications that don't specification, lists references to other specifications that don't
need further elaboration in the RTCWEB context, and gives pointers to need further elaboration in the RTCWEB context, and gives pointers to
other documents that form part of the RTCWEB suite. other documents that form part of the RTCWEB 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 RTCWEB possible to have all information needed to implement an RTCWEB
compatible implementation. compatible implementation.
2.2. Relationship between API and protocol 2.2. Relationship between API and protocol
The total RTCWEB/WEBRTC effort consists of two pieces: The total WebRTC effort consists of two pieces:
o A protocol specification, done in the IETF o A protocol specification, done in the IETF
o A Javascript API specification, done in the W3C o A Javascript API specification, done in the W3C
[W3C.WD-webrtc-20120209][W3C.WD-mediacapture-streams-20120628] [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, viewed in any compatible where Javascript embedded in any page, viewed in any compatible
browser, when suitably authorized by its user, is able to set up browser, when suitably authorized by its user, is able to set up
communication using audio, video and auxiliary data, where the communication using audio, video and auxiliary data, where the
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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, two classes of things that can
claim conformance are defined:
o A WebRTC browser is something that conforms to both the protocol
specification and the Javascript API defined above.
o A WebRTC device is something that conforms to the protocol
specification, but does not claim to implement the Javascript API.
All WebRTC browsers are WebRTC devices, so any requirement on a
WebRTC device also applies to a WebRTC browser.
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 of 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 o Two parties communicate, through some mechanism, what
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Signaling: Communication that happens in order to establish, manage Signaling: Communication that happens in order to establish, manage
and control media paths. and control media 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.
NOTE: Where common definitions exist for these terms, those NOTE: Where common definitions exist for these terms, those
definitions should be used to the greatest extent possible. definitions should be used to the greatest extent possible.
TODO: Extend this list with other terms that might prove slippery.
3. Architecture and Functionality groups 3. Architecture and Functionality groups
The model of real-time support for browser-based applications does The model of real-time support for browser-based applications does
not envisage that the browser will contain all the functions that not assume that the browser will contain all the functions that need
need to be performed in order to have a function such as a telephone to be performed in order to have a function such as a telephone or a
or a video conferencing unit; the vision is that the browser will video conferencing unit; the vision is that the browser will have the
have the functions that are needed for a Web application, working in functions that are needed for a Web application, working in
conjunction with its backend servers, to implement these functions. conjunction with its backend servers, to implement these functions.
This means that two vital interfaces need specification: The This means that two vital interfaces need specification: The
protocols that browsers talk to each other, without any intervening protocols that browsers talk to each other, without any intervening
servers, and the APIs that are offered for a Javascript application servers, and the APIs that are offered for a Javascript application
to take advantage of the browser's functionality. to take advantage of the browser's functionality.
+------------------------+ On-the-wire +------------------------+ On-the-wire
| | Protocols | | Protocols
| Servers |---------> | Servers |--------->
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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 RTCWEB protocol suite; the conformant to the specifications of the RTCWEB 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 massage 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 or XMPP) could be used between servers, while either (for example SIP [RFC3261] or XMPP [RFC6120]) could be used between
a standards-based or proprietary protocol could be used between the servers, while either a standards-based or proprietary protocol could
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 XMPP, XMPP could be used for communication between
XMPP servers, with either a standardized signaling mechanism (e.g. XMPP servers, with either a standardized signaling mechanism (e.g.
XMPP over Websockets or BOSH) or a proprietary signaling mechanism XMPP over Websockets or BOSH) or a proprietary signaling mechanism
used between the application running in the browser and the web used between the application running in the browser and the web
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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: When not to
send data. send data.
The data transport protocols used by RTCWEB are described in WebRTC devices 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
are given in [I-D.ietf-rtcweb-rtp-usage]. The security are given in [I-D.ietf-rtcweb-rtp-usage]. The security
considerations for the RTCWEB use case are in considerations for the RTCWEB use case are in
[I-D.ietf-rtcweb-security], and the resulting security functions are [I-D.ietf-rtcweb-security], and the resulting security functions are
described in [I-D.ietf-rtcweb-security-arch]. 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 is
described in [I-D.ietf-rtcweb-data-channel], and the resulting described in [I-D.ietf-rtcweb-data-channel], and a supporting
protocol is described in [I-D.jesup-rtcweb-data-protocol] (not yet a protocol is described in [I-D.ietf-rtcweb-data-protocol]. Webrtc
WG document) devices MUST implement these two specifications.
WebRTC devices MUST implement [I-D.ietf-rtcweb-rtp-usage],
[I-D.ietf-rtcweb-security], [I-D.ietf-rtcweb-security-arch], and the
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 included at the will of the implementor. be included at the will of the implementor.
The mandatory to implement codecs, as well as any profiling WebRTC devices MUST implement the codecs and profiles required in
requirements for both mandatory and optional codecs, is described in [I-D.ietf-rtcweb-audio]
<WORKING GROUP DRAFT "MEDIA PROCESSING"> (candidate draft:
[I-D.cbran-rtcweb-codec]. NOTE IN DRAFT: At this time (June 2014) there is no consensus on what
to say about video codecs in this section.
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 RTCWEB media negotiations will be capable of representing the 1. The RTCWEB media negotiations will be capable of representing the
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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 a new codec is specified, and the SDP for the new codec is
specified in the MMUSIC WG, no other standardization should be specified in the MMUSIC WG, no other standardization should be
required for it to be possible to use that in the web browsers. required for it to be possible to use that in the web browsers.
Adding new codecs which might have new SDP parameters should not Adding new codecs which might have new SDP parameters should not
change the APIs between the browser and Javascript application. change the APIs between the browser and Javascript application.
As soon as the browsers support the new codecs, old applications As soon as the browsers support the new codecs, old applications
written before the codecs were specified should automatically be written before the codecs were specified should automatically be
able to use the new codecs where appropriate with no changes to able to use the new codecs where appropriate with no changes to
the JS applications. the JS applications.
The particular choices made for RTCWEB, and their implications for The particular choices made for RTCWEB, and their implications for
the API offered by a browser implementing RTCWEB, are described in the API offered by a browser implementing RTCWEB, are described in
[I-D.ietf-rtcweb-jsep]. This document in turn implements the [I-D.ietf-rtcweb-jsep].
solutions described in [I-D.roach-mmusic-unified-plan].
WebRTC browsers MUST implement [I-D.ietf-rtcweb-jsep].
NOTE IN DRAFT: Is there any part of -jsep that WebRTC devices need to
be required to implement, and are not also required via other paths?
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
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 being worked on as part of the W3C API effort, and interface; this is specified in the peer connection API
will be part of the peer connection API [W3C.WD-webrtc-20120209], and [W3C.WD-webrtc-20120209], and the media capture API
the media capture API [W3C.WD-mediacapture-streams-20120628]. [W3C.WD-mediacapture-streams-20120628].
Considerations for the implications of wanting to identify
correspondents are described in [I-D.rescorla-rtcweb-generic-idp] WebRTC browsers MUST implement these two specifications.
(not a WG item).
9. Local system support functions 9. Local system support functions
These are characterized by the fact that the quality of these These are characterized by the fact that the quality of these
functions strongly influence the user experience, but the exact functions strongly influence 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.
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management including focus, zoom, pan/tilt controls (if available), management including focus, zoom, pan/tilt controls (if available),
and more. and more.
Certain parts of the system SHOULD conform to certain properties, for Certain parts of the system SHOULD conform to certain properties, for
instance: instance:
o Echo cancellation should be good enough to achieve the suppression o 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 o Privacy concerns MUST be satisfied; for instance, if remote
control of camera is offered, the APIs should be available to let control of camera is offered, the APIs should be available to let
the local participant figure out who's controlling the camera, and the local participant figure out who's controlling the camera, and
possibly decide to revoke the permission for camera usage. possibly decide to revoke the permission for camera usage.
o Automatic gain control, if present, should normalize a speaking o Automatic gain control, if present, should normalize a speaking
voice into a reasonable dB range. voice into a reasonable dB range.
The requirements on RTCWEB systems with regard to audio processing The requirements on RTCWEB systems with regard to audio processing
are found in [I-D.ietf-rtcweb-audio]; the proposed API for control of are found in [I-D.ietf-rtcweb-audio]; the proposed API for control of
local devices are found in [W3C.WD-mediacapture-streams-20120628]. local devices are found in [W3C.WD-mediacapture-streams-20120628].
WebRTC browsers MUST implement the processing functions in
[I-D.ietf-rtcweb-audio]. (Together with the requirement inSection 6,
this means that browsers MUST implement the whole document.)
10. IANA Considerations 10. IANA Considerations
This document makes no request of IANA. This document makes no request of IANA.
Note to RFC Editor: this section may be removed on publication as an Note to RFC Editor: this section may be removed on publication as an
RFC. 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
skipping to change at page 16, line 5 skipping to change at page 15, line 8
measures are taken. measures are taken.
o Security of the partners' identity: verifying that the o Security of the partners' identity: 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 identity 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, is contained in [I-D.ietf-rtcweb-security].
It is also important to read the security sections of
[W3C.WD-mediacapture-streams-20120628] and [W3C.WD-webrtc-20120209].
12. Acknowledgements 12. Acknowledgements
The number of people who have taken part in the discussions The number of people who have taken part in the discussions
surrounding this draft are too numerous to list, or even to identify. surrounding this draft are too numerous to list, or even to identify.
The ones below have made special, identifiable contributions; this The ones below have made special, identifiable contributions; this
does not mean that others' contributions are less important. does not mean that others' contributions are less important.
Thanks to Cary Bran, Cullen Jennings, Colin Perkins, Magnus Thanks to Cary Bran, Cullen Jennings, Colin Perkins, Magnus
Westerlund and Joerg Ott, who offered technical contributions on Westerlund and Joerg Ott, who offered technical contributions on
various versions of the draft. various versions of the draft.
Thanks to Jonathan Rosenberg, Matthew Kaufman and others at Skype for Thanks to Jonathan Rosenberg, Matthew Kaufman and others at Skype for
the ASCII drawings in section 1. the ASCII drawings in section 1.
Thanks to Eric Rescorla, Justin Uberti, Henry Sinnreich, Colin Thanks to Bjoern Hoehrmann, Colin Perkins, Colton Shields, Eric
Perkins, Bjoern Hoehrmann and Simon Leinen for document review, and Rescorla, Heath Matlock, Henry Sinnreich, Justin Uberti, Keith Drage
to Heath Matlock for grammatical review. and Simon Leinen for document review.
13. References 13. References
13.1. Normative References 13.1. Normative References
[I-D.ietf-rtcweb-audio] [I-D.ietf-rtcweb-audio]
Valin, J. and C. Bran, "WebRTC Audio Codec and Processing Valin, J. and C. Bran, "WebRTC Audio Codec and Processing
Requirements", draft-ietf-rtcweb-audio-02 (work in Requirements", draft-ietf-rtcweb-audio-05 (work in
progress), August 2013. progress), February 2014.
[I-D.ietf-rtcweb-data-channel] [I-D.ietf-rtcweb-data-channel]
Jesup, R., Loreto, S., and M. Tuexen, "RTCWeb Data Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data
Channels", draft-ietf-rtcweb-data-channel-05 (work in Channels", draft-ietf-rtcweb-data-channel-10 (work in
progress), July 2013. progress), June 2014.
[I-D.ietf-rtcweb-data-protocol]
Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel
Establishment Protocol", draft-ietf-rtcweb-data-
protocol-06 (work in progress), June 2014.
[I-D.ietf-rtcweb-jsep] [I-D.ietf-rtcweb-jsep]
Uberti, J. and C. Jennings, "Javascript Session Uberti, J. and C. Jennings, "Javascript Session
Establishment Protocol", draft-ietf-rtcweb-jsep-04 (work Establishment Protocol", draft-ietf-rtcweb-jsep-06 (work
in progress), September 2013. in progress), February 2014.
[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-09 (work in progress), draft-ietf-rtcweb-rtp-usage-15 (work in progress), May
September 2013. 2014.
[I-D.ietf-rtcweb-security] [I-D.ietf-rtcweb-security]
Rescorla, E., "Security Considerations for WebRTC", Rescorla, E., "Security Considerations for WebRTC", draft-
draft-ietf-rtcweb-security-05 (work in progress), ietf-rtcweb-security-06 (work in progress), January 2014.
July 2013.
[I-D.ietf-rtcweb-security-arch] [I-D.ietf-rtcweb-security-arch]
Rescorla, E., "WebRTC Security Architecture", Rescorla, E., "WebRTC Security Architecture", draft-ietf-
draft-ietf-rtcweb-security-arch-07 (work in progress), rtcweb-security-arch-09 (work in progress), February 2014.
July 2013.
[I-D.ietf-rtcweb-transports] [I-D.ietf-rtcweb-transports]
Alvestrand, H., "Transports for RTCWEB", Alvestrand, H., "Transports for RTCWEB", draft-ietf-
draft-ietf-rtcweb-transports-01 (work in progress), rtcweb-transports-05 (work in progress), June 2014.
September 2013.
[I-D.roach-mmusic-unified-plan]
Roach, A., Uberti, J., and M. Thomson, "A Unified Plan for
Using SDP with Large Numbers of Media Flows",
draft-roach-mmusic-unified-plan-00 (work in progress),
July 2013.
[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, June
June 2002. 2002.
[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, July 2003.
[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, March 2004.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT) (ICE): A Protocol for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols", RFC 5245, Traversal for Offer/Answer Protocols", RFC 5245, April
April 2010. 2010.
13.2. Informative References
[I-D.cbran-rtcweb-codec]
Bran, C., Jennings, C., and J. Valin, "WebRTC Codec and
Media Processing Requirements",
draft-cbran-rtcweb-codec-02 (work in progress),
March 2012.
[I-D.ietf-rtcweb-use-cases-and-requirements]
Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
Time Communication Use-cases and Requirements",
draft-ietf-rtcweb-use-cases-and-requirements-11 (work in
progress), June 2013.
[I-D.jesup-rtcweb-data-protocol]
Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel
Protocol", draft-jesup-rtcweb-data-protocol-04 (work in
progress), February 2013.
[I-D.rescorla-rtcweb-generic-idp]
Rescorla, E., "RTCWEB Generic Identity Provider
Interface", draft-rescorla-rtcweb-generic-idp-01 (work in
progress), March 2012.
[RFC3935] Alvestrand, H., "A Mission Statement for the IETF",
BCP 95, RFC 3935, October 2004.
[W3C.WD-html5-20110525]
Hickson, I., "HTML5", World Wide Web Consortium
LastCall WD-html5-20110525, May 2011,
<http://www.w3.org/TR/2011/WD-html5-20110525>.
[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
[I-D.ietf-rtcweb-use-cases-and-requirements]
Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
Time Communication Use-cases and Requirements", draft-
ietf-rtcweb-use-cases-and-requirements-14 (work in
progress), February 2014.
[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
A., Peterson, J., Sparks, R., Handley, M., and E.
Schooler, "SIP: Session Initiation Protocol", RFC 3261,
June 2002.
[RFC3935] Alvestrand, H., "A Mission Statement for the IETF", BCP
95, RFC 3935, October 2004.
[RFC6120] Saint-Andre, P., "Extensible Messaging and Presence
Protocol (XMPP): Core", RFC 6120, March 2011.
[W3C.WD-html5-20110525]
Hickson, I., "HTML5", World Wide Web Consortium LastCall
WD-html5-20110525, May 2011,
<http://www.w3.org/TR/2011/WD-html5-20110525>.
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
Added section "On interoperability and innovation" Added section "On interoperability and innovation"
Added data confidentiality and integrity to the "data framing" layer Added data confidentiality and integrity to the "data framing" layer
skipping to change at page 19, line 4 skipping to change at page 17, line 45
Added section "On interoperability and innovation" Added section "On interoperability and innovation"
Added data confidentiality and integrity to the "data framing" layer Added data confidentiality and integrity to the "data framing" layer
Added congestion management requirements in the "data transport" Added congestion management requirements in the "data transport"
layer section layer section
Changed need for non-media data from "question: do we need this?" to Changed need for non-media data from "question: do we need this?" to
"Open issue: How do we do this?" "Open issue: How do we do this?"
Strengthened disclaimer that listed codecs are placeholders, not Strengthened disclaimer that listed codecs are placeholders, not
decisions. decisions.
More details on why the "local system support functions" section is More details on why the "local system support functions" section is
there. there.
A.2. Changes from draft-alvestrand-dispatch-01 to A.2. Changes from draft-alvestrand-dispatch-01 to draft-alvestrand-
draft-alvestrand-rtcweb-overview-00 rtcweb-overview-00
Added section on "Relationship between API and protocol" Added section on "Relationship between API and protocol"
Added terminology section Added terminology section
Mentioned congestion management as part of the "data transport" layer Mentioned congestion management as part of the "data transport" layer
in the layer list in the layer list
A.3. Changes from draft-alvestrand-rtcweb-00 to -01 A.3. Changes from draft-alvestrand-rtcweb-00 to -01
Removed most technical content, and replaced with pointers to drafts Removed most technical content, and replaced with pointers to drafts
as requested and identified by the RTCWEB WG chairs. as requested and identified by the RTCWEB WG chairs.
Added content to acknowledgments section. Added content to acknowledgments section.
Added change log. Added change log.
Spell-checked document. Spell-checked document.
A.4. Changes from draft-alvestrand-rtcweb-overview-01 to A.4. Changes from draft-alvestrand-rtcweb-overview-01 to draft-ietf-
draft-ietf-rtcweb-overview-00 rtcweb-overview-00
Changed draft name and document date. Changed draft name and document date.
Removed unused references Removed unused references
A.5. Changes from -00 to -01 of draft-ietf-rtcweb-overview A.5. Changes from -00 to -01 of draft-ietf-rtcweb-overview
Added architecture figures to section 2. Added architecture figures to section 2.
Changed the description of "echo cancellation" under "local system Changed the description of "echo cancellation" under "local system
skipping to change at page 20, line 50 skipping to change at page 19, line 44
Minor grammatical fixes. This is mainly a "keepalive" refresh. Minor grammatical fixes. This is mainly a "keepalive" refresh.
A.10. Changes from -05 to -06 A.10. Changes from -05 to -06
Clarifications in response to Last Call review comments. Inserted Clarifications in response to Last Call review comments. Inserted
reference to draft-ietf-rtcweb-audio. reference to draft-ietf-rtcweb-audio.
A.11. Changes from -06 to -07 A.11. Changes from -06 to -07
Added a refereence to the "unified plan" draft, and updated some Added a reference to the "unified plan" draft, and updated some
references. references.
Otherwise, it's a "keepalive" draft. Otherwise, it's a "keepalive" draft.
A.12. Changes from -07 to -08 A.12. Changes from -07 to -08
Removed the appendix that detailed transports, and replaced it with a Removed the appendix that detailed transports, and replaced it with a
reference to draft-ietf-rtcweb-transports. Removed now-unused reference to draft-ietf-rtcweb-transports. Removed now-unused
references. references.
A.13. Changes from -08 to -09 A.13. Changes from -08 to -09
Added text to the Abstract indicating that the intended status is an Added text to the Abstract indicating that the intended status is an
Applicability Statement. 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.
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. 41 change blocks. 
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