draft-ietf-rtcweb-overview-12.txt   draft-ietf-rtcweb-overview-13.txt 
Network Working Group H. Alvestrand Network Working Group H. Alvestrand
Internet-Draft Google Internet-Draft Google
Intended status: Standards Track October 13, 2014 Intended status: Standards Track November 28, 2014
Expires: April 16, 2015 Expires: June 1, 2015
Overview: Real Time Protocols for Browser-based Applications Overview: Real Time Protocols for Browser-based Applications
draft-ietf-rtcweb-overview-12 draft-ietf-rtcweb-overview-13
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
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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 April 16, 2015. This Internet-Draft will expire on June 1, 2015.
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
Provisions Relating to IETF Documents Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of (http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents publication of this document. Please review these documents
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include Simplified BSD License text as described in Section 4.e of include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as the Trust Legal Provisions and are provided without warranty as
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 . . . . . . . . . . 4
2.3. On interoperability and innovation . . . . . . . . . . . 5 2.3. On interoperability and innovation . . . . . . . . . . . 6
2.4. Terminology . . . . . . . . . . . . . . . . . . . . . . . 6 2.4. Terminology . . . . . . . . . . . . . . . . . . . . . . . 7
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 . . . . . . . . . . . . . . . . . . 12
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
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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 . . 20
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 . . 20
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 . . 20
A.10. Changes from -05 to -06 . . . . . . . . . . . . . . . . . 20 A.10. Changes from -05 to -06 . . . . . . . . . . . . . . . . . 20
A.11. Changes from -06 to -07 . . . . . . . . . . . . . . . . . 21 A.11. Changes from -06 to -07 . . . . . . . . . . . . . . . . . 21
A.12. Changes from -07 to -08 . . . . . . . . . . . . . . . . . 21 A.12. Changes from -07 to -08 . . . . . . . . . . . . . . . . . 21
A.13. Changes from -08 to -09 . . . . . . . . . . . . . . . . . 21 A.13. Changes from -08 to -09 . . . . . . . . . . . . . . . . . 21
A.14. Changes from -09 to -10 . . . . . . . . . . . . . . . . . 21 A.14. Changes from -09 to -10 . . . . . . . . . . . . . . . . . 21
A.15. Changes from -10 to -11 . . . . . . . . . . . . . . . . . 21 A.15. Changes from -10 to -11 . . . . . . . . . . . . . . . . . 21
A.16. Changes from -11 to -12 . . . . . . . . . . . . . . . . . 22 A.16. Changes from -11 to -12 . . . . . . . . . . . . . . . . . 22
A.17. Changes from -12 to -13 . . . . . . . . . . . . . . . . . 22
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 22 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 22
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,
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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 pieces of specifications. It defines terms used by other pieces of
specification, lists references to other specifications that don't specification, lists references to other specifications that don't
need further elaboration in the WebRTC context, and gives pointers to need further elaboration in the WebRTC context, and gives pointers to
other documents that form part of the WebRTC suite. 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 RTCWEB possible to have all information needed to implement an 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 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]
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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, five types of entities are defined: For the purpose of this document, we define the following terminology
to talk about WebRTC things:
o A WebRTC User Agent (also called a WebRTC UA or a WebRTC browser) o A WebRTC browser (also called a WebRTC User Agent or WebRTC UA) is
is something that conforms to both the protocol specification and something that conforms to both the protocol specification and the
the Javascript API defined above. Javascript API defined above.
o A WebRTC device is something that conforms to the protocol o A WebRTC non-browser is 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
application".
o A WebRTC endpoint is either a WebRTC User Agent or a WebRTC o A WebRTC endpoint is either a WebRTC browser or a WebRTC non-
device. browser. It conforms to the protocol specification.
o A WebRTC-compatible endpoint is an endpoint that is capable of o A WebRTC-compatible endpoint is an endpoint that is able to
successfully communicating with a WebRTC endpoint, but may fail to successfully communicate with a WebRTC endpoint, but may fail to
meet some requirements of a WebRTC endpoint. This may limit where meet some requirements of a WebRTC endpoint. This may limit where
in the network such an endpoint can be attached, or may limit the in the network such an endpoint can be attached, or may limit the
security guarantees that it offers to others. security guarantees that it offers to others. It is not
constrained by this specification; when it is mentioned at all, it
is to note the implications on WebRTC-compatible endpoints of the
requirements placed on WebRTC endpoints.
o A WebRTC gateway is a WebRTC-compatible endpoint that mediates o A WebRTC gateway is a WebRTC-compatible endpoint that mediates
traffic to non-WebRTC entities. media traffic to non-WebRTC entities.
All WebRTC browsers (UAs) are WebRTC devices, so any requirement on a All WebRTC browsers are WebRTC endpoints, so any requirement on a
WebRTC device also applies to a WebRTC browser. WebRTC endpoint also applies to a WebRTC browser.
WebRTC gateways are described in a separate document A WebRTC non-browser may be capable of hosting applications in a
similar way to the way in which a browser can host Javascript
applications, typically by offering APIs in other languages. For
instance it may be implemented as a library that offers a C++ API
intended to be loaded into applications. In this case, similar
security considerations as for Javascript may be needed; however,
since such APIs are not defined or referenced here, this document
cannot give any specific rules for those interfaces.
WebRTC gateways are described in a separate document,
[I-D.alvestrand-rtcweb-gateways]. [I-D.alvestrand-rtcweb-gateways].
2.3. On interoperability and innovation 2.3. On interoperability and innovation
The "Mission statement of the IETF" [RFC3935] states that "The The "Mission statement 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:
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Interactive: Communication between multiple parties, where the Interactive: Communication between multiple parties, where the
expectation is that an action from one party can cause a reaction expectation is that an action from one party can cause a reaction
by another party, and the reaction can be observed by the first by another party, and the reaction can be observed by the first
party, with the total time required for the action/reaction/ party, with the total time required for the action/reaction/
observation is on the order of no more than hundreds of observation is on the order of no more than hundreds of
milliseconds. milliseconds.
Media: Audio and video content. Not to be confused with Media: Audio and video content. Not to be confused with
"transmission media" such as wires. "transmission media" such as wires.
Media path: The path that media data follows from one WebRTC device Media path: The path that media data follows from one WebRTC
to another. endpoint to another.
Protocol: A specification of a set of data units, their Protocol: A specification of a set of data units, their
representation, and rules for their transmission, with their representation, and rules for their transmission, with their
defined semantics. A protocol is usually thought of as going defined semantics. A protocol is usually thought of as going
between systems. between systems.
Real-time media: Media where generation of content and display of Real-time media: Media where generation of content and display of
content are intended to occur closely together in time (on the content are intended to occur closely together in time (on the
order of no more than hundreds of milliseconds). Real-time media order of no more than hundreds of milliseconds). Real-time media
can be used to support interactive communication. can be used to support interactive communication.
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SDP Agent: The protocol implementation involved in the SDP offer/ SDP Agent: The protocol implementation involved in the SDP offer/
answer exchange, as defined in [RFC3264] section 3. answer exchange, as defined in [RFC3264] section 3.
Signaling: Communication that happens in order to establish, manage Signaling: Communication that happens in order to establish, manage
and control media paths. and 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.
WebRTC User Agent: An entity that conforms to the WebRTC protocol
specifications and offer the WebRTC Javascript APIs. Also called
a WebRTC browser.
WebRTC Device: An unit (software, hardware or combinations) that
conforms to the WebRTC protocol specifications, but does not offer
the WebRTC Javascript APIs.
WebRTC Endpoint: Either a WebRTC browser or a WebRTC device.
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.
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 assume that the browser will contain all the functions that need not assume that the browser will contain all the functions that need
to be performed in order to have a function such as a telephone or a to be performed in order to have a function such as a telephone or a
video conferencing unit; the vision is that the browser will have the video conferencing unit; the vision is that the browser will have the
functions that are needed for a Web application, working in functions that are needed for a Web application, working in
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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 HTTP 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 device 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.
+-----------+ +-----------+ +-----------+ +-----------+
| Web | | Web | | Web | | Web |
| | Signaling | | | | Signaling | |
| |-------------| | | |-------------| |
| Server | path | Server | | Server | path | Server |
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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.
WebRTC devices 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
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 WebRTC use case are in considerations for the WebRTC 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 a supporting described in [I-D.ietf-rtcweb-data-channel], and a supporting
protocol for establishing individual data channels is described in protocol for establishing individual data channels is described in
[I-D.ietf-rtcweb-data-protocol]. Webrtc devices MUST implement these [I-D.ietf-rtcweb-data-protocol]. WebRTC endpoints MUST implement
two specifications. these two specifications.
WebRTC devices MUST implement [I-D.ietf-rtcweb-rtp-usage], WebRTC endpoints MUST implement [I-D.ietf-rtcweb-rtp-usage],
[I-D.ietf-rtcweb-security], [I-D.ietf-rtcweb-security-arch], and the [I-D.ietf-rtcweb-security], [I-D.ietf-rtcweb-security-arch], and the
requirements they include. requirements they include.
6. Data formats 6. Data formats
The intent of this specification is to allow each communications The intent of this specification is to allow each communications
event to use the data formats that are best suited for that event to use the data formats that are best suited for that
particular instance, where a format is supported by both sides of the particular instance, where a format is supported by both sides of the
connection. However, a minimum standard is greatly helpful in order connection. However, a minimum standard is greatly helpful in order
to ensure that communication can be achieved. This document to ensure that communication can be achieved. This document
specifies a minimum baseline that will be supported by all specifies a minimum baseline that will be supported by all
implementations of this specification, and leaves further codecs to implementations of this specification, and leaves further codecs to
be included at the will of the implementor. be included at the will of the implementor.
WebRTC devices MUST implement the codecs and profiles required in WebRTC endpoints that support audio and/or video MUST implement the
[I-D.ietf-rtcweb-audio] codecs and profiles required in [I-D.ietf-rtcweb-audio] and
[I-D.ietf-rtcweb-video].
NOTE IN DRAFT: At this time (October 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 WebRTC media negotiations will be capable of representing the 1. The WebRTC media negotiations will be capable of representing the
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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 WebRTC, and their implications for The particular choices made for WebRTC, and their implications for
the API offered by a WebRTC endpoint, 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 devices 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, RTCP- document that relate to the network layer (for example Bundle, RTCP-
mux and Trickle ICE), but do not need to support the API mux and Trickle ICE), but do not need to support the API
functionality described there. 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
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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 WebRTC devices with regard to audio processing The requirements on WebRTC 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 devices MUST implement the processing functions in WebRTC endpoints MUST implement the processing functions in
[I-D.ietf-rtcweb-audio]. (Together with the requirement inSection 6, [I-D.ietf-rtcweb-audio]. (Together with the requirement inSection 6,
this means that WebRTC devices MUST implement the whole document.) this means that WebRTC endpoints 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
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the ASCII drawings in section 1. the ASCII drawings in section 1.
Thanks to Bjoern Hoehrmann, Colin Perkins, Colton Shields, Eric Thanks to Bjoern Hoehrmann, Colin Perkins, Colton Shields, Eric
Rescorla, Heath Matlock, Henry Sinnreich, Justin Uberti, Keith Drage Rescorla, Heath Matlock, Henry Sinnreich, Justin Uberti, Keith Drage
and Simon Leinen for document review. and Simon Leinen for document review.
13. References 13. References
13.1. Normative References 13.1. Normative References
[I-D.alvestrand-rtcweb-gateways]
Alvestrand, H., "WebRTC Gateways", draft-alvestrand-
rtcweb-gateways-00 (work in progress), August 2014.
[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-06 (work in Requirements", draft-ietf-rtcweb-audio-05 (work in
progress), September 2014. progress), February 2014.
[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-12 (work in Channels", draft-ietf-rtcweb-data-channel-11 (work in
progress), September 2014. progress), July 2014.
[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-08 (work in progress), September 2014. protocol-07 (work in progress), July 2014.
[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-07
(work in progress), July 2014. (work in progress), July 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-17 (work in progress), August draft-ietf-rtcweb-rtp-usage-16 (work in progress), July
2014. 2014.
[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-07 (work in progress), July 2014.
[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-10 (work in progress), July 2014.
[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-06 (work in progress), August 2014.
[I-D.ietf-rtcweb-video]
Roach, A., "WebRTC Video Processing and Codec
Requirements", draft-ietf-rtcweb-video-00 (work in
progress), July 2014.
[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, 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, April Traversal for Offer/Answer Protocols", RFC 5245, April
2010. 2010.
[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- World Wide Web Consortium WD WD-mediacapture-streams-
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- Browsers", World Wide Web Consortium WD WD-webrtc-
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.alvestrand-rtcweb-gateways]
Alvestrand, H., "WebRTC Gateways", draft-alvestrand-
rtcweb-gateways-00 (work in progress), August 2014.
[I-D.ietf-rtcweb-use-cases-and-requirements] [I-D.ietf-rtcweb-use-cases-and-requirements]
Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real- Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
Time Communication Use-cases and Requirements", draft- Time Communication Use-cases and Requirements", draft-
ietf-rtcweb-use-cases-and-requirements-14 (work in ietf-rtcweb-use-cases-and-requirements-14 (work in
progress), February 2014. progress), February 2014.
[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. June 2002.
skipping to change at page 21, line 47 skipping to change at page 21, line 47
Added notes on which referenced documents WebRTC browsers or devices Added notes on which referenced documents WebRTC browsers or devices
MUST conform to. MUST conform to.
Added pointer to the security section of the API drafts. Added pointer to the security section of the API drafts.
A.15. Changes from -10 to -11 A.15. Changes from -10 to -11
Added "WebRTC Gateway" as a third class of device, and referenced the Added "WebRTC Gateway" as a third class of device, and referenced the
doc describing them. doc describing them.
Made a number of text clarifications, in response to document Made a number of text clarifications in response to document reviews.
reviews.
A.16. Changes from -11 to -12 A.16. Changes from -11 to -12
Refined entity definitions to define "WebRTC endpoint" and "WebRTC- Refined entity definitions to define "WebRTC endpoint" and "WebRTC-
compatible endpoint". compatible endpoint".
Changed remaining usage of the term "RTCWEB" to "WebRTC", including Changed remaining usage of the term "RTCWEB" to "WebRTC", including
in the page header. in the page header.
A.17. Changes from -12 to -13
Changed "WebRTC device" to be "WebRTC non-browser", per decision at
IETF 91. This led to the need for "WebRTC endpoint" as the common
label for both, and the usage of that term in the rest of the
document.
Added words about WebRTC APIs in languages other than Javascript.
Referenced draft-ietf-rtcweb-video for video codecs to support.
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. 39 change blocks. 
64 lines changed or deleted 83 lines changed or added

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