draft-ietf-rtcweb-overview-11.txt   draft-ietf-rtcweb-overview-12.txt 
Network Working Group H. Alvestrand Network Working Group H. Alvestrand
Internet-Draft Google Internet-Draft Google
Intended status: Standards Track August 18, 2014 Intended status: Standards Track October 13, 2014
Expires: February 19, 2015 Expires: April 16, 2015
Overview: Real Time Protocols for Browser-based Applications Overview: Real Time Protocols for Browser-based Applications
draft-ietf-rtcweb-overview-11 draft-ietf-rtcweb-overview-12
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.
This document is an Applicability Statement - it does not itself This document is an Applicability Statement - it does not itself
specify any protocol, but specifies which other specifications RTCWEB specify any protocol, but specifies which other specifications WebRTC
compliant implementations are supposed to follow. compliant implementations are supposed to 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 February 19, 2015. This Internet-Draft will expire on April 16, 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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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 . . . . . . . . . . . 5
2.4. Terminology . . . . . . . . . . . . . . . . . . . . . . . 6 2.4. Terminology . . . . . . . . . . . . . . . . . . . . . . . 6
3. Architecture and Functionality groups . . . . . . . . . . . . 7 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
13. References . . . . . . . . . . . . . . . . . . . . . . . . . 16 13. References . . . . . . . . . . . . . . . . . . . . . . . . . 16
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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 . . 20
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
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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the development of HTML5, application developers see much promise in the development of HTML5, application developers see much promise in
the possibility of making those interfaces available in a the possibility of making those interfaces available in a
standardized way within the browser. standardized way within the browser.
This memo describes a set of building blocks that can be made This memo describes a set of building blocks that 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 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 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 RTCWEB protocol specification is to specify a set of The goal of the WebRTC protocol specification is to specify a set of
protocols that, if all are implemented, will allow an implementation protocols that, if all are implemented, will allow an implementation
to communicate with another implementation using audio, video and to communicate with another implementation using audio, video and
data sent along the most direct possible path between the data sent along the most direct possible path between the
participants. participants.
This document is intended to serve as the roadmap to the RTCWEB 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 RTCWEB context, and gives pointers to need further elaboration in the WebRTC context, and gives pointers to
other documents that form part of the RTCWEB 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 RTCWEB
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
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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, three classes of things that can For the purpose of this document, five types of entities are defined:
claim conformance are defined:
o A WebRTC browser is something that conforms to both the protocol o A WebRTC User Agent (also called a WebRTC UA or a WebRTC browser)
specification and the Javascript API defined above. 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 o A WebRTC device 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.
o A WebRTC gateway is something that mediates media traffic to non- o A WebRTC endpoint is either a WebRTC User Agent or a WebRTC
WebRTC entities. It is like a device, but has certain device.
restrictiions on where it can operate, which means that some of
the requirements can be relaxed.
All WebRTC browsers are WebRTC devices, so any requirement on a o A WebRTC-compatible endpoint is an endpoint that is capable of
successfully communicating with a WebRTC endpoint, but may fail to
meet some requirements of a WebRTC endpoint. This may limit where
in the network such an endpoint can be attached, or may limit the
security guarantees that it offers to others.
o A WebRTC gateway is a WebRTC-compatible endpoint that mediates
traffic to non-WebRTC entities.
All WebRTC browsers (UAs) are WebRTC devices, so any requirement on a
WebRTC device also applies to a WebRTC browser. WebRTC device also applies to a WebRTC browser.
WebRTC gateways are described in a separate document, 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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The alternative - that of having no mandatory to implement - does not The alternative - that of 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" - that "and then some" usually being called
a profile of some sort; in the version most antithetical to the a 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 following terms are used in this document, and as far as possible The following terms are used across the documents specifying the
across the documents specifying the RTCWEB suite, in the specific WebRTC suite, in the specific meanings given here. Not all terms are
meanings given here. Not all terms are used in this document. Other used in this document. Other terms are used in their commonly used
terms are used in their commonly used meaning. meaning.
The list is in alphabetical order. The list is in alphabetical order.
Agent: Undefined term. See "SDP Agent" and "ICE Agent". Agent: Undefined term. See "SDP Agent" and "ICE Agent".
API: Application Programming Interface - a specification of a set of API: Application Programming Interface - a specification of a set of
calls and events, usually tied to a programming language or an calls and events, usually tied to a programming language or an
abstract formal specification such as WebIDL, with its defined abstract formal specification such as WebIDL, with its defined
semantics. semantics.
Browser: Used synonymously with "Interactive User Agent" as defined Browser: Used synonymously with "Interactive User Agent" as defined
in the HTML specification [W3C.WD-html5-20110525]. in the HTML specification [W3C.WD-html5-20110525]. See also
"WebRTC User Agent".
ICE Agent: An implementation of the Interactive Connectivty ICE Agent: An implementation of the Interactive Connectivty
Establishment (ICE) [RFC5245] protocol. An ICE Agent may also be Establishment (ICE) [RFC5245] protocol. An ICE Agent may also be
an SDP Agent, but there exist ICE Agents that do not use SDP (for an SDP Agent, but there exist ICE Agents that do not use SDP (for
instance those that use Jingle). instance those that use Jingle).
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/
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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 Browser: Browser that conforms to the WebRTC protocol WebRTC User Agent: An entity that conforms to the WebRTC protocol
specifications and offer the WebRTC Javascript APIs. specifications and offer the WebRTC Javascript APIs. Also called
a WebRTC browser.
WebRTC Device: An unit (software, hardware or combinations) that WebRTC Device: An unit (software, hardware or combinations) that
conforms to the WebRTC protocol specifications, but does not offer conforms to the WebRTC protocol specifications, but does not offer
the WebRTC Javascript APIs. 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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| | | | | | | |
| 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 RTCWEB 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 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 [RFC3261] or XMPP [RFC6120]) could be used between (for example 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.
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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
server. server.
The choice of protocols, and definition of the translation between The choice of protocols, and definition of the translation between
them, is outside the scope of the RTCWEB standards suite described in them, is outside the scope of the WebRTC protocol suite described in
the document. the 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: 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.
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WebRTC devices MUST implement the transport protocols 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 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 devices MUST implement these
two specifications. two specifications.
WebRTC devices MUST implement [I-D.ietf-rtcweb-rtp-usage], WebRTC devices MUST implement [I-D.ietf-rtcweb-rtp-usage],
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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 devices MUST implement the codecs and profiles required in
[I-D.ietf-rtcweb-audio] [I-D.ietf-rtcweb-audio]
NOTE IN DRAFT: At this time (June 2014) there is no consensus on what NOTE IN DRAFT: At this time (October 2014) there is no consensus on
to say about video codecs in this section. 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 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 that are used in SIP [RFC3264],
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 RTCWEB 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 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 WebRTC, and their implications for
the API offered by a browser implementing RTCWEB, are described in the API offered by a WebRTC endpoint, 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 devices 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
skipping to change at page 15, line 20 skipping to change at page 15, line 20
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 WebRTC devices 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 WebRTC devices 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 browsers MUST implement the whole document.) this means that WebRTC devices 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
skipping to change at page 16, line 38 skipping to change at page 16, line 38
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-05 (work in Requirements", draft-ietf-rtcweb-audio-06 (work in
progress), February 2014. progress), September 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-11 (work in Channels", draft-ietf-rtcweb-data-channel-12 (work in
progress), July 2014. progress), September 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-07 (work in progress), July 2014. protocol-08 (work in progress), September 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-16 (work in progress), July draft-ietf-rtcweb-rtp-usage-17 (work in progress), August
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.
skipping to change at page 17, line 47 skipping to change at page 18, line 7
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-streams- World Wide Web Consortium WD WD-mediacapture-
20120628, June 2012, <http://www.w3.org/TR/2012/ streams-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-
20120209, February 2012, webrtc-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 reviews. Made a number of text clarifications, in response to document
reviews.
A.16. Changes from -11 to -12
Refined entity definitions to define "WebRTC endpoint" and "WebRTC-
compatible endpoint".
Changed remaining usage of the term "RTCWEB" to "WebRTC", including
in the page header.
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. 38 change blocks. 
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