draft-ietf-rtcweb-overview-04.txt   draft-ietf-rtcweb-overview-05.txt 
Network Working Group H. Alvestrand Network Working Group H. Alvestrand
Internet-Draft Google Internet-Draft Google
Intended status: Standards Track June 20, 2012 Intended status: Standards Track December 14, 2012
Expires: December 22, 2012 Expires: June 17, 2013
Overview: Real Time Protocols for Brower-based Applications Overview: Real Time Protocols for Brower-based Applications
draft-ietf-rtcweb-overview-04 draft-ietf-rtcweb-overview-05
Abstract Abstract
This document gives an overview and context of a protocol suite This document gives an overview and context of a protocol suite
intended for use with real-time applications that can be deployed in intended for use with real-time applications that can be deployed in
browsers - "real time communication on the Web". browsers - "real time communication on the Web".
It intends to serve as a starting and coordination point to make sure It intends to serve as a starting and coordination point to make sure
all the parts that are needed to achieve this goal are findable, and all the parts that are needed to achieve this goal are findable, and
that the parts that belong in the Internet protocol suite are fully that the parts that belong in the Internet protocol suite are fully
skipping to change at page 1, line 45 skipping to change at page 1, line 45
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 December 22, 2012. This Internet-Draft will expire on June 17, 2013.
Copyright Notice Copyright Notice
Copyright (c) 2012 IETF Trust and the persons identified as the Copyright (c) 2012 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
skipping to change at page 3, line 6 skipping to change at page 3, line 6
draft-alvestrand-dispatch-rtcweb-datagram-00 to -01 . . . 20 draft-alvestrand-dispatch-rtcweb-datagram-00 to -01 . . . 20
B.2. Changes from draft-alvestrand-dispatch-01 to B.2. Changes from draft-alvestrand-dispatch-01 to
draft-alvestrand-rtcweb-overview-00 . . . . . . . . . . . 20 draft-alvestrand-rtcweb-overview-00 . . . . . . . . . . . 20
B.3. Changes from draft-alvestrand-rtcweb-00 to -01 . . . . . . 20 B.3. Changes from draft-alvestrand-rtcweb-00 to -01 . . . . . . 20
B.4. Changes from draft-alvestrand-rtcweb-overview-01 to B.4. Changes from draft-alvestrand-rtcweb-overview-01 to
draft-ietf-rtcweb-overview-00 . . . . . . . . . . . . . . 21 draft-ietf-rtcweb-overview-00 . . . . . . . . . . . . . . 21
B.5. Changes from -00 to -01 of draft-ietf-rtcweb-overview . . 21 B.5. Changes from -00 to -01 of draft-ietf-rtcweb-overview . . 21
B.6. Changes from -01 to -02 of draft-ietf-rtcweb-overview . . 21 B.6. Changes from -01 to -02 of draft-ietf-rtcweb-overview . . 21
B.7. Changes from -02 to -03 of draft-ietf-rtcweb-overview . . 21 B.7. Changes from -02 to -03 of draft-ietf-rtcweb-overview . . 21
B.8. Changes from -03 to -04 of draft-ietf-rtcweb-overview . . 22 B.8. Changes from -03 to -04 of draft-ietf-rtcweb-overview . . 22
B.9. Changes from -04 to -05 of draft-ietf-rtcweb-overview . . 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 videoconferencing. conversations (aka "Internet telephony") and video conferencing.
The first attempts to build this were dependent on special networks, The first attempts to build this were dependent on special networks,
special hardware and custom-built software, often at very high prices special hardware and custom-built software, often at very high prices
or at low quality, placing great demands on the infrastructure. or at low quality, placing great demands on the infrastructure.
As the available bandwidth has increased, and as processors and other As the available bandwidth has increased, and as processors and other
hardware has become ever faster, the barriers to participation have hardware has become ever faster, the barriers to participation have
decreased, and it has become possible to deliver a satisfactory decreased, and it has become possible to deliver a satisfactory
experience on commonly available computing hardware. experience on commonly available computing hardware.
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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.
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 signalling 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. 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 envisage that the browser will contain all the functions that
need to be performed in order to have a function such as a telephone need to be performed in order to have a function such as a telephone
or a videoconferencing unit; the vision is that the browser will have or a video conferencing unit; the vision is that the browser will
the functions that are needed for a Web application, working in have the 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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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 device 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 |
| | Signalling | | | | Signaling | |
| |-------------| | | |-------------| |
| Server | path | Server | | Server | path | Server |
| | | | | | | |
+-----------+ +-----------+ +-----------+ +-----------+
/ \ / \
/ \ Application-defined / \ Application-defined
/ \ over / \ over
/ \ HTTP/Websockets / \ HTTP/Websockets
/ Application-defined over \ / Application-defined over \
/ HTTP/Websockets \ / HTTP/Websockets \
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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 RTCWEB protocol suite; the
signalling 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 signalling 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 or XMPP) could be used between servers, while either
a standards-based or proprietary protocol could be used between the a standards-based or proprietary protocol could be used between the
browser and the web server. 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 couild 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 RTCWEB standards 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
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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.
T are described in <WORKING GROUP DRAFT "TRANSPORTS">. T are described in <WORKING GROUP DRAFT "TRANSPORTS">.
ICE is required for all media paths that use UDP; in addition to the ICE is required for all media paths that use UDP; in addition to the
ability to pass NAT boxes, ICE fulfils the need for guaranteeing that ability to pass NAT boxes, ICE fulfills the need for guaranteeing
the media path is going to an UDP port that is willing to receive the that the media path is going to a UDP port that is willing to receive
data. the data.
The data transport protocols used by RTCWEB, as well as the details The data transport protocols used by RTCWEB, as well as the details
of interactions with intermediate boxes, such as firewalls, relays of interactions with intermediate boxes, such as firewalls, relays
and NAT boxes, are intended to be described in a separate document; and NAT boxes, are intended to be described in a separate document;
for now, notes are gathered in Appendix A. for now, notes are gathered in Appendix A.
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.
skipping to change at page 13, line 47 skipping to change at page 13, line 47
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
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 signalling 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 RTCWEB 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 would should specified in the MMUSIC WG, no other standardization should be
be required for it to be possible to use that in the web required for it to be possible to use that in the web browsers.
browsers. Adding new codecs which might have new SDP parameters Adding new codecs which might have new SDP parameters should not
should not change the APIs between the browser and javascript change the APIs between the browser and Javascript application.
application. As soon as the browsers support the new codecs, old As soon as the browsers support the new codecs, old applications
applications written before the codecs were specified should written before the codecs were specified should automatically be
automatically be able to use the new codecs where appropriate able to use the new codecs where appropriate with no changes to
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] [I-D.ietf-rtcweb-jsep]
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
skipping to change at page 14, line 43 skipping to change at page 14, line 43
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 being worked on as part of the W3C API effort, and
will be part of the peer connection API [W3C.WD-webrtc-20120209], and will be part of the peer connection API [W3C.WD-webrtc-20120209], and
the device control API [getusermedia]. Considerations for the the device control API [getusermedia]. Considerations for the
implications of wanting to identify correspondents are described in implications of wanting to identify correspondents are described in
[I-D.rescorla-rtcweb-generic-idp] (not a WG item). [I-D.rescorla-rtcweb-generic-idp] (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 influences 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.
Local functions include echo cancellation, volume control, camera Local functions include echo cancellation, volume control, camera
management including focus, zoom, pan/tilt controls (if available), management including focus, zoom, 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 to figure out who's controlling the camera, the local participant figure out who's controlling the camera, and
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 <whatever dB metrics makes sense here - most important voice into <whatever dB metrics makes sense here - most important
that we have one only> that we have one only>
The requirements on RTCWEB systems in this category are found in The requirements on RTCWEB systems in this category are found in
<WORKING GROUP DRAFT "MEDIA PROCESSING">; the proposed API for <WORKING GROUP DRAFT "MEDIA PROCESSING">; the proposed API for
control of local devices are found in [getusermedia]. control of local devices are found in [getusermedia].
10. IANA Considerations 10. IANA Considerations
skipping to change at page 16, line 28 skipping to change at page 16, line 28
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 Eric Rescorla, Justin Uberti, Henry Sinnreich, Colin
Perkins and Simon Leinen for document review. Perkins and Simon Leinen for document review, ad to Heath Matlock for
grammatical review.
13. References 13. References
13.1. Normative References 13.1. Normative References
[I-D.ietf-mmusic-sctp-sdp] [I-D.ietf-mmusic-sctp-sdp]
Loreto, S. and G. Camarillo, "Stream Control Transmission Loreto, S. and G. Camarillo, "Stream Control Transmission
Protocol (SCTP)-Based Media Transport in the Session Protocol (SCTP)-Based Media Transport in the Session
Description Protocol (SDP)", draft-ietf-mmusic-sctp-sdp-00 Description Protocol (SDP)", draft-ietf-mmusic-sctp-sdp-01
(work in progress), July 2011. (work in progress), March 2012.
[I-D.ietf-rtcweb-data-channel] [I-D.ietf-rtcweb-data-channel]
Jesup, R., Loreto, S., and M. Tuexen, "RTCWeb Datagram Jesup, R., Loreto, S., and M. Tuexen, "RTCWeb Datagram
Connection", draft-ietf-rtcweb-data-channel-00 (work in Connection", draft-ietf-rtcweb-data-channel-00 (work in
progress), March 2012. progress), March 2012.
[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-00 (work Establishment Protocol", draft-ietf-rtcweb-jsep-01 (work
in progress), March 2012. in progress), June 2012.
[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-01 (work in progress), draft-ietf-rtcweb-rtp-usage-04 (work in progress),
October 2011. July 2012.
[I-D.ietf-rtcweb-security] [I-D.ietf-rtcweb-security]
Rescorla, E., "Security Considerations for RTC-Web", Rescorla, E., "Security Considerations for RTC-Web",
draft-ietf-rtcweb-security-01 (work in progress), draft-ietf-rtcweb-security-03 (work in progress),
October 2011. June 2012.
[I-D.ietf-rtcweb-security-arch] [I-D.ietf-rtcweb-security-arch]
Rescorla, E., "RTCWEB Security Architecture", Rescorla, E., "RTCWEB Security Architecture",
draft-ietf-rtcweb-security-arch-00 (work in progress), draft-ietf-rtcweb-security-arch-03 (work in progress),
January 2012. July 2012.
[I-D.nandakumar-rtcweb-stun-uri] [I-D.nandakumar-rtcweb-stun-uri]
Nandakumar, S., Salgueiro, G., and P. Jones, "URI Scheme Nandakumar, S., Salgueiro, G., Jones, P., and M. Petit-
for Session Traversal Utilities for NAT (STUN) Protocol", Huguenin, "URI Scheme for Session Traversal Utilities for
draft-nandakumar-rtcweb-stun-uri-00 (work in progress), NAT (STUN) Protocol", draft-nandakumar-rtcweb-stun-uri-01
October 2011. (work in progress), March 2012.
[I-D.tuexen-tsvwg-sctp-dtls-encaps] [I-D.tuexen-tsvwg-sctp-dtls-encaps]
Jesup, R., Loreto, S., Stewart, R., and M. Tuexen, "DTLS Jesup, R., Loreto, S., Stewart, R., and M. Tuexen, "DTLS
Encapsulation of SCTP Packets for RTCWEB", Encapsulation of SCTP Packets for RTCWEB",
draft-tuexen-tsvwg-sctp-dtls-encaps-00 (work in progress), draft-tuexen-tsvwg-sctp-dtls-encaps-01 (work in progress),
March 2012. July 2012.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997. Requirement Levels", BCP 14, RFC 2119, March 1997.
[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 2002. June 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
skipping to change at page 18, line 10 skipping to change at page 18, line 12
Traversal for Offer/Answer Protocols", RFC 5245, Traversal for Offer/Answer Protocols", RFC 5245,
April 2010. April 2010.
[RFC5766] Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using [RFC5766] Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using
Relays around NAT (TURN): Relay Extensions to Session Relays around NAT (TURN): Relay Extensions to Session
Traversal Utilities for NAT (STUN)", RFC 5766, April 2010. Traversal Utilities for NAT (STUN)", RFC 5766, April 2010.
13.2. Informative References 13.2. Informative References
[I-D.cbran-rtcweb-codec] [I-D.cbran-rtcweb-codec]
Bran, C. and C. Jennings, "WebRTC Codec and Media Bran, C., Jennings, C., and J. Valin, "WebRTC Codec and
Processing Requirements", draft-cbran-rtcweb-codec-01 Media Processing Requirements",
(work in progress), October 2011. draft-cbran-rtcweb-codec-02 (work in progress),
March 2012.
[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", Time Communication Use-cases and Requirements",
draft-ietf-rtcweb-use-cases-and-requirements-06 (work in draft-ietf-rtcweb-use-cases-and-requirements-09 (work in
progress), October 2011. progress), June 2012.
[I-D.jesup-rtcweb-data-protocol] [I-D.jesup-rtcweb-data-protocol]
Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel
Protocol", draft-jesup-rtcweb-data-protocol-00 (work in Protocol", draft-jesup-rtcweb-data-protocol-02 (work in
progress), March 2012. progress), July 2012.
[I-D.rescorla-rtcweb-generic-idp] [I-D.rescorla-rtcweb-generic-idp]
Rescorla, E., "RTCWEB Generic Identity Provider Rescorla, E., "RTCWEB Generic Identity Provider
Interface", draft-rescorla-rtcweb-generic-idp-00 (work in Interface", draft-rescorla-rtcweb-generic-idp-01 (work in
progress), January 2012. progress), March 2012.
[RFC3935] Alvestrand, H., "A Mission Statement for the IETF", [RFC3935] Alvestrand, H., "A Mission Statement for the IETF",
BCP 95, RFC 3935, October 2004. BCP 95, RFC 3935, October 2004.
[W3C.WD-html5-20110525] [W3C.WD-html5-20110525]
Hickson, I., "HTML5", World Wide Web Consortium Hickson, I., "HTML5", World Wide Web Consortium
LastCall WD-html5-20110525, May 2011, LastCall WD-html5-20110525, May 2011,
<http://www.w3.org/TR/2011/WD-html5-20110525>. <http://www.w3.org/TR/2011/WD-html5-20110525>.
[W3C.WD-webrtc-20120209] [W3C.WD-webrtc-20120209]
skipping to change at page 20, line 49 skipping to change at page 20, line 49
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
B.3. Changes from draft-alvestrand-rtcweb-00 to -01 B.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 acknowledgements section. Added content to acknowledgments section.
Added change log. Added change log.
Spell-checked document. Spell-checked document.
B.4. Changes from draft-alvestrand-rtcweb-overview-01 to B.4. Changes from draft-alvestrand-rtcweb-overview-01 to
draft-ietf-rtcweb-overview-00 draft-ietf-rtcweb-overview-00
Changed draft name and document date. Changed draft name and document date.
skipping to change at page 22, line 14 skipping to change at page 22, line 14
B.8. Changes from -03 to -04 of draft-ietf-rtcweb-overview B.8. Changes from -03 to -04 of draft-ietf-rtcweb-overview
Made introduction more normative. Made introduction more normative.
Several wording changes in response to review comments from EKR Several wording changes in response to review comments from EKR
Added Appendix A to hold references and notes that are not yet in a Added Appendix A to hold references and notes that are not yet in a
separate document. separate document.
B.9. Changes from -04 to -05 of draft-ietf-rtcweb-overview
Minor grammatical fixes. This is mainly a "keepalive" refresh.
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. 30 change blocks. 
54 lines changed or deleted 61 lines changed or added

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