draft-ietf-rtcweb-overview-16.txt   draft-ietf-rtcweb-overview-17.txt 
Network Working Group H. Alvestrand Network Working Group H. Alvestrand
Internet-Draft Google Internet-Draft Google
Intended status: Standards Track November 14, 2016 Intended status: Standards Track February 17, 2017
Expires: May 18, 2017 Expires: August 21, 2017
Overview: Real Time Protocols for Browser-based Applications Overview: Real Time Protocols for Browser-based Applications
draft-ietf-rtcweb-overview-16 draft-ietf-rtcweb-overview-17
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 May 18, 2017. This Internet-Draft will expire on August 21, 2017.
Copyright Notice Copyright Notice
Copyright (c) 2016 IETF Trust and the persons identified as the Copyright (c) 2017 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
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
carefully, as they describe your rights and restrictions with respect carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as the Trust Legal Provisions and are provided without warranty as
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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 . . . . . . . . . . . . . . . . . 22 A.15. Changes from -10 to -11 . . . . . . . . . . . . . . . . . 22
A.16. Changes from -11 to -12 . . . . . . . . . . . . . . . . . 22 A.16. Changes from -11 to -12 . . . . . . . . . . . . . . . . . 22
A.17. Changes from -12 to -13 . . . . . . . . . . . . . . . . . 22 A.17. Changes from -12 to -13 . . . . . . . . . . . . . . . . . 22
A.18. Changes from -13 to -14 . . . . . . . . . . . . . . . . . 22 A.18. Changes from -13 to -14 . . . . . . . . . . . . . . . . . 22
A.19. Changes from -14 to -15 . . . . . . . . . . . . . . . . . 22 A.19. Changes from -14 to -15 . . . . . . . . . . . . . . . . . 22
A.20. Changes from -15 to -16 . . . . . . . . . . . . . . . . . 22 A.20. Changes from -15 to -16 . . . . . . . . . . . . . . . . . 22
A.21. Changes from -16 to -17 . . . . . . . . . . . . . . . . . 23
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 23 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 23
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,
skipping to change at page 4, line 35 skipping to change at page 4, line 35
2.1. Goals of this document 2.1. Goals of this document
The goal of the WebRTC protocol specification is to specify a set of The goal of the WebRTC protocol specification is to specify a set of
protocols that, if all are implemented, will allow an implementation protocols that, if all are implemented, will allow an implementation
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 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 parts of the WebRTC
specification, lists references to other specifications that don't protocol specifications, lists references to other specifications
need further elaboration in the WebRTC context, and gives pointers to that don't need further elaboration in the WebRTC context, and gives
other documents that form part of the WebRTC suite. pointers to other documents that form part of the WebRTC suite.
By reading this document and the documents it refers to, it should be By reading this document and the documents it refers to, it should be
possible to have all information needed to implement an WebRTC possible to have all information needed to implement 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 major parts, each consisting
of multiple documents:
o A protocol specification, done in the IETF o A protocol specification, done in the IETF
o A Javascript API specification, defined in a series of W3C
o A Javascript API specification, done in the W3C documents
[W3C.WD-webrtc-20120209][W3C.WD-mediacapture-streams-20120628] [W3C.WD-webrtc-20120209][W3C.WD-mediacapture-streams-20120628]
Together, these two specifications aim to provide an environment Together, these two specifications aim to provide an environment
where Javascript embedded in any page, viewed in any compatible where Javascript embedded in any page, when suitably authorized by
browser, when suitably authorized by its user, is able to set up its user, is able to set up communication using audio, video and
communication using audio, video and auxiliary data, where the auxiliary data, as long as the browser supports this specification.
browser environment does not constrain the types of application in The browser environment does not constrain the types of application
which this functionality can be used. in which this functionality can be used.
The protocol specification does not assume that all implementations The protocol specification does not assume that all implementations
implement this API; it is not intended to be necessary for implement this API; it is not intended to be necessary for
interoperation to know whether the entity one is communicating with interoperation to know whether the entity one is communicating with
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
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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]. See also in the HTML specification [W3C.WD-html5-20110525]. See also
"WebRTC User Agent". "WebRTC User Agent".
Data channel: An abstraction that allows data to be sent between
WebRTC endpoints in the form of messages. Two endpoints can have
multiple data channels between them.
ICE Agent: An implementation of the Interactive Connectivty ICE Agent: An implementation of the Interactive Connectivty
Establishment (ICE) [RFC5245] protocol. An ICE Agent may also be Establishment (ICE) [I-D.ietf-ice-rfc5245bis] protocol. An ICE
an SDP Agent, but there exist ICE Agents that do not use SDP (for Agent may also be an SDP Agent, but there exist ICE Agents that do
instance those that use Jingle). not use SDP (for 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/
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.
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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.
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 and data paths.
Signaling Path: The communication channels used between entities Signaling Path: The communication channels used between entities
participating in signaling to transfer signaling. There may be participating in signaling to transfer signaling. There may be
more entities in the signaling path than in the media path. more entities in the signaling path than in the media path.
NOTE: Where common definitions exist for these terms, those NOTE: Where common definitions exist for these terms, those
definitions should be used to the greatest extent possible. definitions should be used to the greatest extent possible.
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
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 use to talk to each other, without any
servers, and the APIs that are offered for a Javascript application intervening servers, and the APIs that are offered for a Javascript
to take advantage of the browser's functionality. application to take advantage of the browser's functionality.
+------------------------+ On-the-wire +------------------------+ On-the-wire
| | Protocols | | Protocols
| Servers |---------> | Servers |--------->
| | | |
| | | |
+------------------------+ +------------------------+
^ ^
| |
| |
| HTTP/ | HTTP/
| Websockets | WebSockets
| |
| |
+----------------------------+ +----------------------------+
| Javascript/HTML/CSS | | Javascript/HTML/CSS |
+----------------------------+ +----------------------------+
Other ^ ^RTC Other ^ ^RTC
APIs | |APIs APIs | |APIs
+---|-----------------|------+ +---|-----------------|------+
| | | | | | | |
| +---------+| | +---------+|
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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 endpoint that implements the protocols they are fully specified, any endpoint that implements the protocols
faithfully should be able to interoperate with the application faithfully should be able to interoperate with the application
running in the browser. running in the browser.
A commonly imagined model of deployment is the one depicted below. A commonly imagined model of deployment is the one depicted below.
+-----------+ +-----------+ +-----------+ +-----------+
| Web | | Web | | Web | | Web |
| | Signaling | | | | 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 \
/ \ / \
+-----------+ +-----------+ +-----------+ +-----------+
|JS/HTML/CSS| |JS/HTML/CSS| |JS/HTML/CSS| |JS/HTML/CSS|
+-----------+ +-----------+ +-----------+ +-----------+
+-----------+ +-----------+ +-----------+ +-----------+
| | | | | | | |
| | | | | | | |
| Browser | ------------------------- | Browser | | Browser | ------------------------- | Browser |
| | Media path | | | | Media path | |
| | | | | | | |
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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.
For example, if both operators' servers implement SIP, SIP could be For example, if both operators' servers implement SIP, SIP could be
used for communication between servers, along with either a used for communication between servers, along with either a
standardized signaling mechanism (e.g. SIP over Websockets) or a standardized signaling mechanism (e.g. SIP over WebSockets) or a
proprietary signaling mechanism used between the application running proprietary signaling mechanism used between the application running
in the browser and the web server. Similarly, if both operators' in the browser and the web server. Similarly, if both operators'
servers implement XMPP, XMPP could be used for communication between servers implement XMPP, XMPP could be used for communication between
XMPP servers, with either a standardized signaling mechanism (e.g. XMPP servers, with either a standardized signaling mechanism (e.g.
XMPP over Websockets or BOSH) or a proprietary signaling mechanism XMPP over WebSockets or BOSH) or a proprietary signaling mechanism
used between the application running in the browser and the web used between the application running in the browser and the web
server. server.
The choice of protocols, and definition of the translation between The choice of protocols for client-server and inter-server
them, is outside the scope of the WebRTC protocol suite described in signalling, and definition of the translation between them, is
the document. outside the scope of the WebRTC protocol suite described in 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.
o Data framing: RTP and other data formats that serve as containers, o Data framing: RTP, SCTP and other data formats that serve as
and their functions for data confidentiality and integrity. containers, and their functions for data confidentiality and
integrity.
o Data formats: Codec specifications, format specifications and o Data formats: Codec specifications, format specifications and
functionality specifications for the data passed between systems. functionality specifications for the data passed between systems.
Audio and video codecs, as well as formats for data and document Audio and video codecs, as well as formats for data and document
sharing, belong in this category. In order to make use of data sharing, belong in this category. In order to make use of data
formats, a way to describe them, a session description, is needed. formats, a way to describe them, a session description, is needed.
o Connection management: Setting up connections, agreeing on data o Connection management: Setting up connections, agreeing on data
formats, changing data formats during the duration of a call; SIP formats, changing data formats during the duration of a call; SIP
and Jingle/XMPP belong in this category. and Jingle/XMPP belong in this category.
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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 systems 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 endpoints 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 in
this means that WebRTC endpoints MUST implement the whole document.) Section 6, 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
skipping to change at page 16, line 26 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 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. Magnus Westerlund, Olle E. Johansson and Simon Leinen for document
review.
13. References 13. References
13.1. Normative References 13.1. Normative References
[I-D.ietf-ice-rfc5245bis]
Keranen, A., Holmberg, C., and J. Rosenberg, "Interactive
Connectivity Establishment (ICE): A Protocol for Network
Address Translator (NAT) Traversal", draft-ietf-ice-
rfc5245bis-08 (work in progress), December 2016.
[I-D.ietf-rtcweb-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-11 (work in Requirements", draft-ietf-rtcweb-audio-11 (work in
progress), April 2016. progress), April 2016.
[I-D.ietf-rtcweb-data-channel] [I-D.ietf-rtcweb-data-channel]
Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data
Channels", draft-ietf-rtcweb-data-channel-13 (work in Channels", draft-ietf-rtcweb-data-channel-13 (work in
progress), January 2015. progress), January 2015.
[I-D.ietf-rtcweb-data-protocol] [I-D.ietf-rtcweb-data-protocol]
Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel
Establishment Protocol", draft-ietf-rtcweb-data- Establishment Protocol", draft-ietf-rtcweb-data-
protocol-09 (work in progress), January 2015. protocol-09 (work in progress), January 2015.
[I-D.ietf-rtcweb-jsep] [I-D.ietf-rtcweb-jsep]
Uberti, J., Jennings, C., and E. Rescorla, "Javascript Uberti, J., Jennings, C., and E. Rescorla, "Javascript
Session Establishment Protocol", draft-ietf-rtcweb-jsep-17 Session Establishment Protocol", draft-ietf-rtcweb-jsep-18
(work in progress), October 2016. (work in progress), January 2017.
[I-D.ietf-rtcweb-rtp-usage] [I-D.ietf-rtcweb-rtp-usage]
Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time
Communication (WebRTC): Media Transport and Use of RTP", Communication (WebRTC): Media Transport and Use of RTP",
draft-ietf-rtcweb-rtp-usage-26 (work in progress), March draft-ietf-rtcweb-rtp-usage-26 (work in progress), March
2016. 2016.
[I-D.ietf-rtcweb-security] [I-D.ietf-rtcweb-security]
Rescorla, E., "Security Considerations for WebRTC", draft- Rescorla, E., "Security Considerations for WebRTC", draft-
ietf-rtcweb-security-08 (work in progress), February 2015. ietf-rtcweb-security-08 (work in progress), February 2015.
skipping to change at page 17, line 43 skipping to change at page 18, line 5
[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, DOI 10.17487/RFC3550, Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
July 2003, <http://www.rfc-editor.org/info/rfc3550>. July 2003, <http://www.rfc-editor.org/info/rfc3550>.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)", Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, DOI 10.17487/RFC3711, March 2004, RFC 3711, DOI 10.17487/RFC3711, March 2004,
<http://www.rfc-editor.org/info/rfc3711>. <http://www.rfc-editor.org/info/rfc3711>.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols", RFC 5245,
DOI 10.17487/RFC5245, April 2010,
<http://www.rfc-editor.org/info/rfc5245>.
[W3C.WD-mediacapture-streams-20120628] [W3C.WD-mediacapture-streams-20120628]
Burnett, D. and A. Narayanan, "Media Capture and Streams", Burnett, D. and A. Narayanan, "Media Capture and Streams",
World Wide Web Consortium WD WD-mediacapture-streams- World Wide Web Consortium WD WD-mediacapture-streams-
20120628, June 2012, <http://www.w3.org/TR/2012/ 20120628, June 2012, <http://www.w3.org/TR/2012/
WD-mediacapture-streams-20120628>. WD-mediacapture-streams-20120628>.
[W3C.WD-webrtc-20120209] [W3C.WD-webrtc-20120209]
Bergkvist, A., Burnett, D., Jennings, C., and A. Bergkvist, A., Burnett, D., Jennings, C., and A.
Narayanan, "WebRTC 1.0: Real-time Communication Between Narayanan, "WebRTC 1.0: Real-time Communication Between
Browsers", World Wide Web Consortium WD WD-webrtc- Browsers", World Wide Web Consortium WD WD-webrtc-
skipping to change at page 23, line 5 skipping to change at page 23, line 5
None. This is a "keepalive" update. None. This is a "keepalive" update.
A.19. Changes from -14 to -15 A.19. Changes from -14 to -15
Changed "gateways" reference to point to the WG document. Changed "gateways" reference to point to the WG document.
A.20. Changes from -15 to -16 A.20. Changes from -15 to -16
None. This is a "keepalive" publication. None. This is a "keepalive" publication.
A.21. Changes from -16 to -17
Addressed review comments by Olle E. Johansson and Magnus Westerlund
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. 27 change blocks. 
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