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Versions: (draft-alvestrand-rtcweb-overview) 00 01 02 03 04 05 06 07 08 09 10

Network Working Group                                      H. Alvestrand
Internet-Draft                                                    Google
Intended status: Standards Track                      September 28, 2011
Expires: March 31, 2012


      Overview: Real Time Protocols for Brower-based Applications
                     draft-ietf-rtcweb-overview-02

Abstract

   This document gives an overview and context of a protocol suite
   intended for use with real-time applications that can be deployed in
   browsers - "real time communication on the Web".

   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
   that the parts that belong in the Internet protocol suite are fully
   specified and on the right publication track.

   This work is an attempt to synthesize the input of many people, but
   makes no claims to fully represent the views of any of them.  All
   parts of the document should be regarded as open for discussion,
   unless the RTCWEB chairs have declared consensus on an item.

   This document is a work item of the RTCWEB working group.

Requirements Language

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in RFC 2119 [RFC2119].

Status of this Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."




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   This Internet-Draft will expire on March 31, 2012.

Copyright Notice

   Copyright (c) 2011 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.



































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Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  4
   2.  Principles and Terminology . . . . . . . . . . . . . . . . . .  5
     2.1.  Goals of this document . . . . . . . . . . . . . . . . . .  5
     2.2.  Relationship between API and protocol  . . . . . . . . . .  5
     2.3.  On interoperability and innovation . . . . . . . . . . . .  6
     2.4.  Terminology  . . . . . . . . . . . . . . . . . . . . . . .  7
   3.  Architecture and Functionality groups  . . . . . . . . . . . .  8
   4.  Data transport . . . . . . . . . . . . . . . . . . . . . . . . 12
   5.  Data framing and securing  . . . . . . . . . . . . . . . . . . 12
   6.  Data formats . . . . . . . . . . . . . . . . . . . . . . . . . 12
   7.  Connection management  . . . . . . . . . . . . . . . . . . . . 13
   8.  Presentation and control . . . . . . . . . . . . . . . . . . . 13
   9.  Local system support functions . . . . . . . . . . . . . . . . 14
   10. IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 14
   11. Security Considerations  . . . . . . . . . . . . . . . . . . . 14
   12. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 15
   13. References . . . . . . . . . . . . . . . . . . . . . . . . . . 15
     13.1. Normative References . . . . . . . . . . . . . . . . . . . 15
     13.2. Informative References . . . . . . . . . . . . . . . . . . 16
   Appendix A.  Change log  . . . . . . . . . . . . . . . . . . . . . 16
     A.1.  Changes from
           draft-alvestrand-dispatch-rtcweb-datagram-00 to -01  . . . 17
     A.2.  Changes from draft-alvestrand-dispatch-01 to
           draft-alvestrand-rtcweb-overview-00  . . . . . . . . . . . 17
     A.3.  Changes from draft-alvestrand-rtcweb-00 to -01 . . . . . . 17
     A.4.  Changes from draft-alvestrand-rtcweb-overview-01 to
           draft-ietf-rtcweb-overview-00  . . . . . . . . . . . . . . 17
     A.5.  Changes from draft-ietf-rtcweb-overview -00 to -01 . . . . 18
     A.6.  Changes from draft-ietf-rtcweb-overview -01 to -02 . . . . 18
   Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 18



















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1.  Introduction

   The Internet was, from very early in its lifetime, considered a
   possible vehicle for the deployment of real-time, interactive
   applications - with the most easily imaginable being audio
   conversations (aka "Internet telephony") and videoconferencing.

   The first attempts to build this were dependent on special networks,
   special hardware and custom-built software, often at very high prices
   or at low quality, placing great demands on the infrastructure.

   As the available bandwidth has increased, and as processors and other
   hardware has become ever faster, the barriers to participation have
   decreased, and it is possible to deliver a satisfactory experience on
   commonly available computing hardware.

   Still, there are a number of barriers to the ability to communicate
   universally - one of these is that there is, as of yet, no single set
   of communication protocols that all agree should be made available
   for communication; another is the sheer lack of universal
   identification systems (such as is served by telephone numbers or
   email addresses in other communications systems).

   Development of The Universal Solution has proved hard, however, for
   all the usual reasons.  This memo aims to take a more building-block-
   oriented approach, and try to find consensus on a set of substrate
   components that we think will be useful in any real-time
   communications systems.

   The last few years have also seen a new platform rise for deployment
   of services: The browser-embedded application, or "Web application".
   It turns out that as long as the browser platform has the necessary
   interfaces, it is possible to deliver almost any kind of service on
   it.

   Traditionally, these interfaces have been delivered by plugins, which
   had to be downloaded and installed separately from the browser; in
   the development of HTML5, much promise is seen by the possibility of
   making those interfaces available in a standardized way within the
   browser.

   This memo specifies a set of building blocks that can be made
   accessible and controllable through a Javascript API interface in a
   browser, and which together form a necessary and sufficient set of
   functions to allow the use of interactive audio and video in
   applications that communicate directly between browsers across the
   Internet.  The resulting protocol suite is intended to enable all the
   applications that are described as required scenarios in the RTCWEB



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   use cases document [I-D.ietf-rtcweb-use-cases-and-requirements].

   Other efforts, for instance the W3C WebRTC, Web Applications and
   Device API working groups, focus on making standardized APIs and
   interfaces available, within or alongside the HTML5 effort, for those
   functions; this memo concentrates on specifying the protocols and
   subprotocols that are needed to specify the interactions that happen
   across the network.


2.  Principles and Terminology

2.1.  Goals of this document

   The goal of the RTCWEB protocol specification is to specify a set of
   protocols that, if all are implemented, will allow the implementation
   to communicate with another implementation using audio, video and
   auxiliary data sent along the most direct possible path between the
   participants.

   This document is intended to serve as the roadmap to the RTCWEB
   specifications.  It defines terms used by other pieces of
   specification, lists references to other specifications that don't
   need further elaboration in the RTCWEB context, and gives pointers to
   other documents that form part of the RTCWEB suite.

   By reading this document and the documents it refers to, it should be
   possible to have all information needed to implement an RTCWEB
   compatible implementation.

2.2.  Relationship between API and protocol

   The total RTCWEB/WEBRTC effort consists of two pieces:

   o  A protocol specification, done in the IETF

   o  A Javascript API specification, done in the W3C [webrtc-api]

   Together, these two specifications aim to provide an environment
   where Javascript embedded in any page, viewed in any compatible
   browser, when suitably authorized by its user, is able to set up
   communication using audio, video and auxiliary data, where the
   browser environment does not constrain the types of application in
   which this functionality can be used.

   The protocol specification does not assume that all implementations
   implement this API; it is not intended to be possible by observing
   the bits on the wire whether they come from a browser or from another



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   device implementing this specification.

   The goal of cooperation between the protocol specification and the
   API specification is that for all options and features of the
   protocol specification, it should be clear which API calls to make to
   exercise that option or feature; similarly, for any sequence of API
   calls, it should be clear which protocol options and features will be
   invoked.  Both subject to constraints of the implementation, of
   course.

2.3.  On interoperability and innovation

   The "Mission statement of the IETF" [RFC3935] states that "The
   benefit of a standard to the Internet is in interoperability - that
   multiple products implementing a standard are able to work together
   in order to deliver valuable functions to the Internet's users."

   Communication on the Internet frequently occurs in two phases:

   o  Two parties communicate, through some mechanism, what
      functionality they both are able to support

   o  They use that shared communicative functionality to communicate,
      or, failing to find anything in common, give up on communication.

   There are often many choices that can be made for communicative
   functionality; the history of the Internet is rife with the proposal,
   standardization, implementation, and success or failure of many types
   of options, in all sorts of protocols.

   The goal of having a mandatory to implement function set is to
   prevent negotiation failure, not to preempt or prevent negotiation.

   The presence of a mandatory to implement function set serves as a
   strong changer of the marketplace of deployment - in that it gives a
   guarantee that, as long as you conform to a specification, and the
   other party is willing to accept communication at the base level of
   that specification, you can communicate successfully.

   The alternative - that of having no mandatory to implement - does not
   mean that you cannot communicate, it merely means that in order to be
   part of the communications partnership, you have to implement the
   standard "and then some" - that "and then some" usually being called
   a profile of some sort; in the version most antithetical to the
   Internet ethos, that "and then some" consists of having to use a
   specific vendor's product only.





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2.4.  Terminology

   The following terms are used in this document, and as far as possible
   across the documents specifying the RTCWEB suite, in the specific
   meanings given here.  Not all terms are used in this document.  Other
   terms are used in their commonly used meaning.

   The list is in alphabetical order.

   Agent:  Undefined term.  See "SDP Agent" and "ICE Agent".

   API:  Application Programming Interface - a specification of a set of
      calls and events, usually tied to a programming language or an
      abstract formal specification such as WebIDL, with its defined
      semantics.

   ICE Agent:  An implementation of the ICE [RFC5245] protocol.  An ICE
      Agent may also be an SDP Agent, but there exist ICE Agents that do
      not use SDP (for instance those that use Jingle).

   Interactive:  Communication between multiple parties, where the
      expectation is that an action from one party can cause a reaction
      by another party, and the reaction can be observed by the first
      party, with the total time required for the action/reaction/
      observation is on the order of no more than hundreds of
      milliseconds.

   Media:  Audio and video content.  Not to be confused with
      "transmission media" such as wires.

   Media path:  The path that media data follows from one browser to
      another.

   Protocol:  A specification of a set of data units, their
      representation, and rules for their transmission, with their
      defined semantics.  A protocol is usually thought of as going
      between systems.

   Real-time media:  Media where generation of content and display of
      content are intended to occur closely together in time (on the
      order of no more than hundreds of milliseconds).

   SDP Agent:  The protocol implementation involved in the SDP offer/
      answer exchange, as defined in [RFC3264] section 3.







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   Signaling:  Communication that happens in order to establish, manage
      and control media paths.

   Signaling Path:  The communication channels used between entities
      participating in signalling to transfer signaling.  There may be
      more entities in the signaling path than in the media path.

   NOTE: Where common definitions exist for these terms, those
   definitions should be used to the greatest extent possible.

   TODO: Extend this list with other terms that might prove slippery.


3.  Architecture and Functionality groups

   The model of real-time support for browser-based applications does
   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
   or a videoconferencing unit; the vision is that the browser will have
   the functions that are needed for a Web application, working in
   conjunction with its backend servers, to implement these functions.

   This means that two vital interfaces need specification: The
   protocols that browsers talk to each other, without any intervening
   servers, and the APIs that are offered for a Javascript application
   to take advantage of the browser's functionality.

























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                        +------------------------+  On-the-wire
                        |                        |  Protocols
                        |      Servers           |--------->
                        |                        |
                        |                        |
                        +------------------------+
                                    ^
                                    |
                                    |
                                    | HTTP/
                                    | Websockets
                                    |
                                    |
                      +----------------------------+
                      |    Javascript/HTML/CSS     |
                      +----------------------------+
                   Other  ^                 ^RTC
                   APIs   |                 |APIs
                      +---|-----------------|------+
                      |   |                 |      |
                      |                 +---------+|
                      |                 | Browser ||  On-the-wire
                      | Browser         | RTC     ||  Protocols
                      |                 | Function|----------->
                      |                 |         ||
                      |                 |         ||
                      |                 +---------+|
                      +---------------------|------+
                                            |
                                            V
                                       Native OS Services








                          Figure 1: Browser Model

   As for all protocol and API specifications, there is no restriction
   that the protocols can only be used to talk to another browser; since
   they are fully specified, any device that implements the protocols
   faithfully should be able to interoperate with the application
   running in the browser.

   A commonly imagined model of deployment is the one depicted below.



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                +-----------+             +-----------+
                |   Web     |             |   Web     |
                |           |  Signalling |           |
                |           |-------------|           |
                |  Server   |   path      |  Server   |
                |           |             |           |
                +-----------+             +-----------+
                     /                           \
                    /                             \   Proprietary over
                   /                               \  HTTP/Websockets
                  /                                 \
                 /  Proprietary over                 \
                /   HTTP/Websockets                   \
               /                                       \
         +-----------+                           +-----------+
         |JS/HTML/CSS|                           |JS/HTML/CSS|
         +-----------+                           +-----------+
         +-----------+                           +-----------+
         |           |                           |           |
         |           |                           |           |
         |  Browser  | ------------------------- |  Browser  |
         |           |          Media path       |           |
         |           |                           |           |
         +-----------+                           +-----------+

                      Figure 2: Browser RTC Trapezoid

   If the two Web servers are operated by different entities, the
   signalling path needs to be agreed upon, either by standardization or
   by other means of agreement; for example, both servers might
   implement SIP, and the servers would talk SIP to each other, and each
   would translate between the SIP protocol and their proprietary
   representation for sending to their application running in the
   browser.  This part is outside the scope of the RTCWEB standars
   suite.

   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
   conformant to the specifications of the RTCWEB protocol suite; the
   signalling path ("high path") goes via servers that can modify,
   translate or massage the signals as needed.

   The functionality groups that are needed in the browser can be
   specified, more or less from the bottom up, as:

   o  Data transport: TCP, UDP and the means to securely set up
      connections between entities, as well as the functions for
      deciding when to send data: Congestion management, bandwidth



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      estimation and so on.

   o  Data framing: RTP and other data formats that serve as containers,
      and their functions for data confidentiality and integrity.

   o  Data formats: Codec specifications, format specifications and
      functionality specifications for the data passed between systems.
      Audio and video codecs, as well as formats for data and document
      sharing, belong in this category.  In order to make use of data
      formats, a way to describe them, a session description, is needed.

   o  Connection management: Setting up connections, agreeing on data
      formats, changing data formats during the duration of a call; SIP
      and Jingle/XMPP belong in this category.

   o  Presentation and control: What needs to happen in order to ensure
      that interactions behave in a non-surprising manner.  This can
      include floor control, screen layout, voice activated image
      switching and other such functions - where part of the system
      require the cooperation between parties.  Cisco/Tandberg's TIP was
      one attempt at specifying this functionality.

   o  Local system support functions: These are things that need not be
      specified uniformly, because each participant may choose to do
      these in a way of the participant's choosing, without affecting
      the bits on the wire in a way that others have to be cognizant of.
      Examples in this category include echo cancellation (some forms of
      it), local authentication and authorization mechanisms, OS access
      control and the ability to do local recording of conversations.

   Within each functionality group, it is important to preserve both
   freedom to innovate and the ability for global communication.
   Freedom to innovate is helped by doing the specification in terms of
   interfaces, not implementation; any implementation able to
   communicate according to the interfaces is a valid implementation.
   Ability to communicate globally is helped both by having core
   specifications be unencumbered by IPR issues and by having the
   formats and protocols be fully enough specified to allow for
   independent implementation.

   One can think of the three first groups as forming a "media transport
   infrastructure", and of the three last groups as forming a "media
   service".  In many contexts, it makes sense to use a common
   specification for the media transport infrastructure, which can be
   embedded in browsers and accessed using standard interfaces, and "let
   a thousand flowers bloom" in the "media service" layer; to achieve
   interoperable services, however, at least the first five of the six
   groups need to be specified.



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4.  Data transport

   Data transport refers to the sending and receiving of data over the
   network interfaces, the choice of network-layer addresses at each end
   of the communication, and the interaction with any intermediate
   entities that handle the data, but do not modify it (such as TURN
   relays).

   It includes necessary functions for congestion control: When not to
   send data.

   The data transport protocols used by RTCWEB are described in <WORKING
   GROUP DRAFT "TRANSPORTS">.

   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
   the media path is going to an UDP port that is willing to receive the
   data.

   The details of interactions with intermediate boxes, such as
   firewalls, relays and NAT boxes, is described in <WORKING GROUP DRAFT
   "PEER TO PEER CONNECTIVITY">.


5.  Data framing and securing

   The format for media transport is RTP [RFC3550].  Implementation of
   SRTP [RFC3711] is required for all implementations.

   The detailed considerations for usage of functions from RTP and SRTP
   are given in [I-D.ietf-rtcweb-rtp-usage].  Key negotiation for SRTP
   is described in <MISSING>.  Transfer of data that is not in RTP
   format is described in <MISSING>.


6.  Data formats

   The intent of this specification is to allow each communications
   event to use the data formats that are best suited for that
   particular instance, where a format is supported by both sides of the
   connection.  However, a minimum standard is greatly helpful in order
   to ensure that communication can be achieved.  This document
   specifies a minimum baseline that will be supported by all
   implementations of this specification, and leaves further codecs to
   be included at the will of the implementor.

   The mandatory to implement codecs, as well as any profiling
   requirements for both mandatory and optional codecs, is described in



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   <WORKING GROUP DRAFT "MEDIA PROCESSING">.


7.  Connection management

   The methods, mechanisms and requirements for setting up, negotiating
   and tearing down connections is a large subject, and one where it is
   desirable to have both interoperability and freedom to innovate.

   The following principles apply:

   1.  The media negotiations will be done using the same SDP offer/
       answer semantics that are used in SIP [RFC3264], in such a way
       that it is possible to build a signalling gateway between SIP and
       the RTCWEB media negotiation.

   2.  It will be possible to gateway between legacy SIP devices that
       support ICE and appropriate RTP / SDP mechanisms and codecs
       without using a media gateway.  A signaling gateway to convert
       between the signaling on the web side to the SIP signaling may be
       needed.

   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
       be required for it to be possible to use that in the web
       browsers.  Adding new codecs which might have new SDP parameters
       should not change the APIs between the browser and javascript
       application.  As soon as the browsers support the new codecs, old
       applications written before the codecs were specified should
       automatically be able to use the new codecs where appropriate
       with no changes to the JS applications.

   The particular choices made for RTCWEB are described in <WORKING
   GROUP DRAFT "SIGNALING AND NEGOTIATION">.


8.  Presentation and control

   The most important part of control is the user's control over the
   browser's interaction with input/output devices and communications
   channels.  It is important that the user have some way of figuring
   out where his audio, video or texting is being sent, for what
   purported reason, and what guarantees are made by the parties that
   form part of this control channel.  This is largely a local function
   between the browser, the underlying operating system and the user
   interface; this is being worked on as part of the W3C API effort, and
   will be part of [webrtc-api]




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9.  Local system support functions

   These are characterized by the fact that the quality of these
   functions strongly influences the user experience, but the exact
   algorithm does not need coordination.  In some cases (for instance
   echo cancellation, as described below), the overall system definition
   may need to specify that the overall system needs to have some
   characteristics for which these facilities are useful, without
   requiring them to be implemented a certain way.

   Local functions include echo cancellation, volume control, camera
   management including focus, zoom, pan/tilt controls (if available),
   and more.

   Certain parts of the system SHOULD conform to certain properties, for
   instance:

   o  Echo cancellation should be good enough to achieve the suppression
      of acoustical feedback loops below a perceptually noticeable
      level.

   o  Privacy concerns must be satisfied; for instance, if remote
      control of camera is offered, the APIs should be available to let
      the local participant to figure out who's controlling the camera,
      and possibly decide to revoke the permission for camera usage.

   o  Automatic gain control, if present, should normalize a speaking
      voice into <whatever dB metrics makes sense here - most important
      that we have one only>

   The requirements on RTCWEB systems in this category are found in
   <WORKING GROUP DRAFT "MEDIA PROCESSING">.


10.  IANA Considerations

   This document makes no request of IANA.

   Note to RFC Editor: this section may be removed on publication as an
   RFC.


11.  Security Considerations

   Security of the web-enabled real time communications comes in several
   pieces:





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   o  Security of the components: The browsers, and other servers
      involved.  The most target-rich environment here is probably the
      browser; the aim here should be that the introduction of these
      components introduces no additional vulnerability.

   o  Security of the communication channels: It should be easy for a
      participant to reassure himself of the security of his
      communication - by verifying the crypto parameters of the links he
      himself participates in, and to get reassurances from the other
      parties to the communication that they promise that appropriate
      measures are taken.

   o  Security of the partners' identity: verifying that the
      participants are who they say they are (when positive
      identification is appropriate), or that their identity cannot be
      uncovered (when anonymity is a goal of the application).

   The security analysis, and the requirements derived from that
   analysis, is contained in [I-D.ietf-rtcweb-security].


12.  Acknowledgements

   The number of people who have taken part in the discussions
   surrounding this draft are too numerous to list, or even to identify.
   The ones below have made special, identifiable contributions; this
   does not mean that others' contributions are less important.

   Thanks to Cary Bran, Cullen Jennings, Colin Perkins, Magnus
   Westerlund and Joerg Ott, who offered technical contributions on
   various versions of the draft.

   Thanks to Jonathan Rosenberg, Matthew Kaufman and others at Skype for
   the ASCII drawings in section 1.

   Thanks to Justin Uberti, Henry Sinnreich, Colin Perkins and Simon
   Leinen for document review.


13.  References

13.1.  Normative References

   [I-D.ietf-rtcweb-rtp-usage]
              Perkins, C., Westerlund, M., and J. Ott, "RTP Requirements
              for RTC-Web", draft-ietf-rtcweb-rtp-usage-00 (work in
              progress), September 2011.




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   [I-D.ietf-rtcweb-security]
              Rescorla, E., "Security Considerations for RTC-Web",
              draft-ietf-rtcweb-security-00 (work in progress),
              September 2011.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [RFC3264]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
              with Session Description Protocol (SDP)", RFC 3264,
              June 2002.

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, July 2003.

   [RFC3711]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
              Norrman, "The Secure Real-time Transport Protocol (SRTP)",
              RFC 3711, March 2004.

13.2.  Informative References

   [I-D.ietf-rtcweb-use-cases-and-requirements]
              Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
              Time Communication Use-cases and Requirements",
              draft-ietf-rtcweb-use-cases-and-requirements-05 (work in
              progress), September 2011.

   [RFC3935]  Alvestrand, H., "A Mission Statement for the IETF",
              BCP 95, RFC 3935, October 2004.

   [RFC5245]  Rosenberg, J., "Interactive Connectivity Establishment
              (ICE): A Protocol for Network Address Translator (NAT)
              Traversal for Offer/Answer Protocols", RFC 5245,
              April 2010.

   [webrtc-api]
              Bergkvist, Burnett, Jennings, Narayanan, "WebRTC 1.0:
              Real-time Communication Between Browsers", August 2011.

              Available at
              http://dev.w3.org/2011/webrtc/editor/webrtc.html


Appendix A.  Change log

   This section may be deleted by the RFC Editor when preparing for
   publication.



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Internet-Draft               RTCWEB Overview              September 2011


A.1.  Changes from draft-alvestrand-dispatch-rtcweb-datagram-00 to -01

   Added section "On interoperability and innovation"

   Added data confidentiality and integrity to the "data framing" layer

   Added congestion management requirements in the "data transport"
   layer section

   Changed need for non-media data from "question: do we need this?" to
   "Open issue: How do we do this?"

   Strengthened disclaimer that listed codecs are placeholders, not
   decisions.

   More details on why the "local system support functions" section is
   there.

A.2.  Changes from draft-alvestrand-dispatch-01 to
      draft-alvestrand-rtcweb-overview-00

   Added section on "Relationship between API and protocol"

   Added terminology section

   Mentioned congestion management as part of the "data transport" layer
   in the layer list

A.3.  Changes from draft-alvestrand-rtcweb-00 to -01

   Removed most technical content, and replaced with pointers to drafts
   as requested and identified by the RTCWEB WG chairs.

   Added content to acknowledgements section.

   Added change log.

   Spell-checked document.

A.4.  Changes from draft-alvestrand-rtcweb-overview-01 to
      draft-ietf-rtcweb-overview-00

   Changed draft name and document date.

   Removed unused references






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Internet-Draft               RTCWEB Overview              September 2011


A.5.  Changes from draft-ietf-rtcweb-overview -00 to -01

   Added architecture figures to section 2.

   Changed the description of "echo cancellation" under "local system
   support functions".

   Added a few more definitions.

A.6.  Changes from draft-ietf-rtcweb-overview -01 to -02

   Added pointers to use cases, security and rtp-usage drafts (now WG
   drafts).

   Changed description of SRTP from mandatory-to-use to mandatory-to-
   implement.

   Added the "3 principles of negotiation" to the connection management
   section.

   Added an explicit statement that ICE is required for both NAT and
   consent-to-receive.


Author's Address

   Harald T. Alvestrand
   Google
   Kungsbron 2
   Stockholm,   11122
   Sweden

   Email: harald@alvestrand.no


















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