Network Working Group                                      H. Alvestrand
Internet-Draft                                                    Google
Intended status: Standards Track                            June 5, 2011
Expires: December 7, 2011

      Overview: Real Time Protocols for Brower-based Applications


   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",
   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
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   Drafts is at

   Internet-Drafts are draft documents valid for a maximum of six months
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   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 December 7, 2011.

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
   ( in effect on the date of
   publication of this document.  Please review these documents
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   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 overview . . . . . . . . . . . . . . . . . .  5
     2.2.  Relationship between API and protocol  . . . . . . . . . .  5
     2.3.  On interoperability and innovation . . . . . . . . . . . .  6
     2.4.  Terminology  . . . . . . . . . . . . . . . . . . . . . . .  6
   3.  Functionality groups . . . . . . . . . . . . . . . . . . . . .  7
   4.  Data transport . . . . . . . . . . . . . . . . . . . . . . . .  8
   5.  Data framing and securing  . . . . . . . . . . . . . . . . . .  9
   6.  Data formats . . . . . . . . . . . . . . . . . . . . . . . . . 10
   7.  Connection management  . . . . . . . . . . . . . . . . . . . . 10
   8.  Presentation and control . . . . . . . . . . . . . . . . . . . 11
   9.  Local system support functions . . . . . . . . . . . . . . . . 11
   10. IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 12
   11. Security Considerations  . . . . . . . . . . . . . . . . . . . 12
   12. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 13
   13. References . . . . . . . . . . . . . . . . . . . . . . . . . . 13
     13.1. Normative References . . . . . . . . . . . . . . . . . . . 13
     13.2. Informative References . . . . . . . . . . . . . . . . . . 14
   Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 14

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1.  Introduction

   The Internet was, from very early in its lifetime, considered a
   possible veichle 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 are, 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

   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 possiblitiy of
   making those interfaces available in a standardized way within the

   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.

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2.  Principles and Terminology

2.1.  Goals of this overview

   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
   auxillary data sent along the most direct possible path between the

   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

   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 auxillary 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
   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

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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.

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.  Other terms are used in their commonly used

   The list is in alphabetical order.

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   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

   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

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

   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).

   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.  Functionality groups

   The functionallity groups that are needed 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, bandwith
      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.

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   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.

4.  Data transport

   Datagram transport is the subject of a separate draft, "A Datagram
   Transport for the RTC-Web
   profile".[I-D.alvestrand-dispatch-rtcweb-datagram] The basic approach
   is to use ICE as a setup mechanism, and to specify mechanisms to use
   ICE over connections that utilize UDP and TCP if needed to support a
   basic datagram-passing function with adequate security.  In order to
   deal with complex NAT/firewall situations, relaying using TURN MUST

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   be supported.

   For octet-stream transport, TCP is used.  (QUESTION: Do we need a TCP
   relay specification?  The use of TURN over TCP and TLS is specified
   in the TURN RFC - is it suitable?)

   (The role of Web Sockets [I-D.ietf-hybi-thewebsocketprotocol] needs
   to be clarified.)

   The data transport MUST behave reasonably in the presence of
   congested networks; this is usually interpreted as reducing the send
   rate when congestion is encountered.  TCP, when correctly
   implemented, does this automatically; this is not the case with UDP,
   and the RTP framing specification does not contain a congestion
   control component.

   Determining an useful congestion handling mechanism is a high
   priority for work with this specification suite.

   Usually when designing data transport for media, one separates out
   the functions of bandwith estimation (which is a determinant for
   which codec and which codec parameters to use) and congestion
   management (reacting to events that change the available bandwidth,
   such as congestion or media change, in an appropriate manner).  The
   totality of these features MUST ensure that an implementation of the
   RTCWEB suite is able to coexist on a network with other users,
   including TCP-based data transfers, without starving them of
   resources, and without letting itself be starved.

5.  Data framing and securing

   RTP [RFC3550]and SRTP [RFC3711].  The RTP/SAVP profile, defined as
   part of SRTP, is supported, and "extended RTCP", RTP/SAVPF [RFC4585],
   with its secured version RTP/SAVPF [RFC5124]is used in order to
   support codec functionality that depends on this RTP profile, such as

   The implementation of SRTP used MUST support encryption using AES-CM
   with MIC, on both RTP and RTCP channels. <TODO: Add pointer to
   appropriate profile here> (Note that like for all mandatory-to-
   implement, there is no requirement that these protocols be used, just
   that it is possible to negotiate them.)

   [OPEN ISSUE; We need to specify a securable format of passing data
   that is not RTP.  One proposal has been to use DTLS over DCCCP,
   although specifying a "data codec" and using SRTP has been proposed

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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.

   NOTE IN DRAFT: The particular codecs named are NOT A DECISION.  They
   are included to illustrate possible choices, and to check with the
   group that the references given are necessary and sufficient for the
   purpose of specifying an interoperable codec suite.

   In audio, the OPUS codec[I-D.ietf-codec-opus] MUST be supported.  For
   ease of interoperability with gateways to older equipment, G.711
   U-law, audio/PCMU, defined in RFC 1890 [RFC1890] section 4.4.12, is
   also mandatory to implement.  There is no third mandatory to

   In video, the VP8 codec [I-D.westin-payload-vp8] MUST be supported.

   The Theora codec is also freely available.  H.264/AVC and H.264/SVC
   [I-D.ietf-avt-rtp-svc] are widely enough used that it gives a wider
   range of communications partners if they are supported.

   The overall set of data formats and parameters, and the identifiers
   that allow the partners to bind data streams to application-level
   entities, form a session description.  It is vital that the
   communicating parties have the same session description, and that the
   session description can be updated while the connection is in

7.  Connection management

   This specification is silent on the definition of connection
   management protocols.  It envisions that implementors will make a
   choice on whether to implement connection management protocols as a
   downloadable component, as a browser plug-in, or as a frontend/
   backend split, where a part of the protocol machinery is downloaded
   into the browser and uses some mechanism (for instance WebSockets) to
   communicate back to a backend implementing the rest of the connection
   management protocol.

   XMPP, and its Jingle component, has proved a versatile tool in

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   building interoperable communities, and so has SIP.  This suite
   requires that the browser support establishing and describing
   connections using a data format for session description capable of
   representing the information needed by these two protocols, such as
   one that can be one-to-one transformed into SDP.  The exact
   specification of this API is done elsewhere <insert reference when
   available>; this API is powerful enough that all interesting
   parameters of the transport mechanisms specified above are settable,
   and clear enough that how to connect the API to the protocols is

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.

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

   o  Echo cancellation should be good enough that feedback (defined as
      a rising volume of sound with no local sound input) does not

   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,

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      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>

10.  IANA Considerations

   This document makes no request of IANA.

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

11.  Security Considerations

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

   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 positivie
      identification is appropriate), or that their identity cannot be
      uncovered (when anonymity is a goal of the application).

   This specification addresses some, but not all, of these concerns,
   and makes some assumptions about the security considerations of other
   parts of the environment; it is up to the implementor to see that
   these security assumptions are warranted.  In particular:

   o  We assume that the ICE security mechanism is a necessary and
      sufficient criterion for accepting that a connection attempt is
      from a communications partner.  This means that we trust the
      randomness of ICE "usernames" and the security of ICE "passwords".

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   o  We assume that the SRTP key exchange mechanisms and security
      profiles specified provide an adequate level of protection for
      audio and video media.

   (there needs to be more text here)

12.  Acknowledgements

13.  References

13.1.  Normative References

              Alvestrand, H., "A Datagram Transport for the RTC-Web
              profile", draft-alvestrand-dispatch-rtcweb-datagram-01
              (work in progress), February 2011.

              Valin, J. and K. Vos, "Definition of the Opus Audio
              Codec", draft-ietf-codec-opus-05 (work in progress),
              March 2011.

              Fette, I., "The WebSocket protocol",
              draft-ietf-hybi-thewebsocketprotocol-07 (work in
              progress), April 2011.

              Westin, P. and H. Lundin, "Proposal for the IETF on "RTP
              Payload Format for VP8 Video"",
              draft-westin-payload-vp8-02 (work in progress),
              March 2011.

   [RFC1890]  Schulzrinne, H., "RTP Profile for Audio and Video
              Conferences with Minimal Control", RFC 1890, January 1996.

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

   [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.

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   [RFC4585]  Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
              "Extended RTP Profile for Real-time Transport Control
              Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
              July 2006.

   [RFC5124]  Ott, J. and E. Carrara, "Extended Secure RTP Profile for
              Real-time Transport Control Protocol (RTCP)-Based Feedback
              (RTP/SAVPF)", RFC 5124, February 2008.

13.2.  Informative References

              Wenger, S., Wang, Y., Schierl, T., and A. Eleftheriadis,
              "RTP Payload Format for Scalable Video Coding",
              draft-ietf-avt-rtp-svc-27 (work in progress),
              February 2011.

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

Author's Address

   Harald T. Alvestrand
   Kungsbron 2
   Stockholm,   11122


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