[Docs] [txt|pdf|xml|html] [Tracker] [WG] [Email] [Diff1] [Diff2] [Nits]

Versions: (draft-holmberg-rtcweb-ucreqs) 00 01 02 03 04 05 06 07 08 09 10 11 12 13 14 15 16 RFC 7478

RTCWEB Working Group                                         C. Holmberg
Internet-Draft                                              S. Hakansson
Intended status: Informational                               G. Eriksson
Expires: March 4, 2012                                          Ericsson
                                                       September 1, 2011


         Web Real-Time Communication Use-cases and Requirements
          draft-ietf-rtcweb-use-cases-and-requirements-04.txt

Abstract

   This document describes web based real-time communication use-cases.
   Based on the use-cases, the document also derives requirements
   related to the browser, and the API used by web applications to
   request and control media stream services provided by the browser.

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."

   This Internet-Draft will expire on March 4, 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.



Holmberg, et al.          Expires March 4, 2012                 [Page 1]


Internet-Draft                   RTC-Web                  September 2011


Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Conventions  . . . . . . . . . . . . . . . . . . . . . . . . .  3
   3.  Definitions  . . . . . . . . . . . . . . . . . . . . . . . . .  3
   4.  Use-cases  . . . . . . . . . . . . . . . . . . . . . . . . . .  3
     4.1.  Introduction . . . . . . . . . . . . . . . . . . . . . . .  3
     4.2.  Browser-to-browser use-cases . . . . . . . . . . . . . . .  3
       4.2.1.  Simple Video Communication Service . . . . . . . . . .  3
       4.2.2.  Simple Video Communication Service, NAT/FW that
               blocks UDP . . . . . . . . . . . . . . . . . . . . . .  4
       4.2.3.  Simple Video Communication Service, access change  . .  4
       4.2.4.  Simple Video Communication Service, QoS  . . . . . . .  5
       4.2.5.  Simple video communication service with
               inter-operator calling . . . . . . . . . . . . . . . .  5
       4.2.6.  Hockey Game Viewer . . . . . . . . . . . . . . . . . .  6
       4.2.7.  Multiparty video communication . . . . . . . . . . . .  7
       4.2.8.  Multiparty on-line game with voice communication . . .  7
       4.2.9.  Distributed Music Band . . . . . . . . . . . . . . . .  8
     4.3.  Browser - GW/Server use cases  . . . . . . . . . . . . . .  9
       4.3.1.  Telephony terminal . . . . . . . . . . . . . . . . . .  9
       4.3.2.  Fedex Call . . . . . . . . . . . . . . . . . . . . . .  9
       4.3.3.  Video conferencing system with central server  . . . .  9
   5.  Requirements . . . . . . . . . . . . . . . . . . . . . . . . . 10
     5.1.  General  . . . . . . . . . . . . . . . . . . . . . . . . . 11
     5.2.  Browser requirements . . . . . . . . . . . . . . . . . . . 11
     5.3.  API requirements . . . . . . . . . . . . . . . . . . . . . 13
   6.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 15
   7.  Security Considerations  . . . . . . . . . . . . . . . . . . . 15
     7.1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . 15
     7.2.  Browser Considerations . . . . . . . . . . . . . . . . . . 15
     7.3.  Web Application Considerations . . . . . . . . . . . . . . 15
   8.  Additional use-cases . . . . . . . . . . . . . . . . . . . . . 16
   9.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 16
   10. Change Log . . . . . . . . . . . . . . . . . . . . . . . . . . 17
   11. References . . . . . . . . . . . . . . . . . . . . . . . . . . 18
     11.1. Normative References . . . . . . . . . . . . . . . . . . . 18
     11.2. Informative References . . . . . . . . . . . . . . . . . . 18
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 18












Holmberg, et al.          Expires March 4, 2012                 [Page 2]


Internet-Draft                   RTC-Web                  September 2011


1.  Introduction

   This document presents a few use-cases of web applications that are
   executed in a browser and use real-time communication capabilities.
   Based on the use-cases, the document derives requirements related to
   the browser and the API used by web applications in the browser.

   The requirements related to the browser are named "Fn" and are
   described in Section 5.2

   The requirements related to the API are named "An" and are described
   in Section 5.3

   The document focuses on requirements related to real-time media
   streams.  Requirements related to privacy, signalling between the
   browser and web server etc. are currently not considered.


2.  Conventions

   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 BCP 14, RFC 2119
   [RFC2119].


3.  Definitions

   TBD


4.  Use-cases

4.1.  Introduction

   This section describes web based real-time communication use-cases,
   from which requirements are derived.

4.2.  Browser-to-browser use-cases

4.2.1.  Simple Video Communication Service

4.2.1.1.  Description

   In the service the users have loaded, and logged into, a video
   communication web application into their browsers, provided by the
   same service provider.  The web service publishes information about
   user login status, by pushing updates to the web application in the



Holmberg, et al.          Expires March 4, 2012                 [Page 3]


Internet-Draft                   RTC-Web                  September 2011


   browsers.  By selecting an online peer user, a 1-1 video
   communication session between the browsers of the peers is initiated.
   The invited user might accept or reject the session.

   During session establishment a self-view is displayed, and once the
   session has been established the video sent from the remote peer is
   displayed displayed in addition to the self-view.  The users can
   during the session select to remove, and re-insert the self-view.
   The users can change the sizes of the video displays during the
   session.  The users can also pause sending of media (audio, video, or
   both), and mute incoming media.

   It is essential that the communication can not be eavesdropped.

   Any session participant can end the session at any time.

   The users are using communication devices of different makes, with
   different operating systems and browsers from different vendors.

   One user has an unreliable Internet connection.  It sometimes has
   packet losses, and is sometimes goes down completely.

   One user is located behind a Network Address Translator (NAT).

4.2.1.2.  Derived Requirements

   F1, F2, F3, F4, F5, F6, F8, F9, F10, F17, F22, F25

   A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12, A13

4.2.2.  Simple Video Communication Service, NAT/FW that blocks UDP

4.2.2.1.  Description

   This use-case is almost identical to the previos one.  The difference
   is that one of the users is behind a NAT that blocks UDP traffic.

4.2.2.2.  Derived Requirements

   F1, F2, F3, F4, F5, F6, F8, F9, F10, F17, F22, F23, F25, F26

   A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12, A13

4.2.3.  Simple Video Communication Service, access change







Holmberg, et al.          Expires March 4, 2012                 [Page 4]


Internet-Draft                   RTC-Web                  September 2011


4.2.3.1.  Description

   This use-case is almost identical to "4.2.1 Simple Video
   Communication Service".  The difference is that the user changes
   network access during the session:

   The communication device used by one of the users have several
   network adapters (Ethernet, WiFi, Cellular).  The communication
   device is access the Internet using Ethernet, but the user has to
   start a trip during the session.  The communication device
   automatically changes to use WiFi when the Ethernet cable is removed
   and then moves to cellular access to the Internet when moving out of
   WiFi coverage.  The session continues even though the access method
   changes.

4.2.3.2.  Derived Requirements

   F1, F2, F3, F4, F5, F6, F8, F9, F10, F17, F22, F23, F25

   A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12, A13

4.2.4.  Simple Video Communication Service, QoS

4.2.4.1.  Description

   This use-case is almost identical to the previos one.  The use of QoS
   capabilities is added:

   The user in the previous use case that starts a trip is behind a
   common residential router that supports prioritization of traffic.
   In addition, the user's provider of cellular access has QoS support
   enabled.  The user is able to take advantage of the QoS support both
   when accessing via the residential router and when using cellular.

4.2.4.2.  Derived Requirements

   F1, F2, F3, F4, F5, F6, F8, F9, F10, F17, F21, F22, F23, F25

   A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12, A13

4.2.5.  Simple video communication service with inter-operator calling

4.2.5.1.  Description

   Two users have logged into two different web applications, provided
   by different service providers.

   The service providers are interconnected by some means, but exchange



Holmberg, et al.          Expires March 4, 2012                 [Page 5]


Internet-Draft                   RTC-Web                  September 2011


   no more information about the users than what can be carried using
   SIP.

   NOTE: More profiling of what this means may be needed.

   Each web service publishes information about user login status for
   users that have a relationship with the other user; how this is
   established is out of scope.

   The same functionality as in the "4.2.1 Simple Video Communication
   Service" is available.

   The same issues with connectivity apply.

4.2.5.2.  Derived requirements

   F1, F2, F3, F4, F5, F6, F8, F9, F10, F17, F22, F24, F25

   A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12, A13

4.2.6.  Hockey Game Viewer

4.2.6.1.  Description

   An ice-hockey club uses an application that enables talent scouts to,
   in real-time, show and discuss games and players with the club
   manager.  The talent scouts use a mobile phone with two cameras, one
   front facing and one rear facing.

   The club manager uses a desktop, equipped with one camera, for
   viewing the game and discussing with the talent scout.

   Before the game starts, and during game breaks, the talent scout and
   the manager have a 1-1 video communication.  Only the rear facing
   camera of the mobile phone is used.  On the display of the mobile
   phone, the video of the club manager is shown with a picture-in-
   picture thumbnail of the rear facing camera (self-view).  On the
   display of the desktop, the video of the talent scout is shown with a
   picture-in-picture thumbnail ot the desktop camera (self-view).

   When the game is on-going, the talent scout activates the use of the
   front facing camera, and that stream is sent to the desktop (the
   stream from the rear facing camera continues to be sent all the
   time).  The video stream captured by the front facing camera (that is
   capturing the game) of the mobile phone is shown in a big window on
   the desktop screen, with picture-in-picture thumbnails of the rear
   facing camera and the desktop camera (self-view).  On the display of
   the mobile phone the game is shown (front facing camera) with



Holmberg, et al.          Expires March 4, 2012                 [Page 6]


Internet-Draft                   RTC-Web                  September 2011


   picture-in-picture thumbnails of the rear facing camera (self-view)
   and the desktop camera.

   It is essential that the communication can not be eavesdropped.

4.2.6.2.  Derived Requirements

   F1, F2, F3, F4, F5, F6, F8, F9, F10, F14, F17

   A1, A2, A3, A4, A5, A7, A8, A9, A10, A11, A12, A13, A15

4.2.7.  Multiparty video communication

4.2.7.1.  Description

   In this use-case the simple video communication service is extended
   by allowing multiparty sessions.  No central server is involved - the
   browser of each participant sends and receives streams to and from
   all other session participants.  The web application in the browser
   of each user is responsible for setting up streams to all receivers.

   In order to enhance intelligibility, the web application pans the
   audio from different participants differently when rendering the
   audio.  This is done automatically, but users can change how the
   different participants are placed in the (virtual) room.

   Each video stream received is by default displayed in a thumbnail
   frame within the browser, but users can change the display size.

   It is essential that the communication can not be eavesdropped.

   Note: What this use-case adds in terms of requirements is
   capabilities to send streams to and receive streams from several
   peers concurrently, as well as the capabilities to render the video
   from all recevied streams and be able to spatialize and mix the audio
   from all received streams locally in the browser.

4.2.7.2.  Derived Requirements

   F1, F2, F3, F4, F5, F6, F8, F9, F10, F11, F12, F13, F14, F17, F22

   A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12, A13, A14, A15

4.2.8.  Multiparty on-line game with voice communication







Holmberg, et al.          Expires March 4, 2012                 [Page 7]


Internet-Draft                   RTC-Web                  September 2011


4.2.8.1.  Description

   In this use-case, the voice part of the multiparty video
   communication application is used in the context of an on-line game.
   The received voice audio media is rendered together with game sound
   objects.  For example, the sound of a tank moving from left to right
   over the screen must be rendered and played to the user together with
   the voice media.

   Quick updates of the game state is required.

   It is essential that the communication can not be eavesdropped.

   Note: the difference regarding local audio processing compared to the
   "Multiparty video communication" use-case is that other sound objects
   than the streams must be possible to be included in the
   spatialization and mixing.  "Other sound objects" could for example
   be a file with the sound of the tank, that file could be stored
   locally or remotely.

4.2.8.2.  Derived Requirements

   F1, F2, F3, F4, F5, F6, F8, F9, F11, F12, F13, F15, F17, F20

   A1, A2, A3, A4, A5, A7, A8, A9, A10, A11, A12, A13, A14, A15, A16

4.2.9.  Distributed Music Band

4.2.9.1.  Description

   In this use-case, a music band is playing music while the members are
   at different physical locations.  No central server is used, instead
   all streams are set up in a mesh fashion.

   Discussion: This use-case was briefly discussed at the Quebec webrtc
   meeting and it got support.  So far the only concrete requirement
   (A17) derived is that the application must be able to ask the browser
   to treat the audio signal as audio (in contrast to speech).  However,
   the use case should be further analysed to determine other
   requirements (could be e.g. on delay mic->speaker, level control of
   audio signals, etc.).

4.2.9.2.  Derived Requirements

   F1, F2, F3, F4, F5, F6, F8, F9, F11, F12, F13

   A1, A2, A3, A4, A5, A7, A8, A9, A10, A11, A12, A13, A14, A15, A17




Holmberg, et al.          Expires March 4, 2012                 [Page 8]


Internet-Draft                   RTC-Web                  September 2011


4.3.  Browser - GW/Server use cases

4.3.1.  Telephony terminal

4.3.1.1.  Description

   A mobile telephony operator allows its customers to use a web browser
   to access their services.  After a simple log in the user can place
   and receive calls in the same way as when using a normal mobile
   phone.  When a call is received or placed, the identity is shown in
   the same manner as when a mobile phone used.

   It is essential that the communication can not be eavesdropped.

4.3.1.2.  Derived Requirements

   F1, F2, F3, F4, F5, F6, F8, F9, F10, F17, F18

   A1, A2, A3, A4, A7, A8, A9, A10, A11, A12, A13

4.3.2.  Fedex Call

4.3.2.1.  Description

   Alice uses her web browser with a service something like Skype to be
   able to phone PSTN numbers.  Alice calls 1-800-gofedex.  Alice should
   be able to hear the initial prompts from the fedex IVR and when the
   IVR says press 1, there should be a way for Alice to navigate the
   IVR.

4.3.2.2.  Derived Requirements

   F1, F2, F3, F4, F5, F6, F8, F9, F10, F18, F19

   A1, A2, A3, A4, A7, A8, A9, A10, A11, A12, A13

4.3.3.  Video conferencing system with central server

4.3.3.1.  Description

   An organization uses a video communication system that supports the
   establishment of multiparty video sessions using a central conference
   server.

   The browsers of each participant send an audio stream (type in terms
   of mono, stereo, 5.1, ... depending on the equipment of the
   participant) to the central server.  The central server mixes the
   audio streams (and can in the mixing process naturally add effects



Holmberg, et al.          Expires March 4, 2012                 [Page 9]


Internet-Draft                   RTC-Web                  September 2011


   such as spatialization) and sends towards each participant a mixed
   audio stream which is played to the user.

   The browser of each participant sends video towards the server.  For
   each participant one high resolution video is displayed in a large
   window, while a number of low resolution videos are displayed in
   smaller windows.  The server selects what video streams to be
   forwarded as main- and thumbnail videos respectively, based on speech
   activity.  As the video streams to display can change quite
   frequently (as the conversation flows) it is important that the delay
   from when a video stream is selected for display until the video can
   be displayed is short.

   The organization has an internal network set up with an aggressive
   firewall handling access to the Internet.  If users can not
   physically access the internal network, they can establish a Virtual
   Private Network (VPN).

   It is essential that the communication can not be eavesdropped.

   All participant are authenticated by the central server, and
   authorized to connect to the central server.  The participants are
   identified to each other by the central server, and the participants
   do not have access to each others' credentials such as e-mail
   addresses or login IDs.

   Note: This use-case adds requirements on support for fast stream
   switches F7, on encryption of media and on ability to traverse very
   restrictive FWs.  There exists several solutions that enable the
   server to forward one high resolution and several low resolution
   video streams: a) each browser could send a high resolution, but
   scalable stream, and the server could send just the base layer for
   the low resolution streams, b) each browser could in a simulcast
   fashion send one high resolution and one low resolution stream, the
   server just selects, c) each browser sends just an high resolution
   stream, the server trancodes into low reslution streams as required.

4.3.3.2.  Derived Requirements

   F1, F2, F3, F4, F5, F6, F7, F8, F9, F10, F14, F16, F17

   A1, A2, A3, A4, A5, A7, A8, A9, A10, A11, A12, A13, A15


5.  Requirements






Holmberg, et al.          Expires March 4, 2012                [Page 10]


Internet-Draft                   RTC-Web                  September 2011


5.1.  General

   This section contains the requirements derived from the use-cases in
   section 4.

   NOTE: It is assumed that the user applications are executed on a
   browser.  Whether the capabilities to implement specific browser
   requirements are implemented by the browser application, or are
   provided to the browser application by the underlying operating
   system, is outside the scope of this document.

5.2.  Browser requirements


   REQ-ID          DESCRIPTION
   ---------------------------------------------------------------
   F1              The browser MUST be able to use microphones and
                   cameras as input devices to generate streams.
   ----------------------------------------------------------------
   F2              The browser MUST be able to send streams to a
                   peer in presence of NATs.
   ----------------------------------------------------------------
   F3              Transmitted streams MUST be rate controlled.
   ----------------------------------------------------------------
   F4              The browser MUST be able to receive, process and
                   render streams from peers.
   ----------------------------------------------------------------
   F5              The browser MUST be able to render good quality
                   audio and video even in presence of reasonable
                   levels of jitter and packet losses.

                   TBD: What is a reasonable level?
   ----------------------------------------------------------------
   F6              The browser MUST be able to handle high loss and
                   jitter levels in a graceful way.
   ----------------------------------------------------------------
   F7              The browser MUST support fast stream switches.
   ----------------------------------------------------------------
   F8              The browser MUST detect when a stream from a
                   peer is not received any more
   ----------------------------------------------------------------
   F9              When there are both incoming and outgoing audio
                   streams, echo cancellation MUST be made available to
                   avoid disturbing echo during conversation.

                   QUESTION: How much control should be left to the
                   web application?
   ----------------------------------------------------------------



Holmberg, et al.          Expires March 4, 2012                [Page 11]


Internet-Draft                   RTC-Web                  September 2011


   F10             The browser MUST support synchronization of
                   audio and video.


                   QUESTION: How much control should be left to the
                   web application?
   ----------------------------------------------------------------
   F11             The browser MUST be able to transmit streams to
                   several peers concurrently.
   ----------------------------------------------------------------
   F12             The browser MUST be able to receive streams from
                   multiple peers concurrently.
   ----------------------------------------------------------------
   F13             The browser MUST be able to pan, mix and render
                   several concurrent audio streams.
   ----------------------------------------------------------------
   F14             The browser MUST be able to render several
                   concurrent video streams
   ----------------------------------------------------------------
   F15             The browser MUST be able to process and mix
                   sound objects (media that is retrieved from another
                   source than the established media stream(s) with the
                   peer(s) with audio streams).
   ----------------------------------------------------------------
   F16             Streams MUST be able to pass through restrictive
                   firewalls.
   ----------------------------------------------------------------
   F17             It MUST be possible to protect streams from
                   eavesdropping.
   ----------------------------------------------------------------
   F18             The browser MUST support an audio media format
                   (codec) that is commonly supported by existing
                   telephony services.

                   QUESTION: G.711?
   ----------------------------------------------------------------
   F19             there should be a way to navigate
                   the IVR
   ----------------------------------------------------------------
   F20             The browser must be able to send short
                   latency datagram traffic to a peer browser
   ----------------------------------------------------------------
   F21             The browser MUST be able to take advantage of
                   capabilities to prioritize voice and video
                   appropriately.
   ----------------------------------------------------------------
   F22             The browser SHOULD use encoding of streams
                   suitable for the current rendering (e.g.



Holmberg, et al.          Expires March 4, 2012                [Page 12]


Internet-Draft                   RTC-Web                  September 2011


                   video display size) and SHOULD change parameters
                   if the rendering changes during the session
   ----------------------------------------------------------------
   F23             It MUST be possible to move from one network
                   interface to another one
   ----------------------------------------------------------------
   F24             The browser MUST be able to initiate and accept a
                   media session where the data needed for establishment
                   can be carried in SIP.
   ----------------------------------------------------------------
   F25             The browser MUST support a baseline audio and
                   video codec
   ----------------------------------------------------------------
   F26             The browser MUST be able to send streams to a
                   peer in presence of NATs that block UDP traffic.
   ----------------------------------------------------------------

5.3.  API requirements


   REQ-ID          DESCRIPTION
   ----------------------------------------------------------------
   A1              The web application MUST be able to ask the
                   browser for permission to use cameras
                   and microphones as input devices.
   ----------------------------------------------------------------
   A2              The web application MUST be able to control how
                   streams generated by input devices are used.
   ----------------------------------------------------------------
   A3              The web application MUST be able to control the
                   local rendering of streams (locally generated streams
                   and streams received from a peer).
   ----------------------------------------------------------------
   A4              The web application MUST be able to initiate
                   sending of stream/stream components to a peer.
   ----------------------------------------------------------------
   A5              The web application MUST be able to control the
                   media format (codec) to be used for the streams
                   sent to a peer.

                   NOTE: The level of control depends on whether
                   the codec negotiation is handled by the browser
                   or the web application.
   ----------------------------------------------------------------
   A6              After a media stream has been established, the
                   web application MUST be able to modify the media
                   format for streams sent to a peer.
   ----------------------------------------------------------------



Holmberg, et al.          Expires March 4, 2012                [Page 13]


Internet-Draft                   RTC-Web                  September 2011


   A7              The web application MUST be made aware of
                   whether the establishment of a stream with a
                   peer was successful or not.
   ----------------------------------------------------------------
   A8              The web application MUST be able to
                   pause/unpause the sending of a stream to a peer.
   ----------------------------------------------------------------
   A9              The web application MUST be able to mute/unmute
                   a stream received from a peer.
   ----------------------------------------------------------------
   A10             The web application MUST be able to cease the
                   sending of a stream to a peer.
   ----------------------------------------------------------------
   A11             The web application MUST be able to cease
                   processing and rendering of a stream received
                   from a peer.
   ----------------------------------------------------------------
   A12             The web application MUST be informed when a
                   stream from a peer is no longer received.
   ----------------------------------------------------------------
   A13             The web application MUST be informed when high
                   loss rates occur.
   ----------------------------------------------------------------
   A14             It MUST be possible for the web application to
                   control panning, mixing and other processing for
                   individual streams.
   ----------------------------------------------------------------
   A15             The Web application must be provided with an
                   identifier for the stream that can be communicated
                   to the other party of the communication, and which
                   the other party can associate with its end of the
                   same stream.
   ----------------------------------------------------------------
   A16             It MUST be possible for the web application to
                   send and receive datagrams to/from peer
   ----------------------------------------------------------------
   A17             It MUST be possible for the web application to
                   indicate the type of audio signal (speech, audio)
   ----------------------------------------------------------------
   A18             It must be possible for an initiator or a
                   responder Web application to indicate the types
                   of media he's willing to accept incoming streams
                   for when setting up a connection (audio, video,
                   other). The types of media he's willing to accept
                   can be a subset of the types of media the browser
                   is able to accept.
   ----------------------------------------------------------------




Holmberg, et al.          Expires March 4, 2012                [Page 14]


Internet-Draft                   RTC-Web                  September 2011


6.  IANA Considerations

   TBD


7.  Security Considerations

7.1.  Introduction

   A malicious web application might use the browser to perform Denial
   Of Service (DOS) attacks on NAT infrastructure, or on peer devices.
   Also, a malicious web application might silently establish outgoing,
   and accept incoming, streams on an already established connection.

   Based on the identified security risks, this section will describe
   security considerations for the browser and web application.

7.2.  Browser Considerations

   The browser is expected to provide mechanisms for getting user
   consent to use device resources such as camera and microphone.

   The browser is expected to provide mechanisms for informing the user
   that device resources such as camera and microphone are in use
   ("hot").

   The browser is expected to provide mechanisms for users to revise
   consent to use device resources such as camera and microphone.

   The browser is expected to provide mechanisms in order to assure that
   streams are the ones the recipient intended to receive.

   The browser is needs to ensure that media is not sent, and that
   received media is not rendered, until the associated stream
   establishment and handshake procedures with the remote peer have been
   successfully finished.

   The browser needs to ensure that the stream negotiation procedures
   are not seen as Denial Of Service (DOS) by other entities.

7.3.  Web Application Considerations

   The web application is expected to ensure user consent in sending and
   receiving media streams.







Holmberg, et al.          Expires March 4, 2012                [Page 15]


Internet-Draft                   RTC-Web                  September 2011


8.  Additional use-cases

   Several additional use-cases have been discussed.  At this point
   these use-cases are not included as requirement deriving use-cases
   for different reasons (lack of documentation, overlap with existing
   use-cases, lack of consensus).  For completeness these additional
   use-cases are listed below:
   1.   Use-cases regarding different situations when being invited to a
        "session", e.g. browser open, browser open but another tab
        active, browser open but active in session, browser closed, ....
        (Matthew Kaufman); discussed at webrtc meeting
   2.   Different TURN provider scenarios (Cullen Jennings); discussed
        at the webrtc meeting
   3.   E911 (Paul Beaumont) http://www.ietf.org/mail-archive/web/
        rtcweb/current/msg00525.html, followed up by Stephan Wenger
   4.   Local Recording and Remote recording (John): Discussed a _lot_
        on the mail lists (rtcweb as well as public-webrtc) late August
        2011.  Not concluded at time of writing.
   5.   Emergency access for disabled (Bernard Aboba) http://
        www.ietf.org/mail-archive/web/rtcweb/current/msg00478.html
   6.   Clue use-cases (Roni Even) http://tools.ietf.org/html/
        draft-ietf-clue-telepresence-use-cases-01
   7.   Rohan red cross (Cullen Jennings); http://www.ietf.org/
        mail-archive/web/rtcweb/current/msg00323.html
   8.   Remote assistance (ala VNC or RDP) - User is helping another
        user on their computer with either view-only or view-with-
        control, either of just the browser of the the entire screen. ht
        tp://www.ietf.org/mail-archive/web/rtcweb/current/msg00543.html
   9.   Security camera/baby monitor usage http://www.ietf.org/
        mail-archive/web/rtcweb/current/msg00543.html
   10.  Large multiparty session http://www.ietf.org/mail-archive/web/
        rtcweb/current/msg00530.html


9.  Acknowledgements

   Stephan Wenger has provided a lot of useful input and feedback, as
   well as editorial comments.

   Harald Alvestrand and Ted Hardie have provided comments and feedback
   on the draft.

   Harald Alvestrand and Cullen Jennings have provided additional use-
   cases.

   Thank You to everyone in the RTCWEB community that have provided
   comments, feedback and improvement proposals on the draft content.




Holmberg, et al.          Expires March 4, 2012                [Page 16]


Internet-Draft                   RTC-Web                  September 2011


10.  Change Log

   [RFC EDITOR NOTE: Please remove this section when publishing]

   Changes from draft-ietf-rtcweb-use-cases-and-requirements-03

   o  Editorials
   o  Changed when the self-view is displayed in 4.2.1.1, and added
      words about allowing users to remove and re-insert it.
   o  Clarified 4.2.6.1
   o  Removed the "mono" stuff from 4.2.7.1
   o  Added that communication should not be possible to eavesdrop to
      most use cases - and req.  F17
   o  Re-phrased 4.3.3.1 to not describe the technical solution so much,
      and removed "stereo" stuff.  Solution possibilities are now in a
      note.
   o  Re-inserted API requirements after discussion in the W3C webrtc
      WG.  (Re-phrased A15 and added A18 compared to version -02).

   Changes from draft-ietf-rtcweb-use-cases-and-requirements-02

   o  Removed desrciption/list of API requirements, instead
   o  Reference to W3C webrtc_reqs document for API requirements

   Changes from draft-ietf-rtcweb-ucreqs-01

   o  Changed Intended status to Information
   o  Changed "Ipr" to "trust200902"
   o  Added use case "Simple video communication service, NAT/FW that
      blocks UDP", and derived new req F26
   o  Added use case "Distributed Music Band" and derived new req A17
   o  Added F24 as requirement derived from use case "Simple video
      communication service with inter-operator calling"
   o  Added section "Additional use cases"
   o  Added text about ID handling to multiparty with central server use
      case
   o  Re-phrased A1 slightly

   Changes from draft-ietf-rtcweb-ucreqs-00

   o  - Reshuffled: Just two main groups of use cases (b2b and b2GW/
      Server); removed some specific use cases and added them instead as
      flavors to the base use case (Simple video communciation)
   o  - Changed the fromulation of F19
   o  - Removed the requirement on an API for DTMF
   o  - Removed "FX3: There SHOULD be a mapping of the minimum needed
      data for setting up connections into SIP, so that the restriction
      to SIP-carriable data can be verified.  Not a rew on the browser



Holmberg, et al.          Expires March 4, 2012                [Page 17]


Internet-Draft                   RTC-Web                  September 2011


      but rather on a document"
   o  - (see
      http://www.ietf.org/mail-archive/web/rtcweb/current/msg00227.html
      for more details)
   o  -Added text on informing user of that mic/cam is being used and
      that it must be possible to revoce permission to use them in
      section 7.
   Changes from draft-holmberg-rtcweb-ucreqs-01
   o  - Draft name changed to draft-ietf-rtcweb-ucreqs
   o  - Use-case grouping introduced
   o  - Additional use-cases added
   o  - Additional reqs added (derived from use cases): F19-F25, A16-A17

   Changes from draft-holmberg-rtcweb-ucreqs-00
   o  - Mapping between use-cases and requirements added (Harald
      Alvestrand, 090311)
   o  - Additional security considerations text (Harald Alvestrand,
      090311)
   o  - Clarification that user applications are assumed to be executed
      by a browser (Ted Hardie, 080311)
   o  - Editorial corrections and clarifications


11.  References

11.1.  Normative References

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

11.2.  Informative References

   [webrtc_reqs]
              "Webrt requirements,
              http://dev.w3.org/2011/webrtc/editor/webrtc_reqs.html".


Authors' Addresses

   Christer Holmberg
   Ericsson
   Hirsalantie 11
   Jorvas  02420
   Finland

   Email: christer.holmberg@ericsson.com





Holmberg, et al.          Expires March 4, 2012                [Page 18]


Internet-Draft                   RTC-Web                  September 2011


   Stefan Hakansson
   Ericsson
   Laboratoriegrand 11
   Lulea  97128
   Sweden

   Email: stefan.lk.hakansson@ericsson.com


   Goran AP Eriksson
   Ericsson
   Farogatan 6
   Stockholm  16480
   Sweden

   Email: goran.ap.eriksson@ericsson.com



































Holmberg, et al.          Expires March 4, 2012                [Page 19]


Html markup produced by rfcmarkup 1.129c, available from https://tools.ietf.org/tools/rfcmarkup/