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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: December 29, 2013                                      Ericsson
                                                           June 27, 2013


         Web Real-Time Communication Use-cases and Requirements
          draft-ietf-rtcweb-use-cases-and-requirements-11.txt

Abstract

   This document describes web based real-time communication use-cases.
   Requirements on the browser functionality are derived from use-cases.

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 December 29, 2013.

Copyright Notice

   Copyright (c) 2013 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  . . . . . . . . . . . . . . . . . . . . . . . .   2
   2.  Definitions . . . . . . . . . . . . . . . . . . . . . . . . .   3
   3.  Use-cases . . . . . . . . . . . . . . . . . . . . . . . . . .   3
     3.1.  Introduction  . . . . . . . . . . . . . . . . . . . . . .   3
     3.2.  Browser-to-browser use-cases  . . . . . . . . . . . . . .   4
       3.2.1.  Simple Video Communication Service  . . . . . . . . .   4
       3.2.2.  Simple Video Communication Service, NAT/FW that
               blocks UDP  . . . . . . . . . . . . . . . . . . . . .   5
       3.2.3.  Simple Video Communication Service, FW that only
               allows http . . . . . . . . . . . . . . . . . . . . .   5
       3.2.4.  Simple Video Communication Service, global service
               provider  . . . . . . . . . . . . . . . . . . . . . .   6
       3.2.5.  Simple Video Communication Service, enterprise
               aspects . . . . . . . . . . . . . . . . . . . . . . .   6
       3.2.6.  Simple Video Communication Service, access change . .   7
       3.2.7.  Simple Video Communication Service, QoS . . . . . . .   7
       3.2.8.  Simple Video Communication Service with sharing . . .   8
       3.2.9.  Simple Video Communication Service with file exchange   8
       3.2.10. Simple video communication service with inter-
               operator calling  . . . . . . . . . . . . . . . . . .   8
       3.2.11. Hockey Game Viewer  . . . . . . . . . . . . . . . . .   9
       3.2.12. Multiparty video communication  . . . . . . . . . . .  10
       3.2.13. Multiparty on-line game with voice communication  . .  11
       3.2.14. Distributed Music Band  . . . . . . . . . . . . . . .  12
     3.3.  Browser - GW/Server use cases . . . . . . . . . . . . . .  12
       3.3.1.  Telephony terminal  . . . . . . . . . . . . . . . . .  12
       3.3.2.  Fedex Call  . . . . . . . . . . . . . . . . . . . . .  13
       3.3.3.  Video conferencing system with central server . . . .  13
   4.  Requirements  . . . . . . . . . . . . . . . . . . . . . . . .  14
     4.1.  General . . . . . . . . . . . . . . . . . . . . . . . . .  14
     4.2.  Browser requirements  . . . . . . . . . . . . . . . . . .  15
   5.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  18
   6.  Security Considerations . . . . . . . . . . . . . . . . . . .  18
     6.1.  Introduction  . . . . . . . . . . . . . . . . . . . . . .  18
     6.2.  Browser Considerations  . . . . . . . . . . . . . . . . .  18
     6.3.  Web Application Considerations  . . . . . . . . . . . . .  19
   7.  Additional use-cases  . . . . . . . . . . . . . . . . . . . .  19
   8.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  20
   9.  Change Log  . . . . . . . . . . . . . . . . . . . . . . . . .  21
   10. Normative References  . . . . . . . . . . . . . . . . . . . .  26
   Appendix A.  API requirements . . . . . . . . . . . . . . . . . .  27
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  29

1.  Introduction





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   This document presents a few use-cases of web applications that are
   executed in a browser and use real-time communication capabilities.
   In most of the use-cases all end-user clients are web applications,
   but there are some use-cases where at least one of the end-user
   client is of another type (e.g. a telephone).

   Based on the use-cases, the document derives requirements related to
   browser functionality.  These requirements are named "Fn", where n is
   an integer, and are described in Section 4.2.

   This document was developed in an initial phase of the work with
   rather minor updates at later stages.  It has not really served as a
   tool in deciding features or scope for the WGs efforts so far.  It is
   proposed to be used in a later phase to evaluate the protocols and
   solutions developed by the WG.

   This document also lists requirements related to the API to be used
   by web applications as an appendix.  The reason is that the W3C
   WebRTC WG has decided to not develop its own use-case/requirement
   document, but instead use this document.  These requirements are
   named "An", where n is an integer, and are described in Appendix A-

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

2.  Definitions

   TBD

3.  Use-cases

3.1.  Introduction

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

   The following considerations are applicable to all use cases:

   o  Clients can be on IPv4-only

   o  Clients can be on IPv6-only

   o  Clients can be on dual-stack

   o  Clients can be on wideband (10s of Mbits/sec)




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   o  Clients can be on narrowband (10s to 100s of Kbits/sec)

   o  Clients can be on variable-media-quality networks (wireless)

   o  Clients can be on congested networks

   o  Clients can be on firewalled networks with no UDP allowed

   o  Clients can be on networks with any type (as described in RFC4787)
      of NAT.

3.2.  Browser-to-browser use-cases

3.2.1.  Simple Video Communication Service

3.2.1.1.  Description

   Two or more users have loaded a video communication web application
   into their browsers, provided by the same service provider, and
   logged into the service it provides.  The web service publishes
   information about user login status by pushing updates to the web
   application in the browsers.  When one online user selects a peer
   online user, a 1-1 audiovisual communication session between the
   browsers of the two 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 in addition to the self-view.  During the session, each
   user can select to remove and re-insert the self-view as often as
   desired.  Each user can also change the sizes of his/her two video
   displays during the session.  Each user can also pause sending of
   media (audio, video, or both) and mute incoming media

   It is essential that the communication cannot be wiretapped
   [RFC2804].

   It is essential that media and data be encrypted, authenticated and
   integrity protected on a per-packet basis and that media and data
   packets failing the integrity check not be delivered to the
   application.

   In addition, it is required that browsers enable the media and data
   security keys to be cryptographically bound to the user identity.

   The application gives the users the opportunity to stop it from
   exposing the host IP address to the application of the other user.




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   Any session participant can end the session at any time.

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

   One user has an unreliable Internet connection.  It sometimes loses
   packets, and sometimes goes down completely.

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

   The web service monitors the quality of the service (focus on quality
   of audio and video) the end-users experience.

3.2.1.2.  Derived Requirements

   F1, F2, F3, F4, F5, F8, F9, F10, F20, F25, F28, F35, F36, F38, F39

   A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12, A25, A26

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

3.2.2.1.  Description

   This use-case is almost identical to the Simple Video Communication
   Service use-case (Section 3.2.1).  The difference is that one of the
   users is behind a NAT that blocks UDP traffic.

3.2.2.2.  Derived Requirements

   F1, F2, F3, F4, F5, F8, F9, F10, F20, F25, F28, F29

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

3.2.3.  Simple Video Communication Service, FW that only allows http

3.2.3.1.  Description

   This use-case is almost identical to the Simple Video Communication
   Service use-case (Section 3.2.1).  The difference is that one of the
   users is behind a FW that only allows http traffic.

3.2.3.2.  Derived Requirements

   F1, F2, F3, F4, F5, F8, F9, F10, F20, F25, F28, F37

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





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3.2.4.  Simple Video Communication Service, global service provider

3.2.4.1.  Description

   This use-case is almost identical to the Simple Video Communication
   Service use-case (Section 3.2.1).

   What is added is that the service provider is operating over large
   geographical areas (or even globally).

   Assuming that ICE will be used, this means that the service provider
   would like to be able to provide several STUN and TURN servers (via
   the app) to the browser; selection of which one(s) to use is part of
   the ICE processing.  Other reasons for wanting to provide several
   STUN and TURN servers include support for IPv4 and IPv6, load
   balancing and redundancy.

3.2.4.2.  Derived Requirements

   F1, F2, F3, F4, F5, F8, F9, F10, F20, F25, F28, F31

   A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12, A22

3.2.5.  Simple Video Communication Service, enterprise aspects

3.2.5.1.  Description

   This use-case is similar to the Simple Video Communication Service
   use-case (Section 3.2.1).

   What is added is aspects when using the service in enterprises.  ICE
   is assumed in the further description of this use-case.

   An enterprise that uses a RTCWEB based web application for
   communication desires to audit all RTCWEB based application session
   used from inside the company towards any external peer.  To be able
   to do this they deploy a TURN server that straddle the boundary
   between the internal network and the external.

   The firewall will block all attempts to use STUN with an external
   destination unless they go to the enterprise auditing TURN server.
   In cases where employees are using RTCWEB applications provided by an
   external service provider they still want to have the traffic to stay
   inside their internal network and in addition not load the straddling
   TURN server, thus they deploy a STUN server allowing the RTCWEB
   client to determine its server reflexive address on the internal
   side.  Thus enabling cases where peers are both on the internal side
   to connect without the traffic leaving the internal network.  It must



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   be possibele to configure the browsers used in the enterprise with
   network specific STUN and TURN servers.  This should be possible to
   achieve by autoconfiguration methods.  The RTCWEB functionality will
   need to utilize both network specific STUN and TURN resources and
   STUN and TURN servers provisioned by the web application.

3.2.5.2.  Derived Requirements

   F1, F2, F3, F4, F5, F8, F9, F10, F20, F25, F28, F32

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

3.2.6.  Simple Video Communication Service, access change

3.2.6.1.  Description

   This use-case is almost identical to the Simple Video Communication
   Service use-case (Section 3.2.1).  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 accessing 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.

3.2.6.2.  Derived Requirements

   F1, F2, F3, F4, F5, F8, F9, F10, F20, F25, F26, F28

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

3.2.7.  Simple Video Communication Service, QoS

3.2.7.1.  Description

   This use-case is almost identical to the Simple Video Communication
   Service, access change use-case (Section 3.2.6).  The use of Quality
   of Service (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.



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3.2.7.2.  Derived Requirements

   F1, F2, F3, F4, F5, F8, F9, F10, F20, F24, F25, F26, F28

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

3.2.8.  Simple Video Communication Service with sharing

3.2.8.1.  Description

   This use-case has the audio and video communication of the Simple
   Video Communication Service use-case (Section 3.2.1).

   But in addition to this, one of the users can share what is being
   displayed on her/his screen with a peer.  The user can choose to
   share the entire screen, part of the screen (part selected by the
   user) or what a selected applicaton displays with the peer.

3.2.8.2.  Derived Requirements

   F1, F2, F3, F4, F5, F8, F9, F10, F20, F25, F28, F30

   A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12, A21

3.2.9.  Simple Video Communication Service with file exchange

3.2.9.1.  Description

   This use-case has the audio and video communication of the Simple
   Video Communication Service use-case (Section 3.2.1).

   But in addition to this, the users can send and receive files stored
   in the file system of the device used.

3.2.9.2.  Derived Requirements

   F1, F2, F3, F4, F5, F8, F9, F10, F20, F25, F28, F30, F33

   A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12, A21, A24

3.2.10.  Simple video communication service with inter-operator calling

3.2.10.1.  Description

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





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   The service providers are interconnected by some means, but exchange
   no more information about the users than what can be carried using
   SIP.

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

   For each user Alice who has authorized another user Bob to receive
   login status information, Alice's service publishes Alice's login
   status information to Bob. How this authorization is defined and
   established is out of scope.

   The same functionality as in the the Simple Video Communication
   Service use-case (Section 3.2.1) is available.

   The same issues with connectivity apply.

3.2.10.2.  Derived requirements

   F1, F2, F3, F4, F5, F8, F9, F10, F20, F25, F27, F28

   A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12, A20

3.2.11.  Hockey Game Viewer

3.2.11.1.  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 audiovisual communication session.  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 of 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



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   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
   picture-in-picture thumbnails of the rear facing camera (self-view)
   and the desktop camera.  As the most important stream in this phase
   is the video showing the game, the application used in the talent
   scout's mobile sets higher priority for that stream.

   It is essential that the communication cannot be wiretapped
   [RFC2804].

3.2.11.2.  Derived Requirements

   F1, F2, F3, F4, F5, F8, F9, F10, F17, F20, F34

   A1, A2, A3, A4, A5, A7, A8, A9, A10, A11, A12, A17, A23

3.2.12.  Multiparty video communication

3.2.12.1.  Description

   In this use-case is the Simple Video Communication Service use-case
   (Section 3.2.1) 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.  In addition
   the levels in the audio signals are adjusted before mixing.

   Another feature intended to enhance the use experience is that the
   video window that displays the video of the currently speaking peer
   is highlighted.

   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 cannot be wiretapped
   [RFC2804].

   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, level adjust and



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   mix the audio from all received streams locally in the browser.  It
   also adds the capability to measure the audio level/activity.

3.2.12.2.  Derived Requirements

   F1, F2, F3, F4, F5, F8, F9, F10, F11, F12, F13, F14, F15, F16, F17,
   F20, F25

   A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12, A13, A14, A15,
   A16, A17

3.2.13.  Multiparty on-line game with voice communication

3.2.13.1.  Description

   This use case is based on the previous one.  In this use-case, the
   voice part of the multiparty video communication use case 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, and have higher priority
   than the voice.

   It is essential that the communication cannot be wiretapped
   [RFC2804].

   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.

3.2.13.2.  Derived Requirements

   F1, F2, F3, F4, F5, F8, F9, F11, F12, F13, F14, F15, F16, F18, F20,
   F23, F34

   A1, A2, A3, A4, A5, A7, A8, A9, A10, A11, A12, A13, A14, A15, A16,
   A17, A18, A23









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3.2.14.  Distributed Music Band

3.2.14.1.  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.).

3.2.14.2.  Derived Requirements

   F1, F2, F3, F4, F5, F8, F9, F11, F12, F13, F14, F15, F16

   A1, A2, A3, A4, A5, A7, A8, A9, A10, A11, A12, A13, A14, A15, A16,
   A19

3.3.  Browser - GW/Server use cases

3.3.1.  Telephony terminal

3.3.1.1.  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 is used.

   It is essential that the communication cannot be wiretapped
   [RFC2804].

   Note: With "place and receive calls in the same way as when using a
   normal mobile phone" it is meant that you can dial a number, and that
   your mobile telephony operator has made available your phone contacts
   on line, so they are available and can be clicked to call, and be
   used to present the identity of an incoming call.  If the callee is
   not in your phone contacts the number is displayed.  Furthermore,
   your call logs are available, and updated with the calls made/
   received from the browser.  And for people receiving calls made from
   the web browser the usual identity (i.e. the phone number of the
   mobile phone) will be presented.



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3.3.1.2.  Derived Requirements

   F1, F2, F3, F4, F5, F8, F9, F10, F20, F21

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

3.3.2.  Fedex Call

3.3.2.1.  Description

   Alice uses her web browser with a service that allows her to call
   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.

3.3.2.2.  Derived Requirements

   F1, F2, F3, F4, F5, F8, F9, F10, F21, F22

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

3.3.3.  Video conferencing system with central server

3.3.3.1.  Description

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

   The browser 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
   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 cannot physically



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   access the internal network, they can establish a Virtual Private
   Network (VPN).

   It is essential that the communication cannot be wiretapped
   [RFC2804].

   All participants 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 exist 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, and
   the server just selects or c) each browser sends just a high
   resolution stream, the server transcodes into low resolution streams
   as required.

3.3.3.2.  Derived Requirements

   F1, F2, F3, F4, F5, F7, F8, F9, F10, F17, F19, F20

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

4.  Requirements

4.1.  General

   This section contains the requirements on the browser derived from
   the use-cases in Section 3.

   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.









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4.2.  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 and
           data to a peer in the presence  of NATs.
   ----------------------------------------------------------------
   F3      Transmitted streams and data must be rate
           controlled (meaning that the browser must, regardless
           of application behavior, reduce send rate when
           there is congestion).
   ----------------------------------------------------------------
   F4      The browser must be able to receive, process and
           render streams and data ("render" does not
           apply for data) from peers.
   ----------------------------------------------------------------
   F5      The browser should be able to render good quality
           audio and video even in the presence of
           reasonable levels of jitter and packet losses.
   ----------------------------------------------------------------
   F7      The browser must support insertion of reference frames
           in ougoing media streams when requested by a peer.
   ----------------------------------------------------------------
   F8      The browser must detect when a stream from a
           peer is not received anymore
   ----------------------------------------------------------------
   F9      When there are both incoming and outgoing audio
           streams, echo cancellation must be made
           available to avoid disturbing echo during
           conversation.
   ----------------------------------------------------------------
   F10     The browser must support synchronization of
           audio and video.
   ----------------------------------------------------------------
   F11     The browser must be able to transmit streams and
           data to several peers concurrently.
   ----------------------------------------------------------------
   F12     The browser must be able to receive streams and
           data from multiple peers concurrently.
   ----------------------------------------------------------------
   F13     The browser must be able to apply spatialization
           effects when playing audio streams.
   ----------------------------------------------------------------
   F14     The browser must be able to measure the level



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           in audio streams.
   ----------------------------------------------------------------
   F15     The browser must be able to change the level
           in audio streams.
   ----------------------------------------------------------------
   F16     The browser must be able to render several
           concurrent video streams
   ----------------------------------------------------------------
   F17     The browser must be able to mix several
           audio streams.
   ----------------------------------------------------------------
   F18     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.
   ----------------------------------------------------------------
   F19     Streams and data must be able to pass through
           limited middleboxes.
   ----------------------------------------------------------------
   F20     It must be possible to protect streams and data
           from wiretapping [RFC2804].
   ----------------------------------------------------------------
   F21     The browser must support an audio media format
           (codec) that is commonly supported by existing
           telephony services.
   ----------------------------------------------------------------
   F22     There should be a way to navigate
           a Dual-tone multi-frequency signaling (DTMF)
           based Interactive voice response (IVR) System
   ----------------------------------------------------------------
   F23     The browser must be able to send short
           latency unreliable datagram traffic to a
           peer browser [RFC5405].
   ----------------------------------------------------------------
   F24     The browser should be able to take advantage
           of available capabilities (supplied by network
           nodes) to prioritize voice, video and data
           appropriately.
   ----------------------------------------------------------------
   F25     The browser should use encoding of streams
           suitable for the current rendering (e.g.
           video display size) and should change parameters
           if the rendering changes during the session
   ----------------------------------------------------------------
   F26     It must be possible to move from one network
           interface to another one
   ----------------------------------------------------------------
   F27     The browser must be able to initiate and



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           accept a media session where the data needed
           for establishment can be carried in SIP.
   ----------------------------------------------------------------
   F28     The browser must support a baseline audio and
           video codec
   ----------------------------------------------------------------
   F29     The browser must be able to send streams and
           data to a peer in the presence of NATs that
           block UDP traffic.
   ----------------------------------------------------------------
   F30     The browser must be able to use the screen (or
           a specific area of the screen) or what a certain
           application displays on the screen to generate
           streams.
   ----------------------------------------------------------------
   F31     The browser must be able to use several STUN
           and TURN servers
   ----------------------------------------------------------------
   F32     There browser must support that STUN and TURN
           servers to use are supplied by other entities
           than via the web application (i.e. the network
           provider).
   ----------------------------------------------------------------
   F33     The browser must be able to send reliable
           data traffic to a peer browser.
   ----------------------------------------------------------------
   F34     The browser must support priortization of
           streams and data.
   ----------------------------------------------------------------
   F35     The browser must enable verification, given
           the right circumstances and by use of other
           trusted communication, of that  streams and
           data received have not been manipulated by
           any party.
   ----------------------------------------------------------------
   F36     The browser must encrypt, authenticate and
           integrity protect media and data on a
           per-packet asis, and must drop incoming media
           and data packets that fail the per-packet
           integrity check.  In addition, the browser
           must support a mechanism for cryptographically
           binding media and data security keys to the
           user identity (see R-ID-BINDING in [RFC5479]).
   ----------------------------------------------------------------
   F37     The browser must be able to send streams and
           data to a peer in the presence of FWs that only
           allows http(s) traffic.
   ----------------------------------------------------------------



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   F38     The browser must be able to collect statistics,
           related to the transport of audio and video
           between peers, needed to estimate quality of
           experience.
   ----------------------------------------------------------------
   F39     The browser must make it possible to set up a
           call between two parties without one party
           learning the other party's host IP address.
   ----------------------------------------------------------------




5.  IANA Considerations

   TBD

6.  Security Considerations

6.1.  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.

6.2.  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 and
   even completely revoke consent to use device resources such as camera
   and microphone.

   The browser is expected to provide mechanisms for getting user
   consent to use the screen (or a certain part of it) or what a certain
   application displays on the screen as source for streams.






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   The browser is expected to provide mechanisms for informing the user
   that the screen, part thereof or an application is serving as a
   stream source ("hot").

   The browser is expected to provide mechanisms for users to revise and
   even completely revoke consent to use the screen, part thereof or an
   application is serving as a stream source.

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

   The browser is expected to provide mechanisms that allows the users
   to verify that the streams received have not be manipulated (F35).

   The browser 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.

6.3.  Web Application Considerations

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

7.  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.   E911 (Paul Beaumont) http://www.ietf.org/mail-archive/web/rtcweb
        /current/msg00525.html, followed up by Stephan Wenger









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   3.   Local Recording and Remote recording (John): Discussed a _lot_
        on the mail lists (rtcweb as well as public-webrtc) lAugust and
        September 2011.  Concrete proposal: http://www.ietf.org/mail-
        archive/web/rtcweb/current/msg01006.html (remote) and http://
        www.ietf.org/mail-archive/web/rtcweb/current/msg00734.html
        (local)

   4.   Emergency access for disabled (Bernard Aboba) http://
        www.ietf.org/mail-archive/web/rtcweb/current/msg00478.html

   5.   Clue use-cases (Roni Even) http://tools.ietf.org/html/draft-
        ietf-clue-telepresence-use-cases-01

   6.   Rohan red cross (Cullen Jennings); http://www.ietf.org/mail-
        archive/web/rtcweb/current/msg00323.html

   7.   Security camera/baby monitor usage http://www.ietf.org/mail-
        archive/web/rtcweb/current/msg00543.html

   8.   Large multiparty session http://www.ietf.org/mail-archive/web/
        rtcweb/current/msg00530.html

   9.   Call center http://www.ietf.org/mail-archive/web/rtcweb/current/
        msg04203.html

   10.  Enterprise policies http://www.ietf.org/mail-archive/web/rtcweb/
        current/msg04271.html

   11.  Low-complex multiparty central node http://www.ietf.org/mail-
        archive/web/rtcweb/current/msg04430.html

   12.  Multiparty central node that is not allowed to decipher http://
        www.ietf.org/mail-archive/web/rtcweb/current/msg04457.html

   13.  Enable company coop without being able to decipher http://
        www.ietf.org/mail-archive/web/rtcweb/current/msg04461.html

8.  Acknowledgements

   Bernard Aboba, Gunnar Hellstrom, Martin Thomson, Lars Eggert, Matthew
   Kaufman, Emil Ivov, Eric Rescorla, Eric Burger, John Leslie, Dan
   Wing, Richard Barnes, Barry Dingle, Dale Worley, Ted hardie, Mary
   Barnes,

   Dan Burnett has reviewed and proposed a lot of things that enhances
   the document.  Most of this has been incorporated in rev -05.





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   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.

9.  Change Log

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

   Changes from draft-ietf-rtcweb-use-cases-and-requirements-10

   o  Described that the API requirements are really from a W3C
      perspective and are supplied as an appendix in the introduction.
      Moved API requirements to an Appendix.

   o  Removed the "Conventions" section with the key-words and reference
      to RFC2119.  Also changed uppercase MUST's/SHOULD's to lowercase.

   o  Added a note on the proposed use of the document to the
      introduction.

   o  Removed the note talking about WS from the "FW that only allows
      http" use-case.

   o  Removed the word "Skype" that was used as example in one of the
      use-cases.

   o  Clarified F3 (the req saying the everything the browser sends must
      be rate controlled).

   o  Removed the TBD saying we need to define reasonable levels from
      the requirement saying that quality must be good even in presence
      of packet losses (F5), and changed "must" to "should" (Based on a
      list discussion involving Bernard).

   o  Removed F6 ("The browser must be able to handle high loss and
      jitter levels in a graceful way."), also after a list discussion.

   o  Clarified F7 (used to say that the browser must support fast
      stream switches, now says that reference frames must be inserted
      when requested).



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   o  Removed the questions from F9 (echo cancellation), F10
      (syncronization), F21 (telephony codec).

   o  Exchanged "restrictive firewalls" for "limited middleboxes" in F19
      (as proposed by Martin).

   o  Expanded DTMF and IVR in F22 (proposed by Martin)

   o  Added ref to RFC5405 in F23 (proposed by Lars Eggert).

   o  Exchanged "service provided" for "web application" in F32.

   o  Changed the text in 3.2.1 that motivates F36 (new text "It is
      essential that media and data be encrypted, authenticated ...
      bound to the user identity."); and rewrote F36, included a ref to
      RFC5479.

   o  Changed "quality of service" to "quality of experience" in F38.

   o  Added F39.

   o  Used new formulation of A17 (proposed by Martin).

   o  Updated A20.

   o  Updated A25.

   Changes from draft-ietf-rtcweb-use-cases-and-requirements-09

   o  Changed "video communication session" to "audiovisual
      communication session.

   Changes from draft-ietf-rtcweb-use-cases-and-requirements-08

   o  Changed "eavesdropping" to "wiretapping" and referenced RFC2804.

   o  Removed informal ref webrtc_req; that document has been abandoned
      by the W3C webrtc WG.

   o  Added use-case where one user is behind a FW that only allows
      http; derived req.  F37.

   o  Changed F24 slightly; MUST-> SHOULD, inserted "available".

   o  Added a clause to "Simple video communication service" saying that
      the service provider monitors the quality of service, and derived
      reqs F38 and A26.




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   Changes from draft-ietf-rtcweb-use-cases-and-requirements-07

   o  Added "and data exchange" to 1.  Introduction.

   o  Removed cone and symmetric NAT from 4.1 Introduction, refers to
      RFC4787 instead.

   o  Added text on enabling verifyication of that the media has not
      been manipulated by anyone to use-case "Simple Video Communication
      Service", derived req.  F35

   o  Added text on that the browser should reject media (data) that has
      been created/injected/modified by non-trusted party, derived req.
      F36

   o  Added text on enabling the app to refrain from revealing IP
      address to use-case "Simple Video Communication Service", derived
      req.  A25

   o  Added use-case "Simple Video Communication Service with file
      exchange", derived reqs F33 and A24

   o  Added priority of video streams to "Hockey game viewer" use case,
      added priority of data to "on-line game use-case", derived reqs
      F34 and A23

   o  In F22, "the IVR" -> "a DTMF based IVR".

   o  Updated req F23 to clarify that requirements such as NAT
      traversal, prtoection from eavesdropping, rate control applies
      also to datagram.

   Changes from draft-ietf-rtcweb-use-cases-and-requirements-06

   o  Renaming of requirements (FaI1 -> F31), (FaI2 -> F32) and (AaI1 ->
      A22)

   Changes from draft-ietf-rtcweb-use-cases-and-requirements-05

   o  Added use-case "global service provider", derived reqs associated
      with several STUN/TURN servers

   o  Added use-case "enterprise aspects", derived req associated with
      enabling the network provider to supply STUN and TURN servers

   o  The requirements from the above are ICE specific and labeled
      accordingly




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   o  Separated the requirements phrased like "processing such as pan,
      mix and render" for audio to be specific reqs on spatialization,
      level measurement, level adjustment and mixing (discussed on the
      lists in http://www.ietf.org/mail-archive/web/rtcweb/current/
      msg01648.html and http://lists.w3.org/Archives/Public/public-
      webrtc/2011Sep/0102.html)

   o  Added use-case on sharing as decided in http://www.ietf.org/mail-
      archive/web/rtcweb/current/msg01700.html, derived reqs F30 and A21

   o  Added the list of common considerations proposed in mail http://
      www.ietf.org/mail-archive/web/rtcweb/current/msg01562.html to the
      Introduction of the use-case section

   Changes from draft-ietf-rtcweb-use-cases-and-requirements-04

   o  Most changes based on the input from Dan Burnett http://
      www.ietf.org/mail-archive/web/rtcweb/current/msg00948.html

   o  Many editorial changes

   o  4.2.1.1 Clarified

   o  Some clarification added to 4.3.1.1 as a note

   o  F-requirements updated (see reply to Dan's mail).

   o  Almost all A-requirements updated to start "The Web API MUST
      provide ..."

   o  A8 removed, A9 rephrased to cover A8 and old A9

   o  A15 rephrased

   o  For more details, and discussion, look att the response to Dan's
      mail http://www.ietf.org/mail-archive/web/rtcweb/current/
      msg01177.html

   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



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   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





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   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
      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

10.  Normative References

   [RFC2804]  IAB IESG, "IETF Policy on Wiretapping", RFC 2804, May
              2000.

   [RFC5405]  Eggert, L. and G. Fairhurst, "Unicast UDP Usage Guidelines
              for Application Designers", BCP 145, RFC 5405, November
              2008.

   [RFC5479]  Wing, D., Fries, S., Tschofenig, H., and F. Audet,
              "Requirements and Analysis of Media Security Management
              Protocols", RFC 5479, April 2009.




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Appendix A.  API requirements

   This section contains the requirements on the API derived from the
   use-cases in Section 3.


   REQ-ID      DESCRIPTION
   ----------------------------------------------------------------
   A1      The Web API must provide means for the
           application to ask the browser for permission
           to use cameras and microphones as input devices.
   ----------------------------------------------------------------
   A2      The Web API must provide means for the web
           application to control how streams generated
           by input devices are used.
   ----------------------------------------------------------------
   A3      The Web API must provide means for the web
           application to control the local rendering of
           streams (locally generated streams and streams
           received from a peer).
   ----------------------------------------------------------------
   A4      The Web API must provide means for the web
           application to initiate sending of
           stream/stream components to a peer.
   ----------------------------------------------------------------
   A5      The Web API must provide means for the web
           application 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      The Web API must provide means for the web
           application to modify the media format for
           streams sent to a peer after a media stream
           has been established.
   ----------------------------------------------------------------
   A7      The Web API must provide means for
           informing the web application of whether the
           establishment of a stream with a peer was
           successful or not.
   ----------------------------------------------------------------
   A8      The Web API must provide means for the web
           application to mute/unmute a stream or stream
           component(s). When a stream is sent to a peer
           mute status must be preserved in the stream
           received by the peer.



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   ----------------------------------------------------------------
   A9      The Web API must provide means for the web
           application to cease the sending of a stream
           to a peer.
   ----------------------------------------------------------------
   A10     The Web API must provide means for the web
           application to cease processing and rendering
           of a stream received from a peer.
   ----------------------------------------------------------------
   A11         The Web API must provide means for
           informing the web application when a
           stream from a peer is no longer received.
   ----------------------------------------------------------------
   A12         The Web API must provide means for
           informing the web application when high
           loss rates occur.
   ----------------------------------------------------------------
   A13     The Web API must provide means for the web
           application to apply spatialization effects to
           audio streams.
   ----------------------------------------------------------------
   A14     The Web API must provide means for the web
           application to detect the level in audio
           streams.
   ----------------------------------------------------------------
   A15     The Web API must provide means for the web
           application to adjust the level in audio
           streams.
   ----------------------------------------------------------------
   A16     The Web API must provide means for the web
           application to mix audio streams.
   ----------------------------------------------------------------
   A17     The Web API must provide a way to identify
           streams such that an application is able to
           match streams on a sending peer with the same
           stream on all receiving peers.
   ----------------------------------------------------------------
   A18     The Web API must provide a mechanism for sending
           and receiving isolated discrete chunks of data.
   ----------------------------------------------------------------
   A19     The Web API must provide means for the web
           application to indicate the type of audio signal
           (speech, audio) for audio stream(s)/stream
           component(s).
   ----------------------------------------------------------------
   A20     It must be possible for an initiator or a
           responder web application to indicate the types
           of media it is willing to accept incoming



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           streams for when setting up a connection (audio,
           video, other). The types of media to be accepted
           can be a subset of the types of media the browser
           is able to accept.
   ----------------------------------------------------------------
   A21     The Web API must provide means for the
           application to ask the browser for permission
           to the screen, a certain area on the screen
           or what a certain application displays on the
           screen as input to streams.
   ----------------------------------------------------------------
   A22     The Web API must provide means for the
           application to specify several STUN and/or
           TURN servers to use.
   ----------------------------------------------------------------
   A23     The Web API must provide means for the
           application to specify the priority to
           apply for outgoing streams and data.
   ----------------------------------------------------------------
   A24     The Web API must provide a mechanism for sending
           and receiving files.
   ----------------------------------------------------------------
   A25     It must be possible for the application to
           instruct the browser to refrain from exposing
           the host IP address to the application
   ----------------------------------------------------------------
   A26     The Web API must provide means for the
           application to obtain the statistics (related
           to transport, and collected by the browser)
           needed to estimate quality of service.
   ----------------------------------------------------------------


Authors' Addresses

   Christer Holmberg
   Ericsson
   Hirsalantie 11
   Jorvas  02420
   Finland

   Email: christer.holmberg@ericsson.com









Holmberg, et al.        Expires December 29, 2013              [Page 29]


Internet-Draft                   RTC-Web                       June 2013


   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 December 29, 2013              [Page 30]


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