path: root/doc/rfc/rfc7478.txt

Internet Engineering Task Force (IETF)                       C. Holmberg
Request for Comments: 7478                                  S. Hakansson
Category: Informational                                      G. Eriksson
ISSN: 2070-1721                                                 Ericsson
                                                              March 2015


         Web Real-Time Communication Use Cases and Requirements

Abstract

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

   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 WG's efforts so far.  It
   is being published to record the early conclusions of the WG.  It
   will not be used as a set of rigid guidelines that specifications and
   implementations will be held to in the future.

Status of This Memo

   This document is not an Internet Standards Track specification; it is
   published for informational purposes.

   This document is a product of the Internet Engineering Task Force
   (IETF).  It represents the consensus of the IETF community.  It has
   received public review and has been approved for publication by the
   Internet Engineering Steering Group (IESG).  Not all documents
   approved by the IESG are a candidate for any level of Internet
   Standard; see Section 2 of RFC 5741.

   Information about the current status of this document, any errata,
   and how to provide feedback on it may be obtained at
   http://www.rfc-editor.org/info/rfc7478.














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Copyright Notice

   Copyright (c) 2015 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

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





































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Table of Contents

   1. Introduction ....................................................4
   2. Use Cases .......................................................4
      2.1. Introduction ...............................................4
      2.2. Common Requirements ........................................5
      2.3. Browser-to-Browser Use Cases ...............................5
           2.3.1. Simple Video Communication Service ..................5
           2.3.2. Simple Video Communication Service:
                  NAT/Firewall That Blocks UDP ........................8
           2.3.3. Simple Video Communication Service: Firewall
                  That Only Allows Traffic via an HTTP Proxy ..........8
           2.3.4. Simple Video Communication Service: Global
                  Service Provider ....................................8
           2.3.5. Simple Video Communication Service:
                  Enterprise Aspects ..................................9
           2.3.6. Simple Video Communication Service: Access Change ..10
           2.3.7. Simple Video Communication Service: QoS ............11
           2.3.8. Simple Video Communication Service with
                  Screen Sharing .....................................11
           2.3.9. Simple Video Communication Service with
                  File Exchange ......................................12
           2.3.10. Hockey Game Viewer ................................12
           2.3.11. Multiparty Video Communication ....................14
           2.3.12. Multiparty Online Game with Voice Communication ...15
      2.4. Browser - GW/Server Use Cases .............................17
           2.4.1. Telephony Terminal .................................17
           2.4.2. FedEx Call .........................................17
           2.4.3. Video Conferencing System with Central Server ......18
   3. Requirements Summary ...........................................19
      3.1. General ...................................................19
      3.2. Browser Requirements ......................................19
   4. Security Considerations ........................................23
      4.1. Introduction ..............................................23
      4.2. Browser Considerations ....................................24
      4.3. Web Application Considerations ............................24
   5. Normative References ...........................................25
   Appendix A. API Requirements ......................................26
   Acknowledgements ..................................................29
   Authors' Addresses ................................................29











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1.  Introduction

   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
   clients is of another type (e.g., a mobile phone or a SIP User Agent
   (UA)).

   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 listed in conjunction with the use cases.  A
   summary is provided in Section 3.2.

   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 WG's 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 or requirement
   document, but instead will use this document.  These requirements are
   named "An", where n is an integer, and are described in Appendix A.

   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 WG's efforts so far.  It
   is being published to record the early conclusions of the WG.  It
   will not be used as a set of rigid guidelines that specifications and
   implementations will be held to in the future.

2.  Use Cases

2.1.  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




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   o  Clients can be connected to networks with different throughput
      capabilities

   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 a NAT or IPv4-IPv6 translation
      devices using any type of Mapping and Filtering behaviors (as
      described in RFC 4787).

2.2.  Common Requirements

   The requirements retrieved from the
   Simple Video Communication Service use case (Section 2.3.1) by
   default apply to all other use cases and are considered common.  For
   each use case, only the additional requirements are listed.

2.3.  Browser-to-Browser Use Cases

2.3.1.  Simple Video Communication Service

2.3.1.1.  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:

   o  select to remove and reinsert the self-view as often as desired,

   o  change the sizes of his/her two video displays during the session,
      and

   o  pause the sending of media (audio, video, or both) and mute
      incoming media.




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

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

   Any session participant can end the session at any time.

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

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

2.3.1.2.  Common 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.
   ----------------------------------------------------------------
   F6              The browser must detect when a stream from a
                   peer is not received anymore.









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   ----------------------------------------------------------------
   F7              When there are both incoming and outgoing audio
                   streams, echo cancellation must be made
                   available to avoid disturbing echo during
                   conversation.
   ----------------------------------------------------------------
   F8              The browser must support synchronization of
                   audio and video.
   ----------------------------------------------------------------
   F9              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.
   ----------------------------------------------------------------
   F10             The browser must support a baseline audio and
                   video codec.
   ----------------------------------------------------------------
   F11             It must be possible to protect streams and data
                   from wiretapping [RFC2804] [RFC7258].
   ----------------------------------------------------------------
   F12             The browser must enable verification, given
                   the right circumstances and by use of other
                   trusted communication, that streams and
                   data received have not been manipulated by
                   any party.
   ----------------------------------------------------------------
   F13             The browser must encrypt, authenticate, and
                   integrity protect media and data on a
                   per-IP-packet basis, and it must drop incoming media
                   and data packets that fail the per-IP-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]).
   ----------------------------------------------------------------
   F14             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.
   ----------------------------------------------------------------
   F15             The browser must be able to collect statistics,
                   related to the transport of audio and video
                   between peers, needed to estimate quality of
                   experience.
   ----------------------------------------------------------------

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





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2.3.2.  Simple Video Communication Service: NAT/Firewall That Blocks UDP

2.3.2.1.  Description

   This use case is almost identical to the
   Simple Video Communication Service use case (Section 2.3.1).  The
   difference is that one of the users is behind a NAT/firewall that
   blocks UDP traffic.

2.3.2.2.  Additional Requirements

   ----------------------------------------------------------------
   REQ-ID          DESCRIPTION
   ----------------------------------------------------------------
   F18             The browser must be able to send streams and
                   data to a peer in the presence of NATs and
                   firewalls that block UDP traffic.
   ----------------------------------------------------------------

2.3.3.  Simple Video Communication Service: Firewall That Only Allows
        Traffic via an HTTP Proxy

2.3.3.1.  Description

   This use case is almost identical to the
   Simple Video Communication Service use case (Section 2.3.1).  The
   difference is that one of the users is behind a firewall that only
   allows traffic via an HTTP Proxy.

2.3.3.2.  Additional Requirements

   ----------------------------------------------------------------
   REQ-ID          DESCRIPTION
   ----------------------------------------------------------------
   F21             The browser must be able to send streams and
                   data to a peer in the presence of firewalls that only
                   allow traffic via an HTTP Proxy, when firewall policy
                   allows WebRTC traffic.
   ----------------------------------------------------------------

2.3.4.  Simple Video Communication Service: Global Service Provider

2.3.4.1.  Description

   This use case is almost identical to the
   Simple Video Communication Service use case (Section 2.3.1).  What is
   added is that the service provider is operating over large
   geographical areas (or even globally).



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   Assuming that the Interactive Connectivity Establishment (ICE)
   mechanism [RFC5245] will be used, this means that the service
   provider would like to be able to provide several Session Traversal
   Utilities for NAT (STUN) and Traversal Using Relay NAT (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.

   Note that ICE support being mandatory does not preclude a WebRTC
   endpoint from supporting more traversal mechanisms than ICE using
   STUN and TURN.

2.3.4.2.  Additional Requirements

   ----------------------------------------------------------------
   REQ-ID          DESCRIPTION
   ----------------------------------------------------------------
   F19             The browser must be able to use several STUN
                   and TURN servers.
   ----------------------------------------------------------------

   A22

2.3.5.  Simple Video Communication Service: Enterprise Aspects

2.3.5.1.  Description

   This use case is similar to the Simple Video Communication Service
   use case (Section 2.3.1).

   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 WebRTC-based web application for
   communication desires to audit all WebRTC-based application sessions
   used from inside the company towards any external peer.  To be able
   to do this, they deploy a TURN server that straddles the boundary
   between the internal and the external network.

   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 WebRTC applications provided by an
   external service provider, they still want 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 WebRTC client to
   determine its server reflexive address on the internal side.  Thus,
   enabling cases where peers are both on the internal side to connect



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   without the traffic leaving the internal network.  It must be
   possible 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 WebRTC functionality will
   need to utilize both network specific STUN and TURN resources and
   STUN and TURN servers provisioned by the web application.

2.3.5.2.  Additional Requirements

   ----------------------------------------------------------------
   REQ-ID          DESCRIPTION
   ----------------------------------------------------------------
   F20             The browser must support the use of STUN and TURN
                   servers that are supplied by entities other than
                   the web application (i.e., the network provider).
   ----------------------------------------------------------------

2.3.6.  Simple Video Communication Service: Access Change

2.3.6.1.  Description

   This use case is almost identical to the
   Simple Video Communication Service use case (Section 2.3.1).  The
   difference is that the user changes network access during the
   session.

   The communication device used by one of the users has several network
   adapters (Ethernet, Wi-Fi, 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 Wi-Fi when the Ethernet cable is removed and then
   moves to cellular access to the Internet when moving out of Wi-Fi
   coverage.  The session continues even though the access method
   changes.

2.3.6.2.  Additional Requirements

   ----------------------------------------------------------------
   REQ-ID          DESCRIPTION
   ----------------------------------------------------------------
   F17             The communication session must survive across a
                   change of the network interface used by the
                   session.
   ----------------------------------------------------------------







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2.3.7.  Simple Video Communication Service: QoS

2.3.7.1.  Description

   This use case is almost identical to the
   Simple Video Communication Service: Access Change use case
   (Section 2.3.6).  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 differentiation 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.

2.3.7.2.  Additional Requirements

   ----------------------------------------------------------------
   REQ-ID          DESCRIPTION
   ----------------------------------------------------------------
   F17             The communication session must survive across a
                   change of the network interface used by the
                   session.
   ----------------------------------------------------------------
   F22             The browser should be able to take advantage
                   of available capabilities (supplied by network
                   nodes) to differentiate voice, video, and data
                   appropriately.
   ----------------------------------------------------------------

2.3.8.  Simple Video Communication Service with Screen Sharing

2.3.8.1.  Description

   This use case has the audio and video communication of the
   Simple Video Communication Service use case (Section 2.3.1).

   However, 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 application displays with the peer.










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2.3.8.2.  Additional Requirements

   ----------------------------------------------------------------
   REQ-ID          DESCRIPTION
   ----------------------------------------------------------------
   F36             The browser must be able to generate streams
                   using the entire user display, a specific area
                   of the user display, or the information being
                   displayed by a specific application.
   ----------------------------------------------------------------

   A21

2.3.9.  Simple Video Communication Service with File Exchange

2.3.9.1.  Description

   This use case has the audio and video communication of the
   Simple Video Communication Service use case (Section 3.3.1).

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

2.3.9.2.  Additional Requirements

   ----------------------------------------------------------------
   REQ-ID          DESCRIPTION
   ----------------------------------------------------------------
   F35             The browser must be able to send reliable
                   data traffic to a peer browser.
   ----------------------------------------------------------------

   A21, A24

2.3.10.  Hockey Game Viewer

2.3.10.1.  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.






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   Before the game starts, and during game breaks, the talent scout and
   the manager have a 1:1 audiovisual communication session.  On the
   mobile phone, only the camera facing the talent scout is used.  On
   the user 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 ongoing, 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
   picture-in-picture thumbnails of the rear-facing camera (self view)
   and the desktop camera.  Because the most important stream in this
   phase is the video showing the game, the application used in the
   talent scout's mobile phone sets higher priority for that stream.

2.3.10.2.  Additional Requirements

   ----------------------------------------------------------------
   REQ-ID          DESCRIPTION
   ----------------------------------------------------------------
   F22             The browser should be able to take advantage
                   of available capabilities (supplied by network
                   nodes) to differentiate voice, video, and data
                   appropriately.
   ----------------------------------------------------------------
   F25             The browser must be able to render several
                   concurrent audio and video streams.
   ----------------------------------------------------------------

   A17, A23














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2.3.11.  Multiparty Video Communication

2.3.11.1.  Description

   In this use case, the Simple Video Communication Service use case
   (Section 2.3.1) 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 the user experience, the web application renders
   the audio coming from different participants so that it is
   experienced to come from different spatial locations.  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 user experience is the
   highlighting of the video window that displays the video of the
   currently speaking peer.

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

   Note: What this use case adds in terms of requirements are
   capabilities to send streams to and receive streams from several
   peers concurrently as well as the capabilities to render the video
   from all received streams and be able to spatialize, level adjust,
   and mix the audio from all received streams locally in the browser.
   It also adds the capability to measure the audio level/activity.




















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2.3.11.2.  Additional Requirements

   ----------------------------------------------------------------
   REQ-ID          DESCRIPTION
   ----------------------------------------------------------------
   F23             The browser must be able to transmit streams and
                   data to several peers concurrently.
   ----------------------------------------------------------------
   F24             The browser must be able to receive streams and
                   data from multiple peers concurrently.
   ----------------------------------------------------------------
   F25             The browser must be able to render several
                   concurrent audio and video streams.
   ----------------------------------------------------------------
   F26             The browser must be able to mix several
                   audio streams.
   ----------------------------------------------------------------
   F27             The browser must be able to apply spatialization
                   effects to audio streams.
   ----------------------------------------------------------------
   F28             The browser must be able to measure the
                   voice activity level in audio streams.
   ----------------------------------------------------------------
   F29             The browser must be able to change the
                   voice activity level in audio streams.
   ----------------------------------------------------------------

   A13, A14, A15, A16

2.3.12.  Multiparty Online Game with Voice Communication

2.3.12.1.  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 online 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 are required, and they have higher
   priority than the voice.

   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





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   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.

2.3.12.2.  Additional Requirements

   ----------------------------------------------------------------
   REQ-ID          DESCRIPTION
   ----------------------------------------------------------------
   F22             The browser should be able to take advantage
                   of available capabilities (supplied by network
                   nodes) to differentiate voice, video, and data
                   appropriately.
   ----------------------------------------------------------------
   F23             The browser must be able to transmit streams and
                   data to several peers concurrently.
   ----------------------------------------------------------------
   F24             The browser must be able to receive streams and
                   data from multiple peers concurrently.
   ----------------------------------------------------------------
   F25             The browser must be able to render several
                   concurrent audio and video streams.
   ----------------------------------------------------------------
   F26             The browser must be able to mix several
                   audio streams.
   ----------------------------------------------------------------
   F27             The browser must be able to apply spatialization
                   effects when playing audio streams.
   ----------------------------------------------------------------
   F28             The browser must be able to measure the
                   voice activity level in audio streams.
   ----------------------------------------------------------------
   F29             The browser must be able to change the
                   voice activity level in audio streams.
   ----------------------------------------------------------------
   F30             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).
   ----------------------------------------------------------------
   F34             The browser must be able to send short
                   latency unreliable datagram traffic to a
                   peer browser [RFC5405].
   ----------------------------------------------------------------

   A13, A14, A15, A16, A17, A18, A23





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2.4.  Browser - GW/Server Use Cases

2.4.1.  Telephony Terminal

2.4.1.1.  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.

   Note: "place and receive calls in the same way as when using a normal
   mobile phone" means that you can dial a number and your mobile
   telephony operator has made available your phone contacts online so
   that they are available and can be clicked to call and they can 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.  For people receiving calls made from the
   web browser, the usual identity (i.e., the phone number of the mobile
   phone) will be presented.

2.4.1.2.  Additional Requirements

   ----------------------------------------------------------------
   REQ-ID          DESCRIPTION
   ----------------------------------------------------------------
   F31             The browser must support an audio media format
                   (codec) that is commonly supported by existing
                   telephony services.
   ----------------------------------------------------------------
   F33             The browser must be able to initiate and
                   accept a media session where the data needed
                   for establishment can be carried in SIP.
   ----------------------------------------------------------------

2.4.2.  FedEx Call

2.4.2.1.  Description

   Alice uses her web browser with a service that allows her to call
   Public Switched Telephone Network (PSTN) numbers.  Alice calls
   1-800-123-4567.  Alice should be able to hear the initial prompts
   from the FedEx Interactive Voice Responder (IVR), and when the IVR
   says press 1, there should be a way for Alice to navigate the IVR.





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2.4.2.2.  Additional Requirements

   ----------------------------------------------------------------
   REQ-ID          DESCRIPTION
   ----------------------------------------------------------------
   F31             The browser must support an audio media format
                   (codec) that is commonly supported by existing
                   telephony services.
   ----------------------------------------------------------------
   F32             There should be a way to navigate
                   a dual-tone multi-frequency signaling (DTMF)
                   based Interactive Voice Response (IVR) system.
   ----------------------------------------------------------------

2.4.3.  Video Conferencing System with Central Server

2.4.3.1.  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 sends 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 that 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.

   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 email
   addresses or login IDs.

   Note: This use case adds requirements on support for fast stream
   switches (F16).  There exist several solutions that enable the server
   to forward one high-resolution and several low-resolution video



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   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.

2.4.3.2.  Additional Requirements

  ----------------------------------------------------------------
  REQ-ID          DESCRIPTION
  ----------------------------------------------------------------
  F16             The browser must support insertion of reference frames
                  in outgoing media streams when requested by a peer.
  ----------------------------------------------------------------
  F25             The browser must be able to render several
                  concurrent audio and video streams.
  ----------------------------------------------------------------

3.  Requirements Summary

3.1.  General

   This section contains the requirements on the browser derived from
   the use cases in Section 2.

   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.

3.2.  Browser Requirements

  ----------------------------------------------------------------
  Common, basic 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.







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  ----------------------------------------------------------------
  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.
  ----------------------------------------------------------------
  F6              The browser must detect when a stream from a
                  peer is not received anymore.
  ----------------------------------------------------------------
  F7              When there are both incoming and outgoing audio
                  streams, echo cancellation must be made
                  available to avoid disturbing echo during
                  conversation.
  ----------------------------------------------------------------
  F8              The browser must support synchronization of
                  audio and video.
  ----------------------------------------------------------------
  F9              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
  ----------------------------------------------------------------
  F10             The browser must support a baseline audio and
                  video codec.
  ----------------------------------------------------------------
  F11             It must be possible to protect streams and data
                  from wiretapping [RFC2804] [RFC7258].
  ----------------------------------------------------------------
  F12             The browser must enable verification, given
                  the right circumstances and by use of other
                  trusted communication, that streams and
                  data received have not been manipulated by
                  any party.










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  ----------------------------------------------------------------
  F13             The browser must encrypt, authenticate, and
                  integrity protect media and data on a
                  per-IP-packet basis, and it must drop incoming media
                  and data packets that fail the per-IP-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]).
  ----------------------------------------------------------------
  F14             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.
  ----------------------------------------------------------------
  F15             The browser must be able to collect statistics,
                  related to the transport of audio and video
                  between peers, needed to estimate quality of
                  experience.
  ----------------------------------------------------------------
  Requirements related to network and topology
  ----------------------------------------------------------------
  REQ-ID          DESCRIPTION
  ----------------------------------------------------------------
  F16             The browser must support insertion of reference frames
                  in outgoing media streams when requested by a peer.
  ----------------------------------------------------------------
  F17             The communication session must survive across a
                  change of the network interface used by the
                  session.
  ----------------------------------------------------------------
  F18             The browser must be able to send streams and
                  data to a peer in the presence of NATs and
                  firewalls that block UDP traffic.
  ----------------------------------------------------------------
  F19             The browser must be able to use several STUN
                  and TURN servers.
  ----------------------------------------------------------------
  F20             The browser must support the use of STUN and TURN
                  servers that are supplied by entities other than
                  the web application (i.e., the network provider).
  ----------------------------------------------------------------
  F21             The browser must be able to send streams and
                  data to a peer in the presence of firewalls that only
                  allow traffic via an HTTP Proxy, when firewall policy
                  allows WebRTC traffic.






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  ----------------------------------------------------------------
  F22             The browser should be able to take advantage
                  of available capabilities (supplied by network
                  nodes) to differentiate voice, video, and data
                  appropriately.
  ----------------------------------------------------------------
  Requirements related to multiple peers and streams
  ----------------------------------------------------------------
  REQ-ID          DESCRIPTION
  ----------------------------------------------------------------
  F23             The browser must be able to transmit streams and
                  data to several peers concurrently.
  ----------------------------------------------------------------
  F24             The browser must be able to receive streams and
                  data from multiple peers concurrently.
  ----------------------------------------------------------------
  F25             The browser must be able to render several
                  concurrent audio and video streams.
  ----------------------------------------------------------------
  F26             The browser must be able to mix several
                  audio streams.
  ----------------------------------------------------------------
  Requirements related to audio processing
  ----------------------------------------------------------------
  REQ-ID          DESCRIPTION
  ----------------------------------------------------------------
  F27             The browser must be able to apply spatialization
                  effects when playing audio streams.
  ----------------------------------------------------------------
  F28             The browser must be able to measure the
                  voice activity level in audio streams.
  ----------------------------------------------------------------
  F29             The browser must be able to change the
                  voice activity level in audio streams.
  ----------------------------------------------------------------
  F30             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).
  ----------------------------------------------------------------
  Requirements related to legacy interop
  ----------------------------------------------------------------
  REQ-ID          DESCRIPTION
  ----------------------------------------------------------------
  F31             The browser must support an audio media format
                  (codec) that is commonly supported by existing
                  telephony services.




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  ----------------------------------------------------------------
  F32             There should be a way to navigate
                  a dual-tone multi-frequency signaling (DTMF)
                  based Interactive Voice Response (IVR) system.
  ----------------------------------------------------------------
  F33             The browser must be able to initiate and
                  accept a media session where the data needed
                  for establishment can be carried in SIP.
  ----------------------------------------------------------------
  Other requirements
  ----------------------------------------------------------------
  REQ-ID          DESCRIPTION
  ----------------------------------------------------------------
  F34             The browser must be able to send short
                  latency unreliable datagram traffic to a
                  peer browser [RFC5405].
  ----------------------------------------------------------------
  F35             The browser must be able to send reliable
                  data traffic to a peer browser.
  ----------------------------------------------------------------
  F36             The browser must be able to generate streams
                  using the entire user display, a specific area
                  of the user display or the information being
                  displayed by a specific application.
  ----------------------------------------------------------------

4.  Security Considerations

4.1.  Introduction

   A malicious web application might use the browser to perform Denial-
   of-Service (DoS) attacks on NAT infrastructure, or on peer devices.
   For example, a malicious web application might leak TURN credentials
   to unauthorized parties, allowing them to consume the TURN server's
   bandwidth.  To address this risk, web applications should be prepared
   to revoke TURN credentials and issue new ones.  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.










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4.2.  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 must 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.

   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 must provide mechanisms for users to revise and even
   completely revoke consent to use the screen, part thereof, or an
   application 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 allow the users to
   verify that the streams received have not be manipulated (F12).

   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 DoS by other entities.

4.3.  Web Application Considerations

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








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5.  Normative References

   [RFC2804]  IAB and , "IETF Policy on Wiretapping", RFC 2804, May
              2000, <http://www.rfc-editor.org/info/rfc2804>.

   [RFC5245]  Rosenberg, J., "Interactive Connectivity Establishment
              (ICE): A Protocol for Network Address Translator (NAT)
              Traversal for Offer/Answer Protocols", RFC 5245, April
              2010, <http://www.rfc-editor.org/info/rfc5245>.

   [RFC5405]  Eggert, L. and G. Fairhurst, "Unicast UDP Usage Guidelines
              for Application Designers", BCP 145, RFC 5405, November
              2008, <http://www.rfc-editor.org/info/rfc5405>.

   [RFC5479]  Wing, D., Ed., Fries, S., Tschofenig, H., and F. Audet,
              "Requirements and Analysis of Media Security Management
              Protocols", RFC 5479, April 2009,
              <http://www.rfc-editor.org/info/rfc5479>.

   [RFC7258]  Farrell, S. and H. Tschofenig, "Pervasive Monitoring Is an
              Attack", BCP 188, RFC 7258, May 2014,
              <http://www.rfc-editor.org/info/rfc7258>.





























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Appendix A.  API Requirements

   This section contains the requirements on the API derived from the
   use cases in Section 2.

   Note: As the W3C is responsible for the API, the API requirements in
   this specification are not normative.

   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
                   and to have access to the local file system.
   ----------------------------------------------------------------
   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 the sending of a
                   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 or not
                   the establishment of a stream with a peer was
                   successful.






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   ----------------------------------------------------------------
   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.
   ----------------------------------------------------------------
   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 the 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.






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   ----------------------------------------------------------------
   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
                   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 use 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 the quality of service.
   ----------------------------------------------------------------











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Acknowledgements

   The authors wish to thank 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, Stephan Wenger, Harald
   Alvestrand, Cullen Jennings, Andrew Hutton and everyone else in the
   RTCWEB community that have provided comments, feedback, text and
   improvement proposals on the document.  A big thank you to everyone
   that provided comments as part of the IESG evaluation and to everyone
   else that provided comments and input in order to improve the
   document.

Authors' Addresses

   Christer Holmberg
   Ericsson
   Hirsalantie 11
   Jorvas  02420
   Finland

   EMail: christer.holmberg@ericsson.com


   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











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