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rfc:rfc7478

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.

Holmberg, et al. Informational [Page 1] RFC 7478 WebRTC March 2015

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.

Holmberg, et al. Informational [Page 2] RFC 7478 WebRTC March 2015

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

Holmberg, et al. Informational [Page 3] RFC 7478 WebRTC March 2015

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

Holmberg, et al. Informational [Page 4] RFC 7478 WebRTC March 2015

 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.

Holmberg, et al. Informational [Page 5] RFC 7478 WebRTC March 2015

 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

  1. —————————————————————

REQ-ID DESCRIPTION

  1. —————————————————————

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.

Holmberg, et al. Informational [Page 6] RFC 7478 WebRTC March 2015

  1. —————————————————————

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

Holmberg, et al. Informational [Page 7] RFC 7478 WebRTC March 2015

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

  1. —————————————————————

REQ-ID DESCRIPTION

  1. —————————————————————

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

  1. —————————————————————

REQ-ID DESCRIPTION

  1. —————————————————————

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).

Holmberg, et al. Informational [Page 8] RFC 7478 WebRTC March 2015

 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

  1. —————————————————————

REQ-ID DESCRIPTION

  1. —————————————————————

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

Holmberg, et al. Informational [Page 9] RFC 7478 WebRTC March 2015

 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

  1. —————————————————————

REQ-ID DESCRIPTION

  1. —————————————————————

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

  1. —————————————————————

REQ-ID DESCRIPTION

  1. —————————————————————

F17 The communication session must survive across a

                 change of the network interface used by the
                 session.
 ----------------------------------------------------------------

Holmberg, et al. Informational [Page 10] RFC 7478 WebRTC March 2015

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

  1. —————————————————————

REQ-ID DESCRIPTION

  1. —————————————————————

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.

Holmberg, et al. Informational [Page 11] RFC 7478 WebRTC March 2015

2.3.8.2. Additional Requirements

  1. —————————————————————

REQ-ID DESCRIPTION

  1. —————————————————————

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

  1. —————————————————————

REQ-ID DESCRIPTION

  1. —————————————————————

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.

Holmberg, et al. Informational [Page 12] RFC 7478 WebRTC March 2015

 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

  1. —————————————————————

REQ-ID DESCRIPTION

  1. —————————————————————

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

Holmberg, et al. Informational [Page 13] RFC 7478 WebRTC March 2015

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.

Holmberg, et al. Informational [Page 14] RFC 7478 WebRTC March 2015

2.3.11.2. Additional Requirements

  1. —————————————————————

REQ-ID DESCRIPTION

  1. —————————————————————

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

Holmberg, et al. Informational [Page 15] RFC 7478 WebRTC March 2015

 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

  1. —————————————————————

REQ-ID DESCRIPTION

  1. —————————————————————

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

Holmberg, et al. Informational [Page 16] RFC 7478 WebRTC March 2015

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

  1. —————————————————————

REQ-ID DESCRIPTION

  1. —————————————————————

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.

Holmberg, et al. Informational [Page 17] RFC 7478 WebRTC March 2015

2.4.2.2. Additional Requirements

  1. —————————————————————

REQ-ID DESCRIPTION

  1. —————————————————————

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

Holmberg, et al. Informational [Page 18] RFC 7478 WebRTC March 2015

 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

  1. —————————————————————

REQ-ID DESCRIPTION

  1. —————————————————————

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

  1. —————————————————————

Common, basic requirements

  1. —————————————————————

REQ-ID DESCRIPTION

  1. —————————————————————

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.

Holmberg, et al. Informational [Page 19] RFC 7478 WebRTC March 2015

  1. —————————————————————

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.

Holmberg, et al. Informational [Page 20] RFC 7478 WebRTC March 2015

  1. —————————————————————

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.

Holmberg, et al. Informational [Page 21] RFC 7478 WebRTC March 2015

  1. —————————————————————

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.

Holmberg, et al. Informational [Page 22] RFC 7478 WebRTC March 2015

  1. —————————————————————

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.

Holmberg, et al. Informational [Page 23] RFC 7478 WebRTC March 2015

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.

Holmberg, et al. Informational [Page 24] RFC 7478 WebRTC March 2015

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>.

Holmberg, et al. Informational [Page 25] RFC 7478 WebRTC March 2015

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.

Holmberg, et al. Informational [Page 26] RFC 7478 WebRTC March 2015

  1. —————————————————————

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.

Holmberg, et al. Informational [Page 27] RFC 7478 WebRTC March 2015

  1. —————————————————————

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.
 ----------------------------------------------------------------

Holmberg, et al. Informational [Page 28] RFC 7478 WebRTC March 2015

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

Holmberg, et al. Informational [Page 29]

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