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
----------------------------------------------------------------
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.
Holmberg, et al. Informational [Page 6]RFC 7478 WebRTC March 2015
----------------------------------------------------------------
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
----------------------------------------------------------------
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).
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
----------------------------------------------------------------
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
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
----------------------------------------------------------------
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.
----------------------------------------------------------------
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
----------------------------------------------------------------
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.
Holmberg, et al. Informational [Page 11]RFC 7478 WebRTC March 2015
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.
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
----------------------------------------------------------------
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
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
----------------------------------------------------------------
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
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
----------------------------------------------------------------
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
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
----------------------------------------------------------------
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.
Holmberg, et al. Informational [Page 17]RFC 7478 WebRTC March 2015
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
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
----------------------------------------------------------------
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.
Holmberg, et al. Informational [Page 19]RFC 7478 WebRTC March 2015
----------------------------------------------------------------
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
----------------------------------------------------------------
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
----------------------------------------------------------------
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
----------------------------------------------------------------
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
----------------------------------------------------------------
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
----------------------------------------------------------------
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]
