One document matched: draft-ietf-rtcweb-use-cases-and-requirements-11.xml
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docName="draft-ietf-rtcweb-use-cases-and-requirements-11.txt"
ipr="trust200902" obsoletes="" submissionType="IETF" updates=""
xml:lang="en">
<front>
<title abbrev="RTC-Web">Web Real-Time Communication Use-cases and
Requirements</title>
<author fullname="Christer Holmberg" initials="C.H." surname="Holmberg">
<organization>Ericsson</organization>
<address>
<postal>
<street>Hirsalantie 11</street>
<code>02420</code>
<city>Jorvas</city>
<country>Finland</country>
</postal>
<email>christer.holmberg@ericsson.com</email>
</address>
</author>
<author fullname="Stefan Hakansson" initials="S.H." surname="Hakansson">
<organization>Ericsson</organization>
<address>
<postal>
<street>Laboratoriegrand 11</street>
<code>97128</code>
<city>Lulea</city>
<country>Sweden</country>
</postal>
<email>stefan.lk.hakansson@ericsson.com</email>
</address>
</author>
<author fullname="Goran AP Eriksson" initials="G.E." surname="Eriksson">
<organization>Ericsson</organization>
<address>
<postal>
<street>Farogatan 6</street>
<code>16480</code>
<city>Stockholm</city>
<country>Sweden</country>
</postal>
<email>goran.ap.eriksson@ericsson.com</email>
</address>
</author>
<date year="2013"
month="June"
day="13"/>
<area>Transport</area>
<workgroup>RTCWEB Working Group</workgroup>
<keyword>browser</keyword>
<keyword>websocket</keyword>
<keyword>real-time</keyword>
<abstract>
<t>This document describes web based real-time communication use-cases.
Requirements on the browser functionality are derived from use-cases.</t>
</abstract>
</front>
<middle>
<section title="Introduction" toc="default">
<t>This document presents a few use-cases of web applications that are
executed in a browser and use real-time communication capabilities.
In most of the use-cases all end-user clients are web applications,
but there are some use-cases where at least one of the end-user client
is of another type (e.g. a telephone).</t>
<t>Based on the use-cases, the document derives requirements related to
browser functionality. These requirements are named "Fn", where n is an integer,
and are described in <xref target="browser_reqs"></xref>.</t>
<t>This document was developed in an initial phase of the work with rather minor updates at later stages.
It has not really served as a tool in deciding features or scope for the WGs efforts so far.
It is proposed to be used in a later phase to evaluate the protocols and
solutions developed by the WG.</t>
<t>This document also lists requirements
related to the API to be used by web applications as an appendix. The reason is that
the W3C WebRTC WG has decided to not develop its own use-case/requirement document,
but instead use this document.
These requirements are named "An", where n is an integer, and are described
in <xref target="api_reqs"></xref>-</t>
<t>The document focuses on requirements related to real-time media
streams and data exchange. Requirements related to privacy, signalling
between the browser and web server etc. are currently not considered.</t>
</section>
<section title="Definitions" toc="default">
<t>TBD</t>
</section>
<section anchor="use_cases" title="Use-cases" toc="default">
<section title="Introduction" toc="default">
<t>This section describes web based real-time communication use-cases,
from which requirements are derived.</t>
<t>The following considerations are applicable to all use cases:<list
style="symbols">
<t>Clients can be on IPv4-only</t>
<t>Clients can be on IPv6-only</t>
<t>Clients can be on dual-stack</t>
<t>Clients can be on wideband (10s of Mbits/sec)</t>
<t>Clients can be on narrowband (10s to 100s of Kbits/sec)</t>
<t>Clients can be on variable-media-quality networks
(wireless)</t>
<t>Clients can be on congested networks</t>
<t>Clients can be on firewalled networks with no UDP allowed</t>
<t>Clients can be on networks with any type (as described
in RFC4787) of NAT.</t>
</list></t>
</section>
<section title="Browser-to-browser use-cases" toc="default">
<section anchor="simple-video-comm-service"
title="Simple Video Communication Service" toc="default">
<section title="Description" toc="default">
<t>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.</t>
<t>During session establishment a self-view is displayed, and once
the session has been established the video sent from the remote
peer is displayed in addition to the self-view. During the
session, each user can select to remove and re-insert the
self-view as often as desired. Each user can also change the sizes
of his/her two video displays during the session. Each user can
also pause sending of media (audio, video, or both) and mute
incoming media</t>
<t>It is essential that the communication cannot be
wiretapped <xref format="default" pageno="false" target="RFC2804"></xref>.</t>
<t> It is essential that media and data be encrypted,
authenticated and integrity protected on a per-packet basis
and that media and data packets failing the integrity check
not be delivered to the application.</t>
<t>In addition, it is required that browsers enable the media
and data security keys to be cryptographically bound to the user
identity.</t>
<t>The application gives the users the opportunity to stop it from
exposing the host IP address to the application of the other user.</t>
<t>Any session participant can end the session at any time.</t>
<t>The two users may be using communication devices of different
makes, with different operating systems and browsers from
different vendors.</t>
<t>One user has an unreliable Internet connection. It sometimes
loses packets, and sometimes goes down completely.</t>
<t>One user is located behind a Network Address Translator
(NAT).</t>
<t>The web service monitors the quality of the service (focus on
quality of audio and video) the end-users experience.</t>
</section>
<section title="Derived Requirements" toc="default">
<t>F1, F2, F3, F4, F5, F8, F9, F10, F20, F25, F28, F35, F36, F38, F39</t>
<t>A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12, A25, A26</t>
</section>
</section>
<section title="Simple Video Communication Service, NAT/FW that blocks UDP"
toc="default">
<section title="Description" toc="default">
<t>This use-case is almost identical to the <xref format="title"
pageno="false" target="simple-video-comm-service"></xref> use-case
(<xref target="simple-video-comm-service"></xref>). The difference
is that one of the users is behind a NAT that blocks UDP
traffic.</t>
</section>
<section title="Derived Requirements" toc="default">
<t>F1, F2, F3, F4, F5, F8, F9, F10, F20, F25, F28, F29</t>
<t>A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12</t>
</section>
</section>
<section title="Simple Video Communication Service, FW that only allows http"
toc="default">
<section title="Description" toc="default">
<t>This use-case is almost identical to the <xref format="title"
pageno="false" target="simple-video-comm-service"></xref> use-case
(<xref target="simple-video-comm-service"></xref>). The difference
is that one of the users is behind a FW that only allows http
traffic.</t>
</section>
<section title="Derived Requirements" toc="default">
<t>F1, F2, F3, F4, F5, F8, F9, F10, F20, F25, F28, F37</t>
<t>A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12</t>
</section>
</section>
<section title="Simple Video Communication Service, global service provider"
toc="default">
<section title="Description" toc="default">
<t>This use-case is almost identical to the <xref format="title"
pageno="false" target="simple-video-comm-service"></xref> use-case
(<xref target="simple-video-comm-service"></xref>).</t>
<t>What is added is that the service provider is operating over
large geographical areas (or even globally).</t>
<t>Assuming that ICE will be used, this means that the service
provider would like to be able to provide several STUN and TURN
servers (via the app) to the browser; selection of which one(s) to
use is part of the ICE processing. Other reasons for wanting to
provide several STUN and TURN servers include support for IPv4 and
IPv6, load balancing and redundancy.</t>
</section>
<section title="Derived Requirements" toc="default">
<t>F1, F2, F3, F4, F5, F8, F9, F10, F20, F25, F28, F31</t>
<t>A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12, A22</t>
</section>
</section>
<section title="Simple Video Communication Service, enterprise aspects"
toc="default">
<section title="Description" toc="default">
<t>This use-case is similar to the <xref format="title"
pageno="false" target="simple-video-comm-service"></xref> use-case
(<xref target="simple-video-comm-service"></xref>).</t>
<t>What is added is aspects when using the service in enterprises.
ICE is assumed in the further description of this use-case.</t>
<t>An enterprise that uses a RTCWEB based web application for
communication desires to audit all RTCWEB based application
session used from inside the company towards any external peer. To
be able to do this they deploy a TURN server that straddle the
boundary between the internal network and the external. </t>
<t>The firewall will block all attempts to use STUN with an
external destination unless they go to the enterprise auditing
TURN server. In cases where employees are using RTCWEB
applications provided by an external service provider they still
want to have the traffic to stay inside their internal network and
in addition not load the straddling TURN server, thus they deploy
a STUN server allowing the RTCWEB client to determine its server
reflexive address on the internal side. Thus enabling cases where
peers are both on the internal side to connect without the traffic
leaving the internal network. It must be possibele to configure
the browsers used in the enterprise with network specific STUN and
TURN servers. This should be possible to achieve by
autoconfiguration methods. The RTCWEB functionality will need to
utilize both network specific STUN and TURN resources and STUN and
TURN servers provisioned by the web application.</t>
</section>
<section title="Derived Requirements" toc="default">
<t>F1, F2, F3, F4, F5, F8, F9, F10, F20, F25, F28, F32</t>
<t>A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12</t>
</section>
</section>
<section anchor="simple-video-comm-service-access-change"
title="Simple Video Communication Service, access change"
toc="default">
<section title="Description" toc="default">
<t>This use-case is almost identical to the <xref format="title"
pageno="false" target="simple-video-comm-service"></xref> use-case
(<xref target="simple-video-comm-service"></xref>). The difference
is that the user changes network access during the session:</t>
<t>The communication device used by one of the users have several
network adapters (Ethernet, WiFi, Cellular). The communication
device is accessing the Internet using Ethernet, but the user has
to start a trip during the session. The communication device
automatically changes to use WiFi when the Ethernet cable is
removed and then moves to cellular access to the Internet when
moving out of WiFi coverage. The session continues even though the
access method changes.</t>
</section>
<section title="Derived Requirements" toc="default">
<t>F1, F2, F3, F4, F5, F8, F9, F10, F20, F25, F26, F28</t>
<t>A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12</t>
</section>
</section>
<section title="Simple Video Communication Service, QoS" toc="include">
<section title="Description" toc="default">
<t>This use-case is almost identical to the <xref format="title"
pageno="false"
target="simple-video-comm-service-access-change"></xref> use-case
(<xref target="simple-video-comm-service-access-change"></xref>).
The use of Quality of Service (QoS) capabilities is added:</t>
<t>The user in the previous use case that starts a trip is behind
a common residential router that supports prioritization of
traffic. In addition, the user's provider of cellular access has
QoS support enabled. The user is able to take advantage of the QoS
support both when accessing via the residential router and when
using cellular.</t>
</section>
<section title="Derived Requirements" toc="default">
<t>F1, F2, F3, F4, F5, F8, F9, F10, F20, F24, F25, F26,
F28</t>
<t>A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12</t>
</section>
</section>
<section title="Simple Video Communication Service with sharing"
toc="default">
<section title="Description" toc="default">
<t>This use-case has the audio and video communication of the
<xref format="title" pageno="false"
target="simple-video-comm-service"></xref> use-case (<xref
target="simple-video-comm-service"></xref>).</t>
<t>But in addition to this, one of the users can share what is
being displayed on her/his screen with a peer. The user can choose
to share the entire screen, part of the screen (part selected by
the user) or what a selected applicaton displays with the
peer.</t>
</section>
<section title="Derived Requirements" toc="default">
<t>F1, F2, F3, F4, F5, F8, F9, F10, F20, F25, F28, F30</t>
<t>A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12, A21</t>
</section>
</section>
<section title="Simple Video Communication Service with file exchange"
toc="default">
<section title="Description" toc="default">
<t>This use-case has the audio and video communication of the
<xref format="title" pageno="false"
target="simple-video-comm-service"></xref> use-case (<xref
target="simple-video-comm-service"></xref>).</t>
<t>But in addition to this, the users can send and receive files
stored in the file system of the device used.</t>
</section>
<section title="Derived Requirements" toc="default">
<t>F1, F2, F3, F4, F5, F8, F9, F10, F20, F25, F28, F30, F33</t>
<t>A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12, A21, A24</t>
</section>
</section>
<section title="Simple video communication service with inter-operator calling"
toc="default">
<section title="Description">
<t>Two users have logged into two different web applications,
provided by different service providers.</t>
<t>The service providers are interconnected by some means, but
exchange no more information about the users than what can be
carried using SIP.</t>
<t>NOTE: More profiling of what this means may be needed.</t>
<t>For each user Alice who has authorized another user Bob to
receive login status information, Alice's service publishes
Alice's login status information to Bob. How this authorization is
defined and established is out of scope.</t>
<t>The same functionality as in the the <xref format="title"
pageno="false" target="simple-video-comm-service"></xref> use-case
(<xref target="simple-video-comm-service"></xref>) is
available.</t>
<t>The same issues with connectivity apply.</t>
</section>
<section title="Derived requirements">
<t>F1, F2, F3, F4, F5, F8, F9, F10, F20, F25, F27, F28</t>
<t>A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12, A20</t>
</section>
</section>
<section title="Hockey Game Viewer" toc="default">
<section title="Description" toc="default">
<t>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.</t>
<t>The club manager uses a desktop, equipped with one camera, for
viewing the game and discussing with the talent scout.</t>
<t>Before the game starts, and during game breaks, the talent
scout and the manager have a 1-1 audiovisual communication session. Only the
rear facing camera of the mobile phone is used. On the display of
the mobile phone, the video of the club manager is shown with a
picture-in-picture thumbnail of the rear facing camera
(self-view). On the display of the desktop, the video of the
talent scout is shown with a picture-in-picture thumbnail of the
desktop camera (self-view).</t>
<t>When the game is on-going, the talent scout activates the use
of the front facing camera, and that stream is sent to the desktop
(the stream from the rear facing camera continues to be sent all
the time). The video stream captured by the front facing camera
(that is capturing the game) of the mobile phone is shown in a big
window on the desktop screen, with picture-in-picture thumbnails
of the rear facing camera and the desktop camera (self-view). On
the display of the mobile phone the game is shown (front facing
camera) with picture-in-picture thumbnails of the rear facing
camera (self-view) and the desktop camera. As the most important
stream in this phase is the video showing the game, the application
used in the talent scout's mobile sets higher priority for that
stream.</t>
<t>It is essential that the communication cannot be
wiretapped <xref format="default" pageno="false" target="RFC2804"></xref>.</t>
</section>
<section title="Derived Requirements" toc="default">
<t>F1, F2, F3, F4, F5, F8, F9, F10, F17, F20, F34</t>
<t>A1, A2, A3, A4, A5, A7, A8, A9, A10, A11, A12, A17, A23</t>
</section>
</section>
<section title="Multiparty video communication" toc="default">
<section title="Description" toc="default">
<t>In this use-case is the <xref format="title" pageno="false"
target="simple-video-comm-service"></xref> use-case (<xref
target="simple-video-comm-service"></xref>) 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.</t>
<t>In order to enhance intelligibility, the web application pans
the audio from different participants differently when rendering
the audio. This is done automatically, but users can change how
the different participants are placed in the (virtual) room. In
addition the levels in the audio signals are adjusted before
mixing.</t>
<t>Another feature intended to enhance the use experience is that
the video window that displays the video of the currently speaking
peer is highlighted.</t>
<t>Each video stream received is by default displayed in a
thumbnail frame within the browser, but users can change the
display size.</t>
<t>It is essential that the communication cannot be
wiretapped <xref format="default" pageno="false" target="RFC2804"></xref>.</t>
<t>Note: What this use-case adds in terms of requirements is
capabilities to send streams to and receive streams from several
peers concurrently, as well as the capabilities to render the
video from all recevied streams and be able to spatialize, level
adjust and mix the audio from all received streams locally in the
browser. It also adds the capability to measure the audio
level/activity.</t>
</section>
<section title="Derived Requirements" toc="default">
<t>F1, F2, F3, F4, F5, F8, F9, F10, F11, F12, F13, F14, F15,
F16, F17, F20, F25</t>
<t>A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12, A13, A14,
A15, A16, A17</t>
</section>
</section>
<section title="Multiparty on-line game with voice communication"
toc="default">
<section title="Description" toc="default">
<t>This use case is based on the previous one. In this use-case,
the voice part of the multiparty video communication use case is
used in the context of an on-line game. The received voice audio
media is rendered together with game sound objects. For example,
the sound of a tank moving from left to right over the screen must
be rendered and played to the user together with the voice
media.</t>
<t>Quick updates of the game state is required, and have higher
priority than the voice.</t>
<t>It is essential that the communication cannot be
wiretapped <xref format="default" pageno="false" target="RFC2804"></xref>.</t>
<t>Note: the difference regarding local audio processing compared
to the "Multiparty video communication" use-case is that other
sound objects than the streams must be possible to be included in
the spatialization and mixing. "Other sound objects" could for
example be a file with the sound of the tank; that file could be
stored locally or remotely.</t>
</section>
<section title="Derived Requirements" toc="default">
<t>F1, F2, F3, F4, F5, F8, F9, F11, F12, F13, F14, F15, F16,
F18, F20, F23, F34</t>
<t>A1, A2, A3, A4, A5, A7, A8, A9, A10, A11, A12, A13, A14, A15,
A16, A17, A18, A23</t>
</section>
</section>
<section title="Distributed Music Band" toc="default">
<section title="Description" toc="default">
<t>In this use-case, a music band is playing music while the
members are at different physical locations. No central server is
used, instead all streams are set up in a mesh fashion.</t>
<t>Discussion: This use-case was briefly discussed at the Quebec
webrtc meeting and it got support. So far the only concrete
requirement (A17) derived is that the application must be able to
ask the browser to treat the audio signal as audio (in contrast to
speech). However, the use case should be further analysed to
determine other requirements (could be e.g. on delay
mic->speaker, level control of audio signals, etc.).</t>
</section>
<section title="Derived Requirements" toc="default">
<t>F1, F2, F3, F4, F5, F8, F9, F11, F12, F13, F14, F15,
F16</t>
<t>A1, A2, A3, A4, A5, A7, A8, A9, A10, A11, A12, A13, A14, A15,
A16, A19</t>
</section>
</section>
</section>
<section title="Browser - GW/Server use cases" toc="default">
<section title="Telephony terminal" toc="default">
<section title="Description" toc="default">
<t>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.</t>
<t>It is essential that the communication cannot be
wiretapped <xref format="default" pageno="false" target="RFC2804"></xref>.</t>
<t>Note: With "place and receive calls in the same way as when
using a normal mobile phone" it is meant that you can dial a
number, and that your mobile telephony operator has made available
your phone contacts on line, so they are available and can be
clicked to call, and be used to present the identity of an
incoming call. If the callee is not in your phone contacts the
number is displayed. Furthermore, your call logs are available,
and updated with the calls made/received from the browser. And for
people receiving calls made from the web browser the usual
identity (i.e. the phone number of the mobile phone) will be
presented.</t>
</section>
<section title="Derived Requirements" toc="default">
<t>F1, F2, F3, F4, F5, F8, F9, F10, F20, F21</t>
<t>A1, A2, A3, A4, A7, A8, A9, A10, A11, A12</t>
</section>
</section>
<section title="Fedex Call" toc="default">
<section title="Description" toc="default">
<t>Alice uses her web browser with a service that allows her
to call PSTN numbers. Alice calls 1-800-gofedex. Alice
should be able to hear the initial prompts from the fedex IVR and
when the IVR says press 1, there should be a way for Alice to
navigate the IVR.</t>
</section>
<section title="Derived Requirements" toc="default">
<t>F1, F2, F3, F4, F5, F8, F9, F10, F21, F22</t>
<t>A1, A2, A3, A4, A7, A8, A9, A10, A11, A12</t>
</section>
</section>
<section title="Video conferencing system with central server"
toc="default">
<section title="Description" toc="default">
<t>An organization uses a video communication system that supports
the establishment of multiparty video sessions using a central
conference server.</t>
<t>The browser of each participant send an audio stream (type in
terms of mono, stereo, 5.1, ... depending on the equipment of the
participant) to the central server. The central server mixes the
audio streams (and can in the mixing process naturally add effects
such as spatialization) and sends towards each participant a mixed
audio stream which is played to the user.</t>
<t>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.</t>
<t>The organization has an internal network set up with an
aggressive firewall handling access to the Internet. If users
cannot physically access the internal network, they can establish
a Virtual Private Network (VPN).</t>
<t>It is essential that the communication cannot be
wiretapped <xref format="default" pageno="false" target="RFC2804"></xref>.</t>
<t>All participants are authenticated by the central server, and
authorized to connect to the central server. The participants are
identified to each other by the central server, and the
participants do not have access to each others' credentials such
as e-mail addresses or login IDs.</t>
<t>Note: This use-case adds requirements on support for fast
stream switches F7, on encryption of media and on ability to
traverse very restrictive FWs. There exist several solutions that
enable the server to forward one high resolution and several low
resolution video streams: a) each browser could send a high
resolution, but scalable stream, and the server could send just
the base layer for the low resolution streams, b) each browser
could in a simulcast fashion send one high resolution and one low
resolution stream, and the server just selects or c) each browser
sends just a high resolution stream, the server transcodes into
low resolution streams as required.</t>
</section>
<section title="Derived Requirements" toc="default">
<t>F1, F2, F3, F4, F5, F7, F8, F9, F10, F17, F19, F20</t>
<t>A1, A2, A3, A4, A5, A7, A8, A9, A10, A11, A12, A17</t>
</section>
</section>
</section>
</section>
<section title="Requirements" toc="default">
<section title="General" toc="default">
<t>This section contains the requirements on the browser derived from the use-cases
in <xref target="use_cases"></xref>.</t>
<t>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.</t>
</section>
<section anchor="browser_reqs" title="Browser requirements"
toc="default">
<figure align="left" alt="" height="" suppress-title="false" title=""
width="">
<artwork align="left" alt="" height="" name="" type="" width=""
xml:space="preserve"><![CDATA[
REQ-ID DESCRIPTION
---------------------------------------------------------------
F1 The browser must be able to use microphones and
cameras as input devices to generate streams.
----------------------------------------------------------------
F2 The browser must be able to send streams and
data to a peer in the presence of NATs.
----------------------------------------------------------------
F3 Transmitted streams and data must be rate
controlled (meaning that the browser must, regardless
of application behavior, reduce send rate when
there is congestion).
----------------------------------------------------------------
F4 The browser must be able to receive, process and
render streams and data ("render" does not
apply for data) from peers.
----------------------------------------------------------------
F5 The browser should be able to render good quality
audio and video even in the presence of
reasonable levels of jitter and packet losses.
----------------------------------------------------------------
F7 The browser must support insertion of reference frames
in ougoing media streams when requested by a peer.
----------------------------------------------------------------
F8 The browser must detect when a stream from a
peer is not received anymore
----------------------------------------------------------------
F9 When there are both incoming and outgoing audio
streams, echo cancellation must be made
available to avoid disturbing echo during
conversation.
----------------------------------------------------------------
F10 The browser must support synchronization of
audio and video.
----------------------------------------------------------------
F11 The browser must be able to transmit streams and
data to several peers concurrently.
----------------------------------------------------------------
F12 The browser must be able to receive streams and
data from multiple peers concurrently.
----------------------------------------------------------------
F13 The browser must be able to apply spatialization
effects when playing audio streams.
----------------------------------------------------------------
F14 The browser must be able to measure the level
in audio streams.
----------------------------------------------------------------
F15 The browser must be able to change the level
in audio streams.
----------------------------------------------------------------
F16 The browser must be able to render several
concurrent video streams
----------------------------------------------------------------
F17 The browser must be able to mix several
audio streams.
----------------------------------------------------------------
F18 The browser must be able to process and mix
sound objects (media that is retrieved from
another source than the established media
stream(s) with the peer(s) with audio streams.
----------------------------------------------------------------
F19 Streams and data must be able to pass through
limited middleboxes.
----------------------------------------------------------------
F20 It must be possible to protect streams and data
from wiretapping [RFC2804].
----------------------------------------------------------------
F21 The browser must support an audio media format
(codec) that is commonly supported by existing
telephony services.
----------------------------------------------------------------
F22 There should be a way to navigate
a Dual-tone multi-frequency signaling (DTMF)
based Interactive voice response (IVR) System
----------------------------------------------------------------
F23 The browser must be able to send short
latency unreliable datagram traffic to a
peer browser [RFC5405].
----------------------------------------------------------------
F24 The browser should be able to take advantage
of available capabilities (supplied by network
nodes) to prioritize voice, video and data
appropriately.
----------------------------------------------------------------
F25 The browser should use encoding of streams
suitable for the current rendering (e.g.
video display size) and should change parameters
if the rendering changes during the session
----------------------------------------------------------------
F26 It must be possible to move from one network
interface to another one
----------------------------------------------------------------
F27 The browser must be able to initiate and
accept a media session where the data needed
for establishment can be carried in SIP.
----------------------------------------------------------------
F28 The browser must support a baseline audio and
video codec
----------------------------------------------------------------
F29 The browser must be able to send streams and
data to a peer in the presence of NATs that
block UDP traffic.
----------------------------------------------------------------
F30 The browser must be able to use the screen (or
a specific area of the screen) or what a certain
application displays on the screen to generate
streams.
----------------------------------------------------------------
F31 The browser must be able to use several STUN
and TURN servers
----------------------------------------------------------------
F32 There browser must support that STUN and TURN
servers to use are supplied by other entities
than via the web application (i.e. the network
provider).
----------------------------------------------------------------
F33 The browser must be able to send reliable
data traffic to a peer browser.
----------------------------------------------------------------
F34 The browser must support priortization of
streams and data.
----------------------------------------------------------------
F35 The browser must enable verification, given
the right circumstances and by use of other
trusted communication, of that streams and
data received have not been manipulated by
any party.
----------------------------------------------------------------
F36 The browser must encrypt, authenticate and
integrity protect media and data on a
per-packet asis, and must drop incoming media
and data packets that fail the per-packet
integrity check. In addition, the browser
must support a mechanism for cryptographically
binding media and data security keys to the
user identity (see R-ID-BINDING in [RFC5479]).
----------------------------------------------------------------
F37 The browser must be able to send streams and
data to a peer in the presence of FWs that only
allows http(s) traffic.
----------------------------------------------------------------
F38 The browser must be able to collect statistics,
related to the transport of audio and video
between peers, needed to estimate quality of
experience.
----------------------------------------------------------------
F39 The browser must make it possible to set up a
call between two parties without one party
learning the other party's host IP address.
----------------------------------------------------------------
]]></artwork>
</figure>
</section>
</section>
<section title="IANA Considerations" toc="default">
<t>TBD</t>
</section>
<section anchor="sec-security" title="Security Considerations"
toc="default">
<section anchor="sec-security-int" title="Introduction" toc="default">
<t>A malicious web application might use the browser to perform Denial
Of Service (DOS) attacks on NAT infrastructure, or on peer devices.
Also, a malicious web application might silently establish outgoing,
and accept incoming, streams on an already established connection.</t>
<t>Based on the identified security risks, this section will describe
security considerations for the browser and web application.</t>
</section>
<section anchor="sec-security-browser" title="Browser Considerations"
toc="default">
<t>The browser is expected to provide mechanisms for getting user
consent to use device resources such as camera and microphone.</t>
<t>The browser is expected to provide mechanisms for informing the
user that device resources such as camera and microphone are in use
("hot").</t>
<t>The browser is expected to provide mechanisms for users to revise
and even completely revoke consent to use device resources such as
camera and microphone.</t>
<t>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.</t>
<t>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").</t>
<t>The browser is expected to provide mechanisms for users to revise
and even completely revoke consent to use the screen, part thereof or
an application is serving as a stream source.</t>
<t>The browser is expected to provide mechanisms in order to assure
that streams are the ones the recipient intended to receive.</t>
<t>The browser is expected to provide mechanisms that allows the users
to verify that the streams received have not be manipulated (F35).</t>
<t>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.</t>
<t>The browser needs to ensure that the stream negotiation procedures
are not seen as Denial Of Service (DOS) by other entities.</t>
</section>
<section anchor="sec-security-wepapp"
title="Web Application Considerations" toc="default">
<t>The web application is expected to ensure user consent in sending
and receiving media streams.</t>
</section>
</section>
<section title="Additional use-cases">
<t>Several additional use-cases have been discussed. At this point these
use-cases are not included as requirement deriving use-cases for
different reasons (lack of documentation, overlap with existing
use-cases, lack of consensus). For completeness these additional
use-cases are listed below:<list style="numbers">
<t>Use-cases regarding different situations when being invited to a
“session”, e.g. browser open, browser open but another
tab active, browser open but active in session, browser closed,
…. (Matthew Kaufman); discussed at webrtc meeting</t>
<t>E911 (Paul Beaumont)
http://www.ietf.org/mail-archive/web/rtcweb/current/msg00525.html,
followed up by Stephan Wenger</t>
<t>Local Recording and Remote recording (John): Discussed a _lot_ on
the mail lists (rtcweb as well as public-webrtc) lAugust and
September 2011. Concrete proposal:
http://www.ietf.org/mail-archive/web/rtcweb/current/msg01006.html
(remote) and
http://www.ietf.org/mail-archive/web/rtcweb/current/msg00734.html
(local)</t>
<t>Emergency access for disabled (Bernard Aboba)
http://www.ietf.org/mail-archive/web/rtcweb/current/msg00478.html</t>
<t>Clue use-cases (Roni Even)
http://tools.ietf.org/html/draft-ietf-clue-telepresence-use-cases-01</t>
<t>Rohan red cross (Cullen Jennings);
http://www.ietf.org/mail-archive/web/rtcweb/current/msg00323.html</t>
<t>Security camera/baby monitor usage
http://www.ietf.org/mail-archive/web/rtcweb/current/msg00543.html</t>
<t>Large multiparty session
http://www.ietf.org/mail-archive/web/rtcweb/current/msg00530.html</t>
<t>Call center
http://www.ietf.org/mail-archive/web/rtcweb/current/msg04203.html</t>
<t>Enterprise policies
http://www.ietf.org/mail-archive/web/rtcweb/current/msg04271.html</t>
<t>Low-complex multiparty central node
http://www.ietf.org/mail-archive/web/rtcweb/current/msg04430.html</t>
<t>Multiparty central node that is not allowed to decipher
http://www.ietf.org/mail-archive/web/rtcweb/current/msg04457.html</t>
<t>Enable company coop without being able to decipher
http://www.ietf.org/mail-archive/web/rtcweb/current/msg04461.html</t>
</list></t>
</section>
<section anchor="sec-acks" title="Acknowledgements" toc="default">
<t>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, </t>
<t>Dan Burnett has reviewed and proposed a lot of things that enhances
the document. Most of this has been incorporated in rev -05.</t>
<t>Stephan Wenger has provided a lot of useful input and feedback, as
well as editorial comments.</t>
<t>Harald Alvestrand and Ted Hardie have provided comments and feedback
on the draft.</t>
<t>Harald Alvestrand and Cullen Jennings have provided additional
use-cases.</t>
<t>Thank You to everyone in the RTCWEB community that have provided
comments, feedback and improvement proposals on the draft content.</t>
</section>
<section title="Change Log">
<t>[RFC EDITOR NOTE: Please remove this section when publishing]</t>
<t>Changes from draft-ietf-rtcweb-use-cases-and-requirements-10</t>
<t><list style="symbols">
<t>Described that the API requirements are really from a W3C perspective and are
supplied as an appendix in the introduction. Moved API requirements to an Appendix. </t>
<t>Removed the "Conventions" section with the key-words and reference to RFC2119.
Also changed uppercase MUST's/SHOULD's to lowercase.</t>
<t>Added a note on the proposed use of the document to the introduction.</t>
<t>Removed the note talking about WS from the "FW that only allows http" use-case.</t>
<t>Removed the word "Skype" that was used as example in one of the use-cases.</t>
<t>Clarified F3 (the req saying the everything the browser sends must be rate controlled).</t>
<t>Removed the TBD saying we need to define reasonable levels from the requirement saying that quality must be good even
in presence of packet losses (F5), and changed "must" to "should" (Based on a list discussion
involving Bernard).</t>
<t>Removed F6 ("The browser must be able to handle high loss and
jitter levels in a graceful way."), also after a list discussion.</t>
<t>Clarified F7 (used to say that the browser must support fast stream switches, now says that reference frames must
be inserted when requested).</t>
<t>Removed the questions from F9 (echo cancellation), F10 (syncronization), F21 (telephony codec).</t>
<t>Exchanged "restrictive firewalls" for "limited middleboxes" in F19 (as proposed by Martin).</t>
<t>Expanded DTMF and IVR in F22 (proposed by Martin)</t>
<t>Added ref to RFC5405 in F23 (proposed by Lars Eggert).</t>
<t>Exchanged "service provided" for "web application" in F32.</t>
<t>Changed the text in 3.2.1 that motivates F36 (new text "It is essential
that media and data be encrypted, authenticated ... bound to the user
identity."); and rewrote F36, included a ref to RFC5479.</t>
<t>Changed "quality of service" to "quality of experience" in F38.</t>
<t>Added F39.</t>
<t>Used new formulation of A17 (proposed by Martin).</t>
<t>Updated A20.</t>
<t>Updated A25.</t>
</list>
</t>
<t>Changes from draft-ietf-rtcweb-use-cases-and-requirements-09</t>
<t><list style="symbols">
<t>Changed "video communication session" to "audiovisual communication session.</t>
</list>
</t>
<t>Changes from draft-ietf-rtcweb-use-cases-and-requirements-08</t>
<t><list style="symbols">
<t>Changed "eavesdropping" to "wiretapping" and referenced RFC2804.</t>
<t>Removed informal ref webrtc_req; that document has been
abandoned by the W3C webrtc WG.</t>
<t> Added use-case where one user is behind a FW that only
allows http; derived req. F37.</t>
<t>Changed F24 slightly; MUST-> SHOULD, inserted "available".</t>
<t>Added a clause to "Simple video communication service" saying
that the service provider monitors the quality of service, and
derived reqs F38 and A26.</t>
</list>
</t>
<t>Changes from draft-ietf-rtcweb-use-cases-and-requirements-07</t>
<t><list style="symbols">
<t>Added "and data exchange" to 1. Introduction.</t>
<t>Removed cone and symmetric NAT from 4.1 Introduction, refers
to RFC4787 instead.</t>
<t>Added text on enabling verifyication of that the media has not
been manipulated by anyone to use-case "Simple Video
Communication Service", derived req. F35</t>
<t>Added text on that the browser should reject media (data) that has
been created/injected/modified by non-trusted party, derived req. F36</t>
<t>Added text on enabling the app to refrain from revealing IP
address to use-case "Simple Video Communication Service", derived
req. A25</t>
<t>Added use-case "Simple Video Communication Service with
file exchange", derived reqs F33 and A24</t>
<t>Added priority of video streams to "Hockey game viewer" use case,
added priority of data to "on-line game use-case", derived reqs F34
and A23</t>
<t> In F22, "the IVR" -> "a DTMF based IVR".</t>
<t>Updated req F23 to clarify that requirements such as NAT traversal,
prtoection from eavesdropping, rate control applies also to datagram.</t>
</list></t>
<t>Changes from draft-ietf-rtcweb-use-cases-and-requirements-06</t>
<t><list style="symbols">
<t>Renaming of requirements (FaI1 -> F31), (FaI2 -> F32) and (AaI1 -> A22)</t>
</list></t>
<t>Changes from draft-ietf-rtcweb-use-cases-and-requirements-05</t>
<t><list style="symbols">
<t>Added use-case "global service provider", derived reqs associated
with several STUN/TURN servers</t>
<t>Added use-case "enterprise aspects", derived req associated with
enabling the network provider to supply STUN and TURN servers</t>
<t>The requirements from the above are ICE specific and labeled
accordingly</t>
<t>Separated the requirements phrased like "processing such as pan,
mix and render" for audio to be specific reqs on spatialization,
level measurement, level adjustment and mixing (discussed on the
lists in
http://www.ietf.org/mail-archive/web/rtcweb/current/msg01648.html
and
http://lists.w3.org/Archives/Public/public-webrtc/2011Sep/0102.html)</t>
<t>Added use-case on sharing as decided in
http://www.ietf.org/mail-archive/web/rtcweb/current/msg01700.html,
derived reqs F30 and A21</t>
<t>Added the list of common considerations proposed in mail
http://www.ietf.org/mail-archive/web/rtcweb/current/msg01562.html to
the Introduction of the use-case section</t>
</list></t>
<t>Changes from draft-ietf-rtcweb-use-cases-and-requirements-04</t>
<t><list style="symbols">
<t>Most changes based on the input from Dan Burnett
http://www.ietf.org/mail-archive/web/rtcweb/current/msg00948.html</t>
<t>Many editorial changes</t>
<t>4.2.1.1 Clarified</t>
<t>Some clarification added to 4.3.1.1 as a note</t>
<t>F-requirements updated (see reply to Dan's mail).</t>
<t>Almost all A-requirements updated to start "The Web API MUST
provide ..."</t>
<t>A8 removed, A9 rephrased to cover A8 and old A9</t>
<t>A15 rephrased</t>
<t>For more details, and discussion, look att the response to Dan's
mail
http://www.ietf.org/mail-archive/web/rtcweb/current/msg01177.html</t>
</list></t>
<t>Changes from draft-ietf-rtcweb-use-cases-and-requirements-03</t>
<t><list style="symbols">
<t>Editorials</t>
<t>Changed when the self-view is displayed in 4.2.1.1, and added
words about allowing users to remove and re-insert it.</t>
<t>Clarified 4.2.6.1</t>
<t>Removed the "mono" stuff from 4.2.7.1</t>
<t>Added that communication should not be possible to eavesdrop to
most use cases - and req. F17</t>
<t>Re-phrased 4.3.3.1 to not describe the technical solution so
much, and removed "stereo" stuff. Solution possibilities are now in
a note.</t>
<t>Re-inserted API requirements after discussion in the W3C webrtc
WG. (Re-phrased A15 and added A18 compared to version -02).</t>
</list></t>
<t>Changes from draft-ietf-rtcweb-use-cases-and-requirements-02</t>
<t><list style="symbols">
<t>Removed desrciption/list of API requirements, instead</t>
<t>Reference to W3C webrtc_reqs document for API requirements</t>
</list></t>
<t>Changes from draft-ietf-rtcweb-ucreqs-01</t>
<t><list style="symbols">
<t>Changed Intended status to Information</t>
<t>Changed "Ipr" to "trust200902"</t>
<t>Added use case "Simple video communication service, NAT/FW that
blocks UDP", and derived new req F26</t>
<t>Added use case "Distributed Music Band" and derived new req
A17</t>
<t>Added F24 as requirement derived from use case "Simple video
communication service with inter-operator calling"</t>
<t>Added section "Additional use cases"</t>
<t>Added text about ID handling to multiparty with central server
use case</t>
<t>Re-phrased A1 slightly</t>
</list></t>
<t>Changes from draft-ietf-rtcweb-ucreqs-00</t>
<t><list style="symbols">
<t>- Reshuffled: Just two main groups of use cases (b2b and
b2GW/Server); removed some specific use cases and added them instead
as flavors to the base use case (Simple video communciation)</t>
<t>- Changed the fromulation of F19</t>
<t>- Removed the requirement on an API for DTMF</t>
<t>- Removed "FX3: There SHOULD be a mapping of the minimum needed
data for setting up connections into SIP, so that the restriction to
SIP-carriable data can be verified. Not a rew on the browser but
rather on a document"</t>
<t>- (see
http://www.ietf.org/mail-archive/web/rtcweb/current/msg00227.html
for more details)</t>
<t>-Added text on informing user of that mic/cam is being used and
that it must be possible to revoce permission to use them in section
7.</t>
</list>Changes from draft-holmberg-rtcweb-ucreqs-01 <list
style="symbols">
<t>- Draft name changed to draft-ietf-rtcweb-ucreqs</t>
<t>- Use-case grouping introduced</t>
<t>- Additional use-cases added</t>
<t>- Additional reqs added (derived from use cases): F19-F25,
A16-A17</t>
</list></t>
<t>Changes from draft-holmberg-rtcweb-ucreqs-00 <list style="symbols">
<t>- Mapping between use-cases and requirements added (Harald
Alvestrand, 090311)</t>
<t>- Additional security considerations text (Harald Alvestrand,
090311)</t>
<t>- Clarification that user applications are assumed to be executed
by a browser (Ted Hardie, 080311)</t>
<t>- Editorial corrections and clarifications</t>
</list></t>
</section>
</middle>
<back>
<references title="Normative References">
<?rfc include="reference.RFC.2804"?>
<?rfc include="reference.RFC.5405"?>
<?rfc include="reference.RFC.5479"?>
</references>
<section anchor="api_reqs" title="API requirements" toc="default">
<t>This section contains the requirements on the API derived from the use-cases
in <xref target="use_cases"></xref>.</t>
<figure align="left" alt="" height="" suppress-title="false" title=""
width="">
<artwork align="left" alt="" height="" name="" type="" width=""
xml:space="preserve"><![CDATA[
REQ-ID DESCRIPTION
----------------------------------------------------------------
A1 The Web API must provide means for the
application to ask the browser for permission
to use cameras and microphones as input devices.
----------------------------------------------------------------
A2 The Web API must provide means for the web
application to control how streams generated
by input devices are used.
----------------------------------------------------------------
A3 The Web API must provide means for the web
application to control the local rendering of
streams (locally generated streams and streams
received from a peer).
----------------------------------------------------------------
A4 The Web API must provide means for the web
application to initiate sending of
stream/stream components to a peer.
----------------------------------------------------------------
A5 The Web API must provide means for the web
application to control the media format (codec)
to be used for the streams sent to a peer.
NOTE: The level of control depends on whether
the codec negotiation is handled by the browser
or the web application.
----------------------------------------------------------------
A6 The Web API must provide means for the web
application to modify the media format for
streams sent to a peer after a media stream
has been established.
----------------------------------------------------------------
A7 The Web API must provide means for
informing the web application of whether the
establishment of a stream with a peer was
successful or not.
----------------------------------------------------------------
A8 The Web API must provide means for the web
application to mute/unmute a stream or stream
component(s). When a stream is sent to a peer
mute status must be preserved in the stream
received by the peer.
----------------------------------------------------------------
A9 The Web API must provide means for the web
application to cease the sending of a stream
to a peer.
----------------------------------------------------------------
A10 The Web API must provide means for the web
application to cease processing and rendering
of a stream received from a peer.
----------------------------------------------------------------
A11 The Web API must provide means for
informing the web application when a
stream from a peer is no longer received.
----------------------------------------------------------------
A12 The Web API must provide means for
informing the web application when high
loss rates occur.
----------------------------------------------------------------
A13 The Web API must provide means for the web
application to apply spatialization effects to
audio streams.
----------------------------------------------------------------
A14 The Web API must provide means for the web
application to detect the level in audio
streams.
----------------------------------------------------------------
A15 The Web API must provide means for the web
application to adjust the level in audio
streams.
----------------------------------------------------------------
A16 The Web API must provide means for the web
application to mix audio streams.
----------------------------------------------------------------
A17 The Web API must provide a way to identify
streams such that an application is able to
match streams on a sending peer with the same
stream on all receiving peers.
----------------------------------------------------------------
A18 The Web API must provide a mechanism for sending
and receiving isolated discrete chunks of data.
----------------------------------------------------------------
A19 The Web API must provide means for the web
application to indicate the type of audio signal
(speech, audio) for audio stream(s)/stream
component(s).
----------------------------------------------------------------
A20 It must be possible for an initiator or a
responder web application to indicate the types
of media it is willing to accept incoming
streams for when setting up a connection (audio,
video, other). The types of media to be accepted
can be a subset of the types of media the browser
is able to accept.
----------------------------------------------------------------
A21 The Web API must provide means for the
application to ask the browser for permission
to the screen, a certain area on the screen
or what a certain application displays on the
screen as input to streams.
----------------------------------------------------------------
A22 The Web API must provide means for the
application to specify several STUN and/or
TURN servers to use.
----------------------------------------------------------------
A23 The Web API must provide means for the
application to specify the priority to
apply for outgoing streams and data.
----------------------------------------------------------------
A24 The Web API must provide a mechanism for sending
and receiving files.
----------------------------------------------------------------
A25 It must be possible for the application to
instruct the browser to refrain from exposing
the host IP address to the application
----------------------------------------------------------------
A26 The Web API must provide means for the
application to obtain the statistics (related
to transport, and collected by the browser)
needed to estimate quality of service.
----------------------------------------------------------------
]]></artwork>
</figure>
</section>
</back>
</rfc>
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