One document matched: draft-ietf-rtcweb-use-cases-and-requirements-00.xml
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<rfc category="std" docName="draft-ietf-rtcweb-use-cases-and-requirements-00.txt"
ipr="trust200811" 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="2011" />
<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.
Based on the use-cases, the document also derives requirements related
to the browser, and the API used by web applications to request and
control media stream services provided by the browser.</t>
</abstract>
</front>
<middle>
<section title="Introduction" toc="default">
<t>This document presents a few use-case of web applications that are
executed in a browser and use real-time communication capabilities.
Based on the use-cases, the document derives requirements related to the
browser and the API used by web applications in the browser.</t>
<t>The document focuses on requirements related to real-time media
streams. Requirements related to privacy, signalling between the browser
and web server etc are currently not considered.</t>
</section>
<section title="Conventions" toc="default">
<t>The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in BCP 14, RFC 2119 <xref
format="default" pageno="false" target="RFC2119"></xref>.</t>
</section>
<section title="Definitions" toc="default">
<t>TBD</t>
</section>
<section 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 later derived.</t>
</section>
<section title="Browser-to-browser use-cases" toc="default">
<section title="Simple Video Communication Service" toc="default">
<section title="Description" toc="default">
<t>In the service the users have loaded, and logged into, a video
communication web application into their browsers, provided by the
same service provider. The web service publishes information about
user login status, by pushing updates to the web application in
the browsers. By selecting an online peer user, a 1-1 video
communication session between the browsers of the peers is
initiated. The invited user might accept or reject the
session.</t>
<t>When the session has been established, a self-view, as well as
the video sent from the remote peer, are displayed. The users can
change the display sizes during the session. The users can also
pause sending of media (audio, video, or both), and mute incoming
media.</t>
<t>Any session participant can end the session at any time.</t>
<t>One participant has an unreliable internet connection. It
sometimes has packet losses, and is sometimes goes down
completely.</t>
<t>One participant is located behind a Network Address Translator
(NAT).</t>
</section>
<section title="Derived Requirements" toc="default">
<t>F1, F2, F3, F4, F5, F6, F8, F9, F10, F22</t>
<t>A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12, A13</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>Each web service publishes information about user login status
for users that have a relationship with the other user; how this
is established is out of scope.</t>
<t>The same functionality as in the "Simple Video Communication
Service" is available.</t>
<t>The same issues with connectivity apply.</t>
</section>
<section title="Derived requirements">
<t>F24: The browser MUST be able to initiate and accept a media
session where the data needed for establishment can be carried in
SIP.</t>
<t>F25: The browser MUST support a baseline audio and video
codec</t>
<t>(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>
</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 for viewing the game and
discussing with the talent scout. 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, while a
thumbnail of the rear facing camera is overlaid.</t>
<t>Most of the mobile phone screen is covered by a self view of
the front facing camera. A thumbnail of the rear facing cameras
view is overlaid.</t>
</section>
<section title="Derived Requirements" toc="default">
<t>F1, F2, F3, F4, F5, F6, F8, F9, F10, F14</t>
<t>A1, A2, A3, A4, A5, A7, A8, A9, A10, A11, A12, A13, A15</t>
</section>
</section>
<section title="Video Size Change" toc="default">
<section title="Description" toc="default">
<t>Alice and Bob are in a video call in their browsers and have
negotiate a high resolution video. Bob decides to change the size
of the windows his browser is displaying video to a small
size.</t>
<t>Bob's browser regenerates the video codec paramters with
Alice's browser to change the resolution of the video Alice sends
to match the smaller size.</t>
</section>
<section title="Derived Requirements" toc="default">
<t>F22 ( It SHOULD be possible to modify video codec parameters
during a session.)</t>
</section>
</section>
</section>
<section title="Telephony 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 will
be shown in the same manner as when a mobile phone used.</t>
</section>
<section title="Derived Requirements" toc="default">
<t>F1, F2, F3, F4, F5, F6, F8, F9, F10, F18, F19</t>
<t>A1, A2, A3, A4, A7, A8, A9, A10, A11, A12, A13, A16</t>
</section>
</section>
<section title="Fedex Call" toc="default">
<section title="Description" toc="default">
<t>Alice uses her web browser with a service something like Skype
to be able to phone 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>F19 (DTMF)</t>
<t>A16 (DTMF API)</t>
</section>
</section>
</section>
<section title="Video conferenceing use-cases" toc="default">
<section title="Multiparty video communication" toc="default">
<section title="Description" toc="default">
<t>In this use case the simple video communication service 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.</t>
<t>The audio sent by each participant is a mono stream. However,
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.</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>
</section>
<section title="Derived Requirements" toc="default">
<t>F1, F2, F3, F4, F5, F6, F8, F9, F10, F11, F12, F13, F14</t>
<t>A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12, A13, A14,
A15</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 browsers of all participants send an audio stream (mono or
stereo depending on the equipment of a participant) to the central
server. The central server mixes the audio streams and sends
towards the participants a mixed stereo stream.</t>
<t>All participants send two video streams towards the server, one
low resolution and one high resolution. At 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, based on speech activity.</t>
<t>The organization has an internal network set up with an
aggressive firewall handling access to the internet. If users can
not physically access the internal network, they can establish a
Virtual Private Network (VPN).</t>
<t>It is essential that the communication can not be
eavesdropped.</t>
</section>
<section title="Derived Requirements" toc="default">
<t>F1, F2, F3, F4, F5, F6, F7, F8, F9, F10, F14, F16, F17</t>
<t>A1, A2, A3, A4, A5, A7, A8, A9, A10, A11, A12, A13, A15</t>
</section>
</section>
</section>
<section title="Embedded voice communicatoin use-cases" toc="default">
<section title="Multiparty on-line game with voice communication"
toc="default">
<section title="Description" toc="default">
<t>In this use-case, the voice part of the multiparty video
communication application 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.</t>
</section>
<section title="Derived Requirements" toc="default">
<t>F1, F2, F3, F4, F5, F6, F8, F9, F11, F12, F13, F15, F20</t>
<t>A1, A2, A3, A4, A5, A7, A8, A9, A10, A11, A12, A13, A14, A15,
A17</t>
</section>
</section>
</section>
<section title="Bandwidth/QoS/mobility use-cases" toc="default">
<section title="NIC Change" toc="default">
<section title="Description" toc="default">
<t>Alice is using her notebook computer that is plugged in to 1G
ethernet and has 802.11 wireless interface. Alice is in a call
talking with Bob and decides to unplug her notebook computer and
walk down to a different room, and continue the call from
there.</t>
</section>
<section title="Derived Requirements" toc="default">
<t>F23: It MUST be possible to move from one network interface to
another one.</t>
</section>
</section>
<section title="QoS Marking" toc="default">
<section title="Description" toc="default">
<t>Alice's browser is on a computer behind a common residential
router that supports prioritization of traffic.</t>
<t>F21: The browser MUST be able to take advantage of capabilities
to prioritize voice and video appropriately.</t>
</section>
<section title="Derived Requirements" toc="default">
<t>F19: (DTMF)</t>
</section>
</section>
</section>
</section>
<section title="Requirements" toc="default">
<section title="General" toc="default">
<t>This section contains requirements, derived from the use-cases in
section 4.</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
(OS), is outside the scope of this document.</t>
</section>
<section 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 to a
peer in presence of NATs.
----------------------------------------------------------------
F3 Transmitted streams MUST be rate controlled.
----------------------------------------------------------------
F4 The browser MUST be able to receive, process and
render streams from peers.
----------------------------------------------------------------
F5 The browser MUST be able to render good quality
audio and video even in presence of reasonable
levels of jitter and packet losses.
TBD: What is a reasonable level?
----------------------------------------------------------------
F6 The browser MUST be able to handle high loss and
jitter levels in a graceful way.
----------------------------------------------------------------
F7 The browser MUST support fast stream switches.
----------------------------------------------------------------
F8 The browser MUST detect when a stream from a
peer is not received any more
----------------------------------------------------------------
F9 When there are both incoming and outgoing audio
streams, echo cancellation MUST be made available to
avoid disturbing echo during conversation.
QUESTION: How much control should be left to the
web application?
----------------------------------------------------------------
F10 The browser MUST support synchronization of
audio and video.
QUESTION: How much control should be left to the
web application?
----------------------------------------------------------------
F11 The browser MUST be able to transmit streams to
several peers concurrently.
----------------------------------------------------------------
F12 The browser MUST be able to receive streams from
multiple peers concurrently.
----------------------------------------------------------------
F13 The browser MUST be able to pan, mix and render
several concurrent audio streams.
----------------------------------------------------------------
F14 The browser MUST be able to render several
concurrent video streams
----------------------------------------------------------------
F15 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).
----------------------------------------------------------------
F16 Streams MUST be able to pass through restrictive
firewalls.
----------------------------------------------------------------
F17 It MUST be possible to protect streams from
eavesdropping.
----------------------------------------------------------------
F18 The browser MUST support an audio media format
(codec) that is commonly supported by existing
telephony services.
QUESTION: G.711?
----------------------------------------------------------------
F19 The browser must be able to insert DTMF signals
in a media stream
----------------------------------------------------------------
F20 The browser must be able to send short
latency datagram traffic to a peer browser
----------------------------------------------------------------
F21 The browser MUST be able to take advantage of
capabilities to prioritize voice and video
appropriately.
----------------------------------------------------------------
F22 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
----------------------------------------------------------------
F23 It MUST be possible to move from one network
interface to another one
----------------------------------------------------------------
F24 The browser MUST be able to initiate and accept a
media session where the data needed for establishment
can be carried in SIP.
----------------------------------------------------------------
F25 The browser MUST support a baseline audio and
video codec
----------------------------------------------------------------
]]></artwork>
</figure>
</section>
<section title="API 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
----------------------------------------------------------------
A1 The web application MUST be able to query the
user about the usage of cameras and microphones
as input devices.
----------------------------------------------------------------
A2 The web application MUST be able to control how
streams generated by input devices are used.
----------------------------------------------------------------
A3 The web application MUST be able to control the
local rendering of streams (locally generated streams
and streams received from a peer).
----------------------------------------------------------------
A4 The web application MUST be able to initiate
sending of stream/stream components to a peer.
----------------------------------------------------------------
A5 The web application MUST be able 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 After a media stream has been established, the
web application MUST be able to modify the media
format for streams sent to a peer.
----------------------------------------------------------------
A7 The web application MUST be made aware of
whether the establishment of a stream with a
peer was successful or not.
----------------------------------------------------------------
A8 The web application MUST be able to
pause/unpause the sending of a stream to a peer.
----------------------------------------------------------------
A9 The web application MUST be able to mute/unmute
a stream received from a peer.
----------------------------------------------------------------
A10 The web application MUST be able to cease the
sending of a stream to a peer.
----------------------------------------------------------------
A11 The web application MUST be able to cease
processing and rendering of a stream received
from a peer.
----------------------------------------------------------------
A12 The web application MUST be informed when a
stream from a peer is no longer received.
----------------------------------------------------------------
A13 The web application MUST be informed when high
loss rates occur.
----------------------------------------------------------------
A14 It MUST be possible for the web application to
control panning, mixing and other processing for
individual streams.
----------------------------------------------------------------
A15 The web application MUST be able to identity the
context of a stream.
----------------------------------------------------------------
A16 It MUST be possible for the web application to
order the browser to insert DTMF tones in a stream
----------------------------------------------------------------
A17 It MUST be possible for the web application to
send and receive datagrams to/from peer
----------------------------------------------------------------
]]></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 in order to assure
that streams are the ones the recipient intended to receive.</t>
<t>The browser is 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 anchor="sec-acks" title="Acknowledgements" toc="default">
<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-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>
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