RTCWEB Working Group C. Holmberg
Internet-Draft S. Hakansson
Intended status: Informational G. Eriksson
Expires: March 4, 2012 Ericsson
September 1, 2011
Web Real-Time Communication Use-cases and Requirements
draft-ietf-rtcweb-use-cases-and-requirements-04.txt
Abstract
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.
Status of this Memo
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provisions of BCP 78 and BCP 79.
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This Internet-Draft will expire on March 4, 2012.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Conventions . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Definitions . . . . . . . . . . . . . . . . . . . . . . . . . 3
4. Use-cases . . . . . . . . . . . . . . . . . . . . . . . . . . 3
4.1. Introduction . . . . . . . . . . . . . . . . . . . . . . . 3
4.2. Browser-to-browser use-cases . . . . . . . . . . . . . . . 3
4.2.1. Simple Video Communication Service . . . . . . . . . . 3
4.2.2. Simple Video Communication Service, NAT/FW that
blocks UDP . . . . . . . . . . . . . . . . . . . . . . 4
4.2.3. Simple Video Communication Service, access change . . 4
4.2.4. Simple Video Communication Service, QoS . . . . . . . 5
4.2.5. Simple video communication service with
inter-operator calling . . . . . . . . . . . . . . . . 5
4.2.6. Hockey Game Viewer . . . . . . . . . . . . . . . . . . 6
4.2.7. Multiparty video communication . . . . . . . . . . . . 7
4.2.8. Multiparty on-line game with voice communication . . . 7
4.2.9. Distributed Music Band . . . . . . . . . . . . . . . . 8
4.3. Browser - GW/Server use cases . . . . . . . . . . . . . . 9
4.3.1. Telephony terminal . . . . . . . . . . . . . . . . . . 9
4.3.2. Fedex Call . . . . . . . . . . . . . . . . . . . . . . 9
4.3.3. Video conferencing system with central server . . . . 9
5. Requirements . . . . . . . . . . . . . . . . . . . . . . . . . 10
5.1. General . . . . . . . . . . . . . . . . . . . . . . . . . 11
5.2. Browser requirements . . . . . . . . . . . . . . . . . . . 11
5.3. API requirements . . . . . . . . . . . . . . . . . . . . . 13
6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 15
7. Security Considerations . . . . . . . . . . . . . . . . . . . 15
7.1. Introduction . . . . . . . . . . . . . . . . . . . . . . . 15
7.2. Browser Considerations . . . . . . . . . . . . . . . . . . 15
7.3. Web Application Considerations . . . . . . . . . . . . . . 15
8. Additional use-cases . . . . . . . . . . . . . . . . . . . . . 16
9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 16
10. Change Log . . . . . . . . . . . . . . . . . . . . . . . . . . 17
11. References . . . . . . . . . . . . . . . . . . . . . . . . . . 18
11.1. Normative References . . . . . . . . . . . . . . . . . . . 18
11.2. Informative References . . . . . . . . . . . . . . . . . . 18
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 18
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1. Introduction
This document presents a few use-cases of web applications that are
executed in a browser and use real-time communication capabilities.
Based on the use-cases, the document derives requirements related to
the browser and the API used by web applications in the browser.
The requirements related to the browser are named "Fn" and are
described in Section 5.2
The requirements related to the API are named "An" and are described
in Section 5.3
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.
2. Conventions
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
[RFC2119].
3. Definitions
TBD
4. Use-cases
4.1. Introduction
This section describes web based real-time communication use-cases,
from which requirements are derived.
4.2. Browser-to-browser use-cases
4.2.1. Simple Video Communication Service
4.2.1.1. Description
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
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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.
During session establishment a self-view is displayed, and once the
session has been established the video sent from the remote peer is
displayed displayed in addition to the self-view. The users can
during the session select to remove, and re-insert the self-view.
The users can change the sizes of the video displays during the
session. The users can also pause sending of media (audio, video, or
both), and mute incoming media.
It is essential that the communication can not be eavesdropped.
Any session participant can end the session at any time.
The users are using communication devices of different makes, with
different operating systems and browsers from different vendors.
One user has an unreliable Internet connection. It sometimes has
packet losses, and is sometimes goes down completely.
One user is located behind a Network Address Translator (NAT).
4.2.1.2. Derived Requirements
F1, F2, F3, F4, F5, F6, F8, F9, F10, F17, F22, F25
A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12, A13
4.2.2. Simple Video Communication Service, NAT/FW that blocks UDP
4.2.2.1. Description
This use-case is almost identical to the previos one. The difference
is that one of the users is behind a NAT that blocks UDP traffic.
4.2.2.2. Derived Requirements
F1, F2, F3, F4, F5, F6, F8, F9, F10, F17, F22, F23, F25, F26
A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12, A13
4.2.3. Simple Video Communication Service, access change
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4.2.3.1. Description
This use-case is almost identical to "4.2.1 Simple Video
Communication Service". The difference is that the user changes
network access during the session:
The communication device used by one of the users have several
network adapters (Ethernet, WiFi, Cellular). The communication
device is access 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.
4.2.3.2. Derived Requirements
F1, F2, F3, F4, F5, F6, F8, F9, F10, F17, F22, F23, F25
A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12, A13
4.2.4. Simple Video Communication Service, QoS
4.2.4.1. Description
This use-case is almost identical to the previos one. The use of QoS
capabilities is added:
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.
4.2.4.2. Derived Requirements
F1, F2, F3, F4, F5, F6, F8, F9, F10, F17, F21, F22, F23, F25
A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12, A13
4.2.5. Simple video communication service with inter-operator calling
4.2.5.1. Description
Two users have logged into two different web applications, provided
by different service providers.
The service providers are interconnected by some means, but exchange
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no more information about the users than what can be carried using
SIP.
NOTE: More profiling of what this means may be needed.
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.
The same functionality as in the "4.2.1 Simple Video Communication
Service" is available.
The same issues with connectivity apply.
4.2.5.2. Derived requirements
F1, F2, F3, F4, F5, F6, F8, F9, F10, F17, F22, F24, F25
A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12, A13
4.2.6. Hockey Game Viewer
4.2.6.1. 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.
Before the game starts, and during game breaks, the talent scout and
the manager have a 1-1 video communication. 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 ot the desktop camera (self-view).
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
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picture-in-picture thumbnails of the rear facing camera (self-view)
and the desktop camera.
It is essential that the communication can not be eavesdropped.
4.2.6.2. Derived Requirements
F1, F2, F3, F4, F5, F6, F8, F9, F10, F14, F17
A1, A2, A3, A4, A5, A7, A8, A9, A10, A11, A12, A13, A15
4.2.7. Multiparty video communication
4.2.7.1. Description
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. The web application in the browser
of each user is responsible for setting up streams to all receivers.
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.
Each video stream received is by default displayed in a thumbnail
frame within the browser, but users can change the display size.
It is essential that the communication can not be eavesdropped.
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 and mix the audio
from all received streams locally in the browser.
4.2.7.2. Derived Requirements
F1, F2, F3, F4, F5, F6, F8, F9, F10, F11, F12, F13, F14, F17, F22
A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12, A13, A14, A15
4.2.8. Multiparty on-line game with voice communication
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4.2.8.1. Description
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.
Quick updates of the game state is required.
It is essential that the communication can not be eavesdropped.
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.
4.2.8.2. Derived Requirements
F1, F2, F3, F4, F5, F6, F8, F9, F11, F12, F13, F15, F17, F20
A1, A2, A3, A4, A5, A7, A8, A9, A10, A11, A12, A13, A14, A15, A16
4.2.9. Distributed Music Band
4.2.9.1. Description
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.
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.).
4.2.9.2. Derived Requirements
F1, F2, F3, F4, F5, F6, F8, F9, F11, F12, F13
A1, A2, A3, A4, A5, A7, A8, A9, A10, A11, A12, A13, A14, A15, A17
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4.3. Browser - GW/Server use cases
4.3.1. Telephony terminal
4.3.1.1. 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 used.
It is essential that the communication can not be eavesdropped.
4.3.1.2. Derived Requirements
F1, F2, F3, F4, F5, F6, F8, F9, F10, F17, F18
A1, A2, A3, A4, A7, A8, A9, A10, A11, A12, A13
4.3.2. Fedex Call
4.3.2.1. Description
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.
4.3.2.2. Derived Requirements
F1, F2, F3, F4, F5, F6, F8, F9, F10, F18, F19
A1, A2, A3, A4, A7, A8, A9, A10, A11, A12, A13
4.3.3. Video conferencing system with central server
4.3.3.1. Description
An organization uses a video communication system that supports the
establishment of multiparty video sessions using a central conference
server.
The browsers 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
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such as spatialization) and sends towards each participant a mixed
audio stream which 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.
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).
It is essential that the communication can not be eavesdropped.
All participant 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.
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 exists 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, the
server just selects, c) each browser sends just an high resolution
stream, the server trancodes into low reslution streams as required.
4.3.3.2. Derived Requirements
F1, F2, F3, F4, F5, F6, F7, F8, F9, F10, F14, F16, F17
A1, A2, A3, A4, A5, A7, A8, A9, A10, A11, A12, A13, A15
5. Requirements
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5.1. General
This section contains the requirements derived from the use-cases in
section 4.
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.
5.2. Browser 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 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?
----------------------------------------------------------------
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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 there should be a way to navigate
the IVR
----------------------------------------------------------------
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.
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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
----------------------------------------------------------------
F26 The browser MUST be able to send streams to a
peer in presence of NATs that block UDP traffic.
----------------------------------------------------------------
5.3. API requirements
REQ-ID DESCRIPTION
----------------------------------------------------------------
A1 The web application MUST be able to ask the
browser for permission to use 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.
----------------------------------------------------------------
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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 provided with an
identifier for the stream that can be communicated
to the other party of the communication, and which
the other party can associate with its end of the
same stream.
----------------------------------------------------------------
A16 It MUST be possible for the web application to
send and receive datagrams to/from peer
----------------------------------------------------------------
A17 It MUST be possible for the web application to
indicate the type of audio signal (speech, audio)
----------------------------------------------------------------
A18 It must be possible for an initiator or a
responder Web application to indicate the types
of media he's willing to accept incoming streams
for when setting up a connection (audio, video,
other). The types of media he's willing to accept
can be a subset of the types of media the browser
is able to accept.
----------------------------------------------------------------
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6. IANA Considerations
TBD
7. Security Considerations
7.1. Introduction
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.
Based on the identified security risks, this section will describe
security considerations for the browser and web application.
7.2. 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 is expected to provide mechanisms for users to revise
consent to use device resources such as camera and microphone.
The browser is expected to provide mechanisms in order to assure that
streams are the ones the recipient intended to receive.
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.
The browser needs to ensure that the stream negotiation procedures
are not seen as Denial Of Service (DOS) by other entities.
7.3. Web Application Considerations
The web application is expected to ensure user consent in sending and
receiving media streams.
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8. Additional use-cases
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:
1. 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
2. Different TURN provider scenarios (Cullen Jennings); discussed
at the webrtc meeting
3. E911 (Paul Beaumont) http://www.ietf.org/mail-archive/web/
rtcweb/current/msg00525.html, followed up by Stephan Wenger
4. Local Recording and Remote recording (John): Discussed a _lot_
on the mail lists (rtcweb as well as public-webrtc) late August
2011. Not concluded at time of writing.
5. Emergency access for disabled (Bernard Aboba) http://
www.ietf.org/mail-archive/web/rtcweb/current/msg00478.html
6. Clue use-cases (Roni Even) http://tools.ietf.org/html/
draft-ietf-clue-telepresence-use-cases-01
7. Rohan red cross (Cullen Jennings); http://www.ietf.org/
mail-archive/web/rtcweb/current/msg00323.html
8. Remote assistance (ala VNC or RDP) - User is helping another
user on their computer with either view-only or view-with-
control, either of just the browser of the the entire screen. ht
tp://www.ietf.org/mail-archive/web/rtcweb/current/msg00543.html
9. Security camera/baby monitor usage http://www.ietf.org/
mail-archive/web/rtcweb/current/msg00543.html
10. Large multiparty session http://www.ietf.org/mail-archive/web/
rtcweb/current/msg00530.html
9. Acknowledgements
Stephan Wenger has provided a lot of useful input and feedback, as
well as editorial comments.
Harald Alvestrand and Ted Hardie have provided comments and feedback
on the draft.
Harald Alvestrand and Cullen Jennings have provided additional use-
cases.
Thank You to everyone in the RTCWEB community that have provided
comments, feedback and improvement proposals on the draft content.
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10. Change Log
[RFC EDITOR NOTE: Please remove this section when publishing]
Changes from draft-ietf-rtcweb-use-cases-and-requirements-03
o Editorials
o Changed when the self-view is displayed in 4.2.1.1, and added
words about allowing users to remove and re-insert it.
o Clarified 4.2.6.1
o Removed the "mono" stuff from 4.2.7.1
o Added that communication should not be possible to eavesdrop to
most use cases - and req. F17
o 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.
o Re-inserted API requirements after discussion in the W3C webrtc
WG. (Re-phrased A15 and added A18 compared to version -02).
Changes from draft-ietf-rtcweb-use-cases-and-requirements-02
o Removed desrciption/list of API requirements, instead
o Reference to W3C webrtc_reqs document for API requirements
Changes from draft-ietf-rtcweb-ucreqs-01
o Changed Intended status to Information
o Changed "Ipr" to "trust200902"
o Added use case "Simple video communication service, NAT/FW that
blocks UDP", and derived new req F26
o Added use case "Distributed Music Band" and derived new req A17
o Added F24 as requirement derived from use case "Simple video
communication service with inter-operator calling"
o Added section "Additional use cases"
o Added text about ID handling to multiparty with central server use
case
o Re-phrased A1 slightly
Changes from draft-ietf-rtcweb-ucreqs-00
o - 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)
o - Changed the fromulation of F19
o - Removed the requirement on an API for DTMF
o - 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
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but rather on a document"
o - (see
http://www.ietf.org/mail-archive/web/rtcweb/current/msg00227.html
for more details)
o -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.
Changes from draft-holmberg-rtcweb-ucreqs-01
o - Draft name changed to draft-ietf-rtcweb-ucreqs
o - Use-case grouping introduced
o - Additional use-cases added
o - Additional reqs added (derived from use cases): F19-F25, A16-A17
Changes from draft-holmberg-rtcweb-ucreqs-00
o - Mapping between use-cases and requirements added (Harald
Alvestrand, 090311)
o - Additional security considerations text (Harald Alvestrand,
090311)
o - Clarification that user applications are assumed to be executed
by a browser (Ted Hardie, 080311)
o - Editorial corrections and clarifications
11. References
11.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
11.2. Informative References
[webrtc_reqs]
"Webrt requirements,
http://dev.w3.org/2011/webrtc/editor/webrtc_reqs.html".
Authors' Addresses
Christer Holmberg
Ericsson
Hirsalantie 11
Jorvas 02420
Finland
Email: christer.holmberg@ericsson.com
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Stefan Hakansson
Ericsson
Laboratoriegrand 11
Lulea 97128
Sweden
Email: stefan.lk.hakansson@ericsson.com
Goran AP Eriksson
Ericsson
Farogatan 6
Stockholm 16480
Sweden
Email: goran.ap.eriksson@ericsson.com
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