One document matched: draft-holmberg-rtcweb-ucreqs-00.txt
RTCWEB Working Group C. Holmberg
Internet-Draft S. Hakansson
Intended status: Standards Track G. Eriksson
Expires: September 8, 2011 Ericsson
March 7, 2011
Web Real-Time Communication Use-cases and Requirements
draft-holmberg-rtcweb-ucreqs-00.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
This Internet-Draft is submitted to IETF in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on September 8, 2011.
Copyright Notice
Copyright (c) 2011 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Holmberg, et al. Expires September 8, 2011 [Page 1]
Internet-Draft RTC-Web March 2011
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Conventions . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Definitions . . . . . . . . . . . . . . . . . . . . . . . . . 3
4. Use-cases . . . . . . . . . . . . . . . . . . . . . . . . . . 3
4.1. Introduction . . . . . . . . . . . . . . . . . . . . . . . 3
4.2. Use-case: Simple Video Communication Service . . . . . . . 3
4.2.1. Description . . . . . . . . . . . . . . . . . . . . . 3
4.3. Use-case: Multiparty video communication . . . . . . . . . 4
4.3.1. Description . . . . . . . . . . . . . . . . . . . . . 4
4.4. Use-case: Multiparty on-line game with voice
communication . . . . . . . . . . . . . . . . . . . . . . 4
4.4.1. Description . . . . . . . . . . . . . . . . . . . . . 4
4.5. Use-case: Video conferencing system with central server . 4
4.5.1. Description . . . . . . . . . . . . . . . . . . . . . 4
4.6. Use-case: Hockey game viewer . . . . . . . . . . . . . . . 5
4.6.1. Description . . . . . . . . . . . . . . . . . . . . . 5
4.7. Use-case: Telephony terminal . . . . . . . . . . . . . . . 5
4.7.1. Description . . . . . . . . . . . . . . . . . . . . . 5
5. Requirements . . . . . . . . . . . . . . . . . . . . . . . . . 5
5.1. General . . . . . . . . . . . . . . . . . . . . . . . . . 6
5.2. Browser requirements . . . . . . . . . . . . . . . . . . . 6
5.3. API requirements . . . . . . . . . . . . . . . . . . . . . 7
6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 9
7. Security Considerations . . . . . . . . . . . . . . . . . . . 9
7.1. Introduction . . . . . . . . . . . . . . . . . . . . . . . 9
7.2. Browser Considerations . . . . . . . . . . . . . . . . . . 9
7.3. Web Application Considerations . . . . . . . . . . . . . . 9
8. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 9
9. Change Log . . . . . . . . . . . . . . . . . . . . . . . . . . 9
10. References . . . . . . . . . . . . . . . . . . . . . . . . . . 10
10.1. Normative References . . . . . . . . . . . . . . . . . . . 10
10.2. Informative References . . . . . . . . . . . . . . . . . . 10
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 10
Holmberg, et al. Expires September 8, 2011 [Page 2]
Internet-Draft RTC-Web March 2011
1. Introduction
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.
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 later retrieved.
4.2. Use-case: Simple Video Communication Service
4.2.1. 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
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.
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.
Holmberg, et al. Expires September 8, 2011 [Page 3]
Internet-Draft RTC-Web March 2011
Any session participant can end the session at any time.
One participant has an unreliable internet connection. It sometimes
has packet losses, and is sometimes goes down completely.
One participant is located behind a Network Address Translator (NAT).
4.3. Use-case: Multiparty video communication
4.3.1. 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 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.
Each video stream received is by default displayed in a thumbnail
frame within the browser, but users can change the display size.
4.4. Use-case: Multiparty on-line game with voice communication
4.4.1. 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.
4.5. Use-case: Video conferencing system with central server
4.5.1. Description
An organization uses a video communication system that supports the
establishment of multiparty video sessions using a central conference
server.
The browsers of 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.
Holmberg, et al. Expires September 8, 2011 [Page 4]
Internet-Draft RTC-Web March 2011
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.
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.
4.6. Use-case: Hockey game viewer
4.6.1. 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 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.
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.
4.7. Use-case: Telephony terminal
4.7.1. 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 will be shown
in the same manner as when a mobile phone used.
5. Requirements
Holmberg, et al. Expires September 8, 2011 [Page 5]
Internet-Draft RTC-Web March 2011
5.1. General
This section contains requirements, derived from the use-cases in
section 4.
NOTE: 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.
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 provided to
avoid disturbing echo during conversation.
QUESTION: How much control should be left to the
web application?
----------------------------------------------------------------
F10 The browser MUST support synchronization of
Holmberg, et al. Expires September 8, 2011 [Page 6]
Internet-Draft RTC-Web March 2011
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 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?
----------------------------------------------------------------
5.3. API requirements
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.
----------------------------------------------------------------
Holmberg, et al. Expires September 8, 2011 [Page 7]
Internet-Draft RTC-Web March 2011
A3 The web application MUST be able to control the
local layout 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.
Holmberg, et al. Expires September 8, 2011 [Page 8]
Internet-Draft RTC-Web March 2011
----------------------------------------------------------------
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 describes
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.
7.3. Web Application Considerations
The web application is expected to ensure user consent in sending and
receiving media streams.
8. Acknowledgements
TBD
9. Change Log
[RFC EDITOR NOTE: Please remove this section when publishing]
Changes from draft-holmberg-rtcweb-ucreqs-old
o TBD
10. References
Holmberg, et al. Expires September 8, 2011 [Page 9]
Internet-Draft RTC-Web March 2011
10.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
10.2. Informative References
Authors' Addresses
Christer Holmberg
Ericsson
Hirsalantie 11
Jorvas 02420
Finland
Email: christer.holmberg@ericsson.com
Stefan Hakansson
Ericsson
Laboratoriegrand 11
Lulea 97128
Sweden
Email: stefan.lk.hakansson@ericsson.com
Goran Eriksson
Ericsson
Farogatan 6
Stockholm 16480
Sweden
Email: goran.ap.eriksson@ericsson.com
Holmberg, et al. Expires September 8, 2011 [Page 10]
| PAFTECH AB 2003-2026 | 2026-04-23 19:42:15 |