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Versions: (draft-holmberg-rtcweb-ucreqs) 00
01 02 03 04 05 06 07 08 09 10 11 12
13 14 15 16 RFC 7478
RTCWEB Working Group C. Holmberg
Internet-Draft S. Hakansson
Intended status: Informational G. Eriksson
Expires: February 29, 2012 Ericsson
August 28, 2011
Web Real-Time Communication Use-cases and Requirements
draft-ietf-rtcweb-use-cases-and-requirements-03.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 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 February 29, 2012.
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.
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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 . . . . . . . . . . . . 6
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 . . . . . . . . . . . . . . 8
4.3.1. Telephony terminal . . . . . . . . . . . . . . . . . . 8
4.3.2. Fedex Call . . . . . . . . . . . . . . . . . . . . . . 8
4.3.3. Video conferencing system with central server . . . . 9
5. Requirements . . . . . . . . . . . . . . . . . . . . . . . . . 10
5.1. General . . . . . . . . . . . . . . . . . . . . . . . . . 10
5.2. Browser requirements . . . . . . . . . . . . . . . . . . . 10
6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 12
7. Security Considerations . . . . . . . . . . . . . . . . . . . 12
7.1. Introduction . . . . . . . . . . . . . . . . . . . . . . . 12
7.2. Browser Considerations . . . . . . . . . . . . . . . . . . 13
7.3. Web Application Considerations . . . . . . . . . . . . . . 13
8. Additional use cases . . . . . . . . . . . . . . . . . . . . . 13
9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 14
10. Change Log . . . . . . . . . . . . . . . . . . . . . . . . . . 14
11. References . . . . . . . . . . . . . . . . . . . . . . . . . . 15
11.1. Normative References . . . . . . . . . . . . . . . . . . . 15
11.2. Informative References . . . . . . . . . . . . . . . . . . 16
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 16
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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 requirements related to the browser are named "Fn" and are
described in section Section 5.2
The requirements related to the API are named "An" and are described
in the external document [webrtc_reqs]
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 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.
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 sizes of the video displays during the session. The users can
also pause sending of media (audio, video, or both), and mute
incoming media.
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, 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, 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
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 byt one of the users have several
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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, 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, 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
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
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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, 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 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.
4.2.6.2. Derived Requirements
F1, F2, F3, F4, F5, F6, F8, F9, F10, F14
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.
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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.
Note: What this uses 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, 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
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.
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 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, F20
A1, A2, A3, A4, A5, A7, A8, A9, A10, A11, A12, A13, A14, A15, A16
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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
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 will be shown
in the same manner as when a mobile phone used.
4.3.1.2. Derived Requirements
F1, F2, F3, F4, F5, F6, F8, F9, F10, 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
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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 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 audio stream.
Each participant sends two video streams in a simulcast fashion
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.
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. It also introduces simulcast, but no concrete
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requirement is put for this.
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
5.1. General
This section contains the browser requirements, derived from the use-
cases in section 4. For the API requirements refer to [webrtc_reqs].
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.
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.
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----------------------------------------------------------------
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 there should be a way to navigate
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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.
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.
----------------------------------------------------------------
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.
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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.
8. Additional use cases
Several additional use cases have been discusses. 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
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.
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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
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.
10. Change Log
[RFC EDITOR NOTE: Please remove this section when publishing]
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"
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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
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.
Holmberg, et al. Expires February 29, 2012 [Page 15]
Internet-Draft RTC-Web August 2011
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
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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