RTCWEB Working Group C. Holmberg
Internet-Draft S. Hakansson
Intended status: Informational G. Eriksson
Expires: August 16, 2014 Ericsson
February 12, 2014
Web Real-Time Communication Use-cases and Requirements
draft-ietf-rtcweb-use-cases-and-requirements-14.txt
Abstract
This document describes web based real-time communication use-cases.
Requirements on the browser functionality are derived from the use-
cases.
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 August 16, 2014.
Copyright Notice
Copyright (c) 2014 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 August 16, 2014 [Page 1]
Internet-Draft RTC-Web February 2014
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Conventions . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Use-cases . . . . . . . . . . . . . . . . . . . . . . . . . . 3
3.1. Introduction . . . . . . . . . . . . . . . . . . . . . . 3
3.2. Common requirements . . . . . . . . . . . . . . . . . . . 4
3.3. Browser-to-browser use-cases . . . . . . . . . . . . . . 4
3.3.1. Simple Video Communication Service . . . . . . . . . 4
3.3.2. Simple Video Communication Service, NAT/Firewall that
blocks UDP . . . . . . . . . . . . . . . . . . . . . 6
3.3.3. Simple Video Communication Service, Firewall that
only allows traffic via a HTTP Proxy . . . . . . . . 7
3.3.4. Simple Video Communication Service, global service
provider . . . . . . . . . . . . . . . . . . . . . . 7
3.3.5. Simple Video Communication Service, enterprise
aspects . . . . . . . . . . . . . . . . . . . . . . . 8
3.3.6. Simple Video Communication Service, access change . . 9
3.3.7. Simple Video Communication Service, QoS . . . . . . . 9
3.3.8. Simple Video Communication Service with screen
sharing . . . . . . . . . . . . . . . . . . . . . . . 10
3.3.9. Simple Video Communication Service with file exchange 10
3.3.10. Hockey Game Viewer . . . . . . . . . . . . . . . . . 11
3.3.11. Multiparty video communication . . . . . . . . . . . 12
3.3.12. Multiparty on-line game with voice communication . . 13
3.4. Browser - GW/Server use cases . . . . . . . . . . . . . . 15
3.4.1. Telephony terminal . . . . . . . . . . . . . . . . . 15
3.4.2. Fedex Call . . . . . . . . . . . . . . . . . . . . . 15
3.4.3. Video conferencing system with central server . . . . 16
4. Requirements summary . . . . . . . . . . . . . . . . . . . . 17
4.1. General . . . . . . . . . . . . . . . . . . . . . . . . . 17
4.2. Browser requirements . . . . . . . . . . . . . . . . . . 17
5. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 21
6. Security Considerations . . . . . . . . . . . . . . . . . . . 21
6.1. Introduction . . . . . . . . . . . . . . . . . . . . . . 21
6.2. Browser Considerations . . . . . . . . . . . . . . . . . 21
6.3. Web Application Considerations . . . . . . . . . . . . . 22
7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 22
8. Change Log . . . . . . . . . . . . . . . . . . . . . . . . . 22
9. Normative References . . . . . . . . . . . . . . . . . . . . 28
Appendix A. API requirements . . . . . . . . . . . . . . . . . . 28
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 31
1. Introduction
This document presents a few use-cases of web applications that are
executed in a browser and use real-time communication capabilities.
In most of the use-cases all end-user clients are web applications,
Holmberg, et al. Expires August 16, 2014 [Page 2]
Internet-Draft RTC-Web February 2014
but there are some use-cases where at least one of the end-user
clients is of another type (e.g. a mobile phone or a SIP UA).
Based on the use-cases, the document derives requirements related to
browser functionality. These requirements are named "Fn", where n is
an integer, and are listed in conjunction with the use-cases. A
summary is provided in Section 4.2.
This document was developed in an initial phase of the work with
rather minor updates at later stages. It has not really served as a
tool in deciding features or scope for the WGs efforts so far. It is
proposed to be used in a later phase to evaluate the protocols and
solutions developed by the WG.
This document also lists requirements related to the API to be used
by web applications as an appendix. The reason is that the W3C
WebRTC WG has decided to not develop its own use-case/requirement
document, but instead use this document. These requirements are
named "An", where n is an integer, and are described in Appendix A-
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. Use-cases
3.1. Introduction
This section describes web based real-time communication use-cases,
from which requirements are derived.
The following considerations are applicable to all use cases:
o Clients can be on IPv4-only
o Clients can be on IPv6-only
o Clients can be on dual-stack
o Clients can be connected to networks with different throughput
capabilities
o Clients can be on variable-media-quality networks (wireless)
o Clients can be on congested networks
Holmberg, et al. Expires August 16, 2014 [Page 3]
Internet-Draft RTC-Web February 2014
o Clients can be on firewalled networks with no UDP allowed
o Clients can be on networks with a NAT using any type of Mapping
and Filtering behaviors (as described in RFC4787).
3.2. Common requirements
The requirements retrived from the Simple Video Communication Service
use-case (Section 3.3.1) by default apply to all other use-cases, and
are considred common. For each individual use-case, only the
additional requirements are listed.
3.3. Browser-to-browser use-cases
3.3.1. Simple Video Communication Service
3.3.1.1. Description
Two or more users have loaded a video communication web application
into their browsers, provided by the same service provider, and
logged into the service it provides. The web service publishes
information about user login status by pushing updates to the web
application in the browsers. When one online user selects a peer
online user, a 1-1 audiovisual communication session between the
browsers of the two peers is initiated. The invited user might
accept or reject the session.
During session establishment a self-view is displayed, and once the
session has been established the video sent from the remote peer is
displayed in addition to the self-view. During the session, each
user can select to remove and re-insert the self-view as often as
desired. Each user can also change the sizes of his/her two video
displays during the session. Each user can also pause sending of
media (audio, video, or both) and mute incoming media
It is essential that media and data be encrypted, authenticated and
integrity protected on a per-packet basis and that media and data
packets failing the integrity check not be delivered to the
application.
The application gives the users the opportunity to stop it from
exposing the host IP address to the application of the other user.
Any session participant can end the session at any time.
The two users may be using communication devices with different
operating systems and browsers from different vendors.
Holmberg, et al. Expires August 16, 2014 [Page 4]
Internet-Draft RTC-Web February 2014
The web service monitors the quality of the service (focus on quality
of audio and video) the end-users experience.
3.3.1.2. Common 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 and
data to a peer in the presence of NATs.
----------------------------------------------------------------
F3 Transmitted streams and data must be rate
controlled (meaning that the browser must, regardless
of application behavior, reduce send rate when
there is congestion).
----------------------------------------------------------------
F4 The browser must be able to receive, process and
render streams and data ("render" does not
apply for data) from peers.
----------------------------------------------------------------
F5 The browser should be able to render good quality
audio and video even in the presence of
reasonable levels of jitter and packet losses.
----------------------------------------------------------------
F6 The browser must detect when a stream from a
peer is not received anymore
----------------------------------------------------------------
F7 When there are both incoming and outgoing audio
streams, echo cancellation must be made
available to avoid disturbing echo during
conversation.
----------------------------------------------------------------
F8 The browser must support synchronization of
audio and video.
----------------------------------------------------------------
F9 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
----------------------------------------------------------------
F10 The browser must support a baseline audio and
video codec
----------------------------------------------------------------
F11 It must be possible to protect streams and data
from wiretapping [RFC2804].
Holmberg, et al. Expires August 16, 2014 [Page 5]
Internet-Draft RTC-Web February 2014
----------------------------------------------------------------
F12 The browser must enable verification, given
the right circumstances and by use of other
trusted communication, that streams and
data received have not been manipulated by
any party.
----------------------------------------------------------------
F13 The browser must encrypt, authenticate and
integrity protect media and data on a
per-packet basis, and must drop incoming media
and data packets that fail the per-packet
integrity check. In addition, the browser
must support a mechanism for cryptographically
binding media and data security keys to the
user identity (see R-ID-BINDING in [RFC5479]).
----------------------------------------------------------------
F14 The browser must make it possible to set up a
call between two parties without one party
learning the other party's host IP address.
----------------------------------------------------------------
F15 The browser must be able to collect statistics,
related to the transport of audio and video
between peers, needed to estimate quality of
experience.
----------------------------------------------------------------
A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12, A25, A26
3.3.2. Simple Video Communication Service, NAT/Firewall that blocks UDP
3.3.2.1. Description
This use-case is almost identical to the
Simple Video Communication Service use-case (Section 3.3.1). The
difference is that one of the users is behind a NAT/Firewall that
blocks UDP traffic.
3.3.2.2. Additional Requirements
----------------------------------------------------------------
REQ-ID DESCRIPTION
----------------------------------------------------------------
F18 The browser must be able to send streams and
data to a peer in the presence of NATs and
Firewalls that block UDP traffic.
----------------------------------------------------------------
Holmberg, et al. Expires August 16, 2014 [Page 6]
Internet-Draft RTC-Web February 2014
3.3.3. Simple Video Communication Service, Firewall that only allows
traffic via a HTTP Proxy
3.3.3.1. Description
This use-case is almost identical to the
Simple Video Communication Service use-case (Section 3.3.1). The
difference is that one of the users is behind a Firewall that only
allows traffic via a HTTP Proxy.
3.3.3.2. Additional Requirements
----------------------------------------------------------------
REQ-ID DESCRIPTION
----------------------------------------------------------------
F21 The browser must be able to send streams and
data to a peer in the presence of Firewalls that only
allows traffic via a HTTP Proxy, when Firewall policy
allows WebRTC traffic.
----------------------------------------------------------------
3.3.4. Simple Video Communication Service, global service provider
3.3.4.1. Description
This use-case is almost identical to the
Simple Video Communication Service use-case (Section 3.3.1).
What is added is that the service provider is operating over large
geographical areas (or even globally).
Assuming that ICE will be used, this means that the service provider
would like to be able to provide several STUN and TURN servers (via
the app) to the browser; selection of which one(s) to use is part of
the ICE processing. Other reasons for wanting to provide several
STUN and TURN servers include support for IPv4 and IPv6, load
balancing and redundancy.
Note that ICE support being mandatory does not preclude a WebRTC
endpoint from supporting more traversal mechanisms than ICE using
STUN and TURN.
3.3.4.2. Additional Requirements
Holmberg, et al. Expires August 16, 2014 [Page 7]
Internet-Draft RTC-Web February 2014
----------------------------------------------------------------
REQ-ID DESCRIPTION
----------------------------------------------------------------
F19 The browser must be able to use several STUN
and TURN servers
----------------------------------------------------------------
A22
3.3.5. Simple Video Communication Service, enterprise aspects
3.3.5.1. Description
This use-case is similar to the Simple Video Communication Service
use-case (Section 3.3.1).
What is added is aspects when using the service in enterprises. ICE
is assumed in the further description of this use-case.
An enterprise that uses a RTCWEB based web application for
communication desires to audit all RTCWEB based application sessions
used from inside the company towards any external peer. To be able
to do this they deploy a TURN server that straddles the boundary
between the internal and the external network.
The firewall will block all attempts to use STUN with an external
destination unless they go to the enterprise auditing TURN server.
In cases where employees are using RTCWEB applications provided by an
external service provider they still want the traffic to stay inside
their internal network and in addition not load the straddling TURN
server, thus they deploy a STUN server allowing the RTCWEB client to
determine its server reflexive address on the internal side. Thus
enabling cases where peers are both on the internal side to connect
without the traffic leaving the internal network. It must be
possible to configure the browsers used in the enterprise with
network specific STUN and TURN servers. This should be possible to
achieve by auto-configuration methods. The RTCWEB functionality will
need to utilize both network specific STUN and TURN resources and
STUN and TURN servers provisioned by the web application.
3.3.5.2. Additional Requirements
Holmberg, et al. Expires August 16, 2014 [Page 8]
Internet-Draft RTC-Web February 2014
----------------------------------------------------------------
REQ-ID DESCRIPTION
----------------------------------------------------------------
F20 The browser must support the use of STUN and TURN
servers that are supplied by entities other than
the web application (i.e. the network provider).
----------------------------------------------------------------
3.3.6. Simple Video Communication Service, access change
3.3.6.1. Description
This use-case is almost identical to the
Simple Video Communication Service use-case (Section 3.3.1). The
difference is that the user changes network access during the
session.
The communication device used by one of the users has several network
adapters (Ethernet, WiFi, Cellular). The communication device is
accessing the Internet using Ethernet, but the user has to start a
trip during the session. The communication device automatically
changes to use WiFi when the Ethernet cable is removed and then moves
to cellular access to the Internet when moving out of WiFi coverage.
The session continues even though the access method changes.
3.3.6.2. Additional Requirements
----------------------------------------------------------------
REQ-ID DESCRIPTION
----------------------------------------------------------------
F17 The communication session must survive across a
change of the network interface used by the
session
----------------------------------------------------------------
3.3.7. Simple Video Communication Service, QoS
3.3.7.1. Description
This use-case is almost identical to the
Simple Video Communication Service, access change use-case
(Section 3.3.6). The use of Quality of Service (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
Holmberg, et al. Expires August 16, 2014 [Page 9]
Internet-Draft RTC-Web February 2014
enabled. The user is able to take advantage of the QoS support both
when accessing via the residential router and when using cellular.
3.3.7.2. Additional Requirements
----------------------------------------------------------------
REQ-ID DESCRIPTION
----------------------------------------------------------------
F17 The communication session must survive across a
change of the network interface used by the
session
----------------------------------------------------------------
F22 The browser must be able to receive streams and
data from multiple peers concurrently.
----------------------------------------------------------------
3.3.8. Simple Video Communication Service with screen sharing
3.3.8.1. Description
This use-case has the audio and video communication of the
Simple Video Communication Service use-case (Section 3.3.1).
But in addition to this, one of the users can share what is being
displayed on her/his screen with a peer. The user can choose to
share the entire screen, part of the screen (part selected by the
user) or what a selected application displays with the peer.
3.3.8.2. Additional Requirements
----------------------------------------------------------------
REQ-ID DESCRIPTION
----------------------------------------------------------------
F36 The browser must be able to generate streams
using the entire user display, a specific area
of the user's display or the information being
displayed by a specific application.
----------------------------------------------------------------
A21
3.3.9. Simple Video Communication Service with file exchange
3.3.9.1. Description
This use-case has the audio and video communication of the
Simple Video Communication Service use-case (Section 3.3.1).
Holmberg, et al. Expires August 16, 2014 [Page 10]
Internet-Draft RTC-Web February 2014
But in addition to this, the users can send and receive files stored
in the file system of the device used.
3.3.9.2. Additional Requirements
----------------------------------------------------------------
REQ-ID DESCRIPTION
----------------------------------------------------------------
F35 The browser must be able to send reliable
data traffic to a peer browser.
----------------------------------------------------------------
A21, A24
3.3.10. Hockey Game Viewer
3.3.10.1. 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 audiovisual communication session. On the
mobile phone, only the camera facing the talent scout is used. On
the user display of the mobile phone, the video of the club manager
is shown with a picture-in-picture thumbnail of the rear facing
camera (self-view). On the display of the desktop, the video of the
talent scout is shown with a picture-in-picture thumbnail of the
desktop camera (self-view).
When the game is on-going, the talent scout activates the use of the
front facing camera, and that stream is sent to the desktop (the
stream from the rear facing camera continues to be sent all the
time). The video stream captured by the front facing camera (that is
capturing the game) of the mobile phone is shown in a big window on
the desktop screen, with picture-in-picture thumbnails of the rear
facing camera and the desktop camera (self-view). On the display of
the mobile phone the game is shown (front facing camera) with
picture-in-picture thumbnails of the rear facing camera (self-view)
and the desktop camera. As the most important stream in this phase
is the video showing the game, the application used in the talent
scout's mobile sets higher priority for that stream.
Holmberg, et al. Expires August 16, 2014 [Page 11]
Internet-Draft RTC-Web February 2014
3.3.10.2. Additional Requirements
----------------------------------------------------------------
REQ-ID DESCRIPTION
----------------------------------------------------------------
F22 The browser should be able to take advantage
of available capabilities (supplied by network
nodes) to prioritize voice, video and data
appropriately.
----------------------------------------------------------------
F25 The browser must be able to render several
concurrent video streams
----------------------------------------------------------------
A17, A23
3.3.11. Multiparty video communication
3.3.11.1. Description
In this use-case, the Simple Video Communication Service use-case
(Section 3.3.1) 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 the user experience, the web application renders
the audio coming from different particiapants so that it is
experienced to come from different spatial locations. This is done
automatically, but users can change how the different participants
are placed in the (virtual) room. In addition the levels in the
audio signals are adjusted before mixing.
Another feature intended to enhance the use experience is that the
video window that displays the video of the currently speaking peer
is highlighted.
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 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 received streams and be able to spatialize, level adjust and
mix the audio from all received streams locally in the browser. It
also adds the capability to measure the audio level/activity.
Holmberg, et al. Expires August 16, 2014 [Page 12]
Internet-Draft RTC-Web February 2014
3.3.11.2. Additional Requirements
----------------------------------------------------------------
REQ-ID DESCRIPTION
----------------------------------------------------------------
F23 The browser must be able to transmit streams and
data to several peers concurrently.
----------------------------------------------------------------
F24 The browser must be able to receive streams and
data from multiple peers concurrently.
----------------------------------------------------------------
F25 The browser must be able to render several
concurrent video streams
----------------------------------------------------------------
F26 The browser must be able to mix several
audio streams.
----------------------------------------------------------------
F27 The browser must be able to apply spatialization
effects when playing audio streams.
----------------------------------------------------------------
F28 The browser must be able to measure the
voice activity level in audio streams.
----------------------------------------------------------------
F29 The browser must be able to change the
voice activity level in audio streams.
----------------------------------------------------------------
A13, A14, A15, A16
3.3.12. Multiparty on-line game with voice communication
3.3.12.1. Description
This use case is based on the previous one. In this use-case, the
voice part of the multiparty video communication use case is used in
the context of an on-line game. The received voice audio media is
rendered together with game sound objects. For example, the sound of
a tank moving from left to right over the screen must be rendered and
played to the user together with the voice media.
Quick updates of the game state is required, and have higher priority
than the voice.
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
Holmberg, et al. Expires August 16, 2014 [Page 13]
Internet-Draft RTC-Web February 2014
be a file with the sound of the tank; that file could be stored
locally or remotely.
3.3.12.2. Additional Requirements
----------------------------------------------------------------
REQ-ID DESCRIPTION
----------------------------------------------------------------
F22 The browser should be able to take advantage
of available capabilities (supplied by network
nodes) to prioritize voice, video and data
appropriately.
----------------------------------------------------------------
F23 The browser must be able to transmit streams and
data to several peers concurrently.
----------------------------------------------------------------
F24 The browser must be able to receive streams and
data from multiple peers concurrently.
----------------------------------------------------------------
F25 The browser must be able to render several
concurrent video streams
----------------------------------------------------------------
F26 The browser must be able to mix several
audio streams.
----------------------------------------------------------------
F27 The browser must be able to apply spatialization
effects when playing audio streams.
----------------------------------------------------------------
F28 The browser must be able to measure the
voice activity level in audio streams.
----------------------------------------------------------------
F29 The browser must be able to change the
voice activity level in audio streams.
----------------------------------------------------------------
F30 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.
----------------------------------------------------------------
F34 The browser must be able to send short
latency unreliable datagram traffic to a
peer browser [RFC5405].
----------------------------------------------------------------
A13, A14, A15, A16, A17, A18, A23
Holmberg, et al. Expires August 16, 2014 [Page 14]
Internet-Draft RTC-Web February 2014
3.4. Browser - GW/Server use cases
3.4.1. Telephony terminal
3.4.1.1. 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 is used.
Note: With "place and receive calls in the same way as when using a
normal mobile phone" it is meant that you can dial a number, and that
your mobile telephony operator has made available your phone contacts
on line, so they are available and can be clicked to call, and be
used to present the identity of an incoming call. If the callee is
not in your phone contacts the number is displayed. Furthermore,
your call logs are available, and updated with the calls made/
received from the browser. And for people receiving calls made from
the web browser the usual identity (i.e. the phone number of the
mobile phone) will be presented.
3.4.1.2. Additional Requirements
----------------------------------------------------------------
REQ-ID DESCRIPTION
----------------------------------------------------------------
F31 The browser must support an audio media format
(codec) that is commonly supported by existing
telephony services.
----------------------------------------------------------------
F33 The browser must be able to initiate and
accept a media session where the data needed
for establishment can be carried in SIP.
----------------------------------------------------------------
3.4.2. Fedex Call
3.4.2.1. Description
Alice uses her web browser with a service that allows her to call
PSTN numbers. Alice calls 1-800-gofedex. Alice should be able to
hear the initial prompts from the fedex Interactive Voice Responder
(IVR) and when the IVR says press 1, there should be a way for Alice
to navigate the IVR.
Holmberg, et al. Expires August 16, 2014 [Page 15]
Internet-Draft RTC-Web February 2014
3.4.2.2. Additional Requirements
----------------------------------------------------------------
REQ-ID DESCRIPTION
----------------------------------------------------------------
F31 The browser must support an audio media format
(codec) that is commonly supported by existing
telephony services.
----------------------------------------------------------------
F32 There should be a way to navigate
a Dual-tone multi-frequency signaling (DTMF)
based Interactive voice response (IVR) System
----------------------------------------------------------------
3.4.3. Video conferencing system with central server
3.4.3.1. Description
An organization uses a video communication system that supports the
establishment of multiparty video sessions using a central conference
server.
The browser of each participant sends an audio stream (type in terms
of mono, stereo, 5.1, ... depending on the equipment of the
participant) to the central server. The central server mixes the
audio streams (and can in the mixing process naturally add effects
such as spatialization) and sends towards each participant a mixed
audio stream which is played to the user.
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.
All participants are authenticated by the central server, and
authorized to connect to the central server. The participants are
identified to each other by the central server, and the participants
do not have access to each others' credentials such as e-mail
addresses or login IDs.
Note: This use-case adds requirements on support for fast stream
switches F16. There exist several solutions that enable the server
Holmberg, et al. Expires August 16, 2014 [Page 16]
Internet-Draft RTC-Web February 2014
to forward one high resolution and several low resolution video
streams: a) each browser could send a high resolution, but scalable
stream, and the server could send just the base layer for the low
resolution streams, b) each browser could in a simulcast fashion send
one high resolution and one low resolution stream, and the server
just selects or c) each browser sends just a high resolution stream,
the server transcodes into low resolution streams as required.
3.4.3.2. Additional Requirements
----------------------------------------------------------------
REQ-ID DESCRIPTION
----------------------------------------------------------------
F16 The browser must support insertion of reference frames
in outgoing media streams when requested by a peer.
----------------------------------------------------------------
F25 The browser must be able to render several
concurrent video streams
----------------------------------------------------------------
4. Requirements summary
4.1. General
This section contains the requirements on the browser derived from
the use-cases in Section 3.
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.
4.2. Browser requirements
----------------------------------------------------------------
Common, basic 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 and
data to a peer in the presence of NATs.
----------------------------------------------------------------
F3 Transmitted streams and data must be rate
controlled (meaning that the browser must, regardless
Holmberg, et al. Expires August 16, 2014 [Page 17]
Internet-Draft RTC-Web February 2014
of application behavior, reduce send rate when
there is congestion).
----------------------------------------------------------------
F4 The browser must be able to receive, process and
render streams and data ("render" does not
apply for data) from peers.
----------------------------------------------------------------
F5 The browser should be able to render good quality
audio and video even in the presence of
reasonable levels of jitter and packet losses.
----------------------------------------------------------------
F6 The browser must detect when a stream from a
peer is not received anymore
----------------------------------------------------------------
F7 When there are both incoming and outgoing audio
streams, echo cancellation must be made
available to avoid disturbing echo during
conversation.
----------------------------------------------------------------
F8 The browser must support synchronization of
audio and video.
----------------------------------------------------------------
F9 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
----------------------------------------------------------------
F10 The browser must support a baseline audio and
video codec
----------------------------------------------------------------
F11 It must be possible to protect streams and data
from wiretapping [RFC2804].
----------------------------------------------------------------
F12 The browser must enable verification, given
the right circumstances and by use of other
trusted communication, that streams and
data received have not been manipulated by
any party.
----------------------------------------------------------------
F13 The browser must encrypt, authenticate and
integrity protect media and data on a
per-packet basis, and must drop incoming media
and data packets that fail the per-packet
integrity check. In addition, the browser
must support a mechanism for cryptographically
binding media and data security keys to the
user identity (see R-ID-BINDING in [RFC5479]).
----------------------------------------------------------------
Holmberg, et al. Expires August 16, 2014 [Page 18]
Internet-Draft RTC-Web February 2014
F14 The browser must make it possible to set up a
call between two parties without one party
learning the other party's host IP address.
----------------------------------------------------------------
F15 The browser must be able to collect statistics,
related to the transport of audio and video
between peers, needed to estimate quality of
experience.
----------------------------------------------------------------
Requirements related to network and topology
----------------------------------------------------------------
REQ-ID DESCRIPTION
----------------------------------------------------------------
F16 The browser must support insertion of reference frames
in outgoing media streams when requested by a peer.
----------------------------------------------------------------
F17 The communication session must survive across a
change of the network interface used by the
session
----------------------------------------------------------------
F18 The browser must be able to send streams and
data to a peer in the presence of NATs and
Firewalls that block UDP traffic.
----------------------------------------------------------------
F19 The browser must be able to use several STUN
and TURN servers
----------------------------------------------------------------
F20 The browser must support the use of STUN and TURN
servers that are supplied by entities other than
the web application (i.e. the network provider).
----------------------------------------------------------------
F21 The browser must be able to send streams and
data to a peer in the presence of Firewalls that only
allows traffic via a HTTP Proxy, when Firewall policy
allows WebRTC traffic.
----------------------------------------------------------------
F22 The browser should be able to take advantage
of available capabilities (supplied by network
nodes) to prioritize voice, video and data
appropriately.
----------------------------------------------------------------
Requirements related to multiple peers and streams
----------------------------------------------------------------
REQ-ID DESCRIPTION
----------------------------------------------------------------
F23 The browser must be able to transmit streams and
data to several peers concurrently.
----------------------------------------------------------------
Holmberg, et al. Expires August 16, 2014 [Page 19]
Internet-Draft RTC-Web February 2014
F24 The browser must be able to receive streams and
data from multiple peers concurrently.
----------------------------------------------------------------
F25 The browser must be able to render several
concurrent video streams
----------------------------------------------------------------
F26 The browser must be able to mix several
audio streams.
----------------------------------------------------------------
Requirements related to audio processing
----------------------------------------------------------------
REQ-ID DESCRIPTION
----------------------------------------------------------------
F27 The browser must be able to apply spatialization
effects when playing audio streams.
----------------------------------------------------------------
F28 The browser must be able to measure the
voice activity level in audio streams.
----------------------------------------------------------------
F29 The browser must be able to change the
voice activity level in audio streams.
----------------------------------------------------------------
F30 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.
----------------------------------------------------------------
Requirements related to legacy interop
----------------------------------------------------------------
REQ-ID DESCRIPTION
----------------------------------------------------------------
F31 The browser must support an audio media format
(codec) that is commonly supported by existing
telephony services.
----------------------------------------------------------------
F32 There should be a way to navigate
a Dual-tone multi-frequency signaling (DTMF)
based Interactive voice response (IVR) System
----------------------------------------------------------------
F33 The browser must be able to initiate and
accept a media session where the data needed
for establishment can be carried in SIP.
----------------------------------------------------------------
Other requirements
----------------------------------------------------------------
REQ-ID DESCRIPTION
----------------------------------------------------------------
F34 The browser must be able to send short
Holmberg, et al. Expires August 16, 2014 [Page 20]
Internet-Draft RTC-Web February 2014
latency unreliable datagram traffic to a
peer browser [RFC5405].
----------------------------------------------------------------
F35 The browser must be able to send reliable
data traffic to a peer browser.
----------------------------------------------------------------
F36 The browser must be able to generate streams
using the entire user display, a specific area
of the user's display or the information being
displayed by a specific application.
----------------------------------------------------------------
5. IANA Considerations
There are no IANA actions in this document.
6. Security Considerations
6.1. 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.
6.2. 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 and
even completely revoke consent to use device resources such as camera
and microphone.
Holmberg, et al. Expires August 16, 2014 [Page 21]
Internet-Draft RTC-Web February 2014
The browser is expected to provide mechanisms for getting user
consent to use the screen (or a certain part of it) or what a certain
application displays on the screen as source for streams.
The browser is expected to provide mechanisms for informing the user
that the screen, part thereof or an application is serving as a
stream source ("hot").
The browser is expected to provide mechanisms for users to revise and
even completely revoke consent to use the screen, part thereof or an
application is serving as a stream source.
The browser is expected to provide mechanisms in order to assure that
streams are the ones the recipient intended to receive.
The browser is expected to provide mechanisms that allows the users
to verify that the streams received have not be manipulated (F12).
The browser needs to ensure that media is not sent, and that received
media is not rendered, until the associated stream establishment and
handshake procedures with the remote peer have been successfully
finished.
The browser needs to ensure that the stream negotiation procedures
are not seen as Denial Of Service (DOS) by other entities.
6.3. Web Application Considerations
The web application is expected to ensure user consent in sending and
receiving media streams.
7. Acknowledgements
The authors wish to thank Bernard Aboba, Gunnar Hellstrom, Martin
Thomson, Lars Eggert, Matthew Kaufman, Emil Ivov, Eric Rescorla, Eric
Burger, John Leslie, Dan Wing, Richard Barnes, Barry Dingle, Dale
Worley, Ted hardie, Mary Barnes, Dan Burnett, Stephan Wenger, Harald
Alvestrand, Cullen Jennings, Andrew Hutton and everyone else in the
RTCWEB community that have provided comments, feedback, text and
improvement proposals on the document.
8. Change Log
[RFC EDITOR NOTE: Please remove this section when publishing]
Changes from draft-ietf-rtcweb-use-cases-and-requirements-10
Holmberg, et al. Expires August 16, 2014 [Page 22]
Internet-Draft RTC-Web February 2014
o Described that the API requirements are really from a W3C
perspective and are supplied as an appendix in the introduction.
Moved API requirements to an Appendix.
o Removed the "Conventions" section with the key-words and reference
to RFC2119. Also changed uppercase MUST's/SHOULD's to lowercase.
o Added a note on the proposed use of the document to the
introduction.
o Removed the note talking about WS from the "Firewall that only
allows http" use-case.
o Removed the word "Skype" that was used as example in one of the
use-cases.
o Clarified F3 (the req saying the everything the browser sends must
be rate controlled).
o Removed the TBD saying we need to define reasonable levels from
the requirement saying that quality must be good even in presence
of packet losses (F5), and changed "must" to "should" (Based on a
list discussion involving Bernard).
o Removed F6 ("The browser must be able to handle high loss and
jitter levels in a graceful way."), also after a list discussion.
o Clarified F7 (used to say that the browser must support fast
stream switches, now says that reference frames must be inserted
when requested).
o Removed the questions from F9 (echo cancellation), F10
(synchronization), F21 (telephony codec).
o Exchanged "restrictive firewalls" for "limited middleboxes" in F19
(as proposed by Martin).
o Expanded DTMF and IVR in F22 (proposed by Martin)
o Added ref to RFC5405 in F23 (proposed by Lars Eggert).
o Exchanged "service provided" for "web application" in F32.
o Changed the text in 3.2.1 that motivates F36 (new text "It is
essential that media and data be encrypted, authenticated ...
bound to the user identity."); and rewrote F36, included a ref to
RFC5479.
Holmberg, et al. Expires August 16, 2014 [Page 23]
Internet-Draft RTC-Web February 2014
o Changed "quality of service" to "quality of experience" in F38.
o Added F39.
o Used new formulation of A17 (proposed by Martin).
o Updated A20.
o Updated A25.
Changes from draft-ietf-rtcweb-use-cases-and-requirements-09
o Changed "video communication session" to "audiovisual
communication session.
Changes from draft-ietf-rtcweb-use-cases-and-requirements-08
o Changed "eavesdropping" to "wiretapping" and referenced RFC2804.
o Removed informal ref webrtc_req; that document has been abandoned
by the W3C webrtc WG.
o Added use-case where one user is behind a Firewall that only
allows http; derived req. F37.
o Changed F24 slightly; MUST-> SHOULD, inserted "available".
o Added a clause to "Simple video communication service" saying that
the service provider monitors the quality of service, and derived
reqs F38 and A26.
Changes from draft-ietf-rtcweb-use-cases-and-requirements-07
o Added "and data exchange" to 1. Introduction.
o Removed cone and symmetric NAT from 4.1 Introduction, refers to
RFC4787 instead.
o Added text on enabling verification of that the media has not been
manipulated by anyone to use-case "Simple Video Communication
Service", derived req. F35
o Added text on that the browser should reject media (data) that has
been created/injected/modified by non-trusted party, derived req.
F36
Holmberg, et al. Expires August 16, 2014 [Page 24]
Internet-Draft RTC-Web February 2014
o Added text on enabling the app to refrain from revealing IP
address to use-case "Simple Video Communication Service", derived
req. A25
o Added use-case "Simple Video Communication Service with file
exchange", derived reqs F33 and A24
o Added priority of video streams to "Hockey game viewer" use case,
added priority of data to "on-line game use-case", derived reqs
F34 and A23
o In F22, "the IVR" -> "a DTMF based IVR".
o Updated req F23 to clarify that requirements such as NAT
traversal, protection from eavesdropping, rate control applies
also to datagram.
Changes from draft-ietf-rtcweb-use-cases-and-requirements-06
o Renaming of requirements (FaI1 -> F31), (FaI2 -> F32) and (AaI1 ->
A22)
Changes from draft-ietf-rtcweb-use-cases-and-requirements-05
o Added use-case "global service provider", derived reqs associated
with several STUN/TURN servers
o Added use-case "enterprise aspects", derived req associated with
enabling the network provider to supply STUN and TURN servers
o The requirements from the above are ICE specific and labeled
accordingly
o Separated the requirements phrased like "processing such as pan,
mix and render" for audio to be specific reqs on spatialization,
level measurement, level adjustment and mixing (discussed on the
lists in http://www.ietf.org/mail-archive/web/rtcweb/current/
msg01648.html and http://lists.w3.org/Archives/Public/public-
webrtc/2011Sep/0102.html)
o Added use-case on sharing as decided in http://www.ietf.org/mail-
archive/web/rtcweb/current/msg01700.html, derived reqs F30 and A21
o Added the list of common considerations proposed in mail http://
www.ietf.org/mail-archive/web/rtcweb/current/msg01562.html to the
Introduction of the use-case section
Changes from draft-ietf-rtcweb-use-cases-and-requirements-04
Holmberg, et al. Expires August 16, 2014 [Page 25]
Internet-Draft RTC-Web February 2014
o Most changes based on the input from Dan Burnett http://
www.ietf.org/mail-archive/web/rtcweb/current/msg00948.html
o Many editorial changes
o 4.2.1.1 Clarified
o Some clarification added to 4.3.1.1 as a note
o F-requirements updated (see reply to Dan's mail).
o Almost all A-requirements updated to start "The Web API MUST
provide ..."
o A8 removed, A9 rephrased to cover A8 and old A9
o A15 rephrased
o For more details, and discussion, look at the response to Dan's
mail http://www.ietf.org/mail-archive/web/rtcweb/current/
msg01177.html
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 description/list of API requirements, instead
o Reference to W3C webrtc_reqs document for API requirements
Holmberg, et al. Expires August 16, 2014 [Page 26]
Internet-Draft RTC-Web February 2014
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/Firewall
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 communication)
o - Changed the formulation 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
Holmberg, et al. Expires August 16, 2014 [Page 27]
Internet-Draft RTC-Web February 2014
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
9. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC2804] IAB and IESG, "IETF Policy on Wiretapping", RFC 2804, May
2000.
[RFC5405] Eggert, L. and G. Fairhurst, "Unicast UDP Usage Guidelines
for Application Designers", BCP 145, RFC 5405, November
2008.
[RFC5479] Wing, D., Fries, S., Tschofenig, H., and F. Audet,
"Requirements and Analysis of Media Security Management
Protocols", RFC 5479, April 2009.
Appendix A. API requirements
This section contains the requirements on the API derived from the
use-cases in Section 3.
REQ-ID DESCRIPTION
----------------------------------------------------------------
A1 The Web API must provide means for the
application to ask the browser for permission
to use cameras and microphones as input devices.
----------------------------------------------------------------
A2 The Web API must provide means for the web
application to control how streams generated
by input devices are used.
Holmberg, et al. Expires August 16, 2014 [Page 28]
Internet-Draft RTC-Web February 2014
----------------------------------------------------------------
A3 The Web API must provide means for the web
application to control the local rendering of
streams (locally generated streams and streams
received from a peer).
----------------------------------------------------------------
A4 The Web API must provide means for the web
application to initiate sending of
stream/stream components to a peer.
----------------------------------------------------------------
A5 The Web API must provide means for the web
application to control the media format (codec)
to be used for the streams sent to a peer.
NOTE: The level of control depends on whether
the codec negotiation is handled by the browser
or the web application.
----------------------------------------------------------------
A6 The Web API must provide means for the web
application to modify the media format for
streams sent to a peer after a media stream
has been established.
----------------------------------------------------------------
A7 The Web API must provide means for
informing the web application of whether the
establishment of a stream with a peer was
successful or not.
----------------------------------------------------------------
A8 The Web API must provide means for the web
application to mute/unmute a stream or stream
component(s). When a stream is sent to a peer
mute status must be preserved in the stream
received by the peer.
----------------------------------------------------------------
A9 The Web API must provide means for the web
application to cease the sending of a stream
to a peer.
----------------------------------------------------------------
A10 The Web API must provide means for the web
application to cease processing and rendering
of a stream received from a peer.
----------------------------------------------------------------
A11 The Web API must provide means for
informing the web application when a
stream from a peer is no longer received.
----------------------------------------------------------------
A12 The Web API must provide means for
informing the web application when high
Holmberg, et al. Expires August 16, 2014 [Page 29]
Internet-Draft RTC-Web February 2014
loss rates occur.
----------------------------------------------------------------
A13 The Web API must provide means for the web
application to apply spatialization effects to
audio streams.
----------------------------------------------------------------
A14 The Web API must provide means for the web
application to detect the level in audio
streams.
----------------------------------------------------------------
A15 The Web API must provide means for the web
application to adjust the level in audio
streams.
----------------------------------------------------------------
A16 The Web API must provide means for the web
application to mix audio streams.
----------------------------------------------------------------
A17 The Web API must provide a way to identify
streams such that an application is able to
match streams on a sending peer with the same
stream on all receiving peers.
----------------------------------------------------------------
A18 The Web API must provide a mechanism for sending
and receiving isolated discrete chunks of data.
----------------------------------------------------------------
A19 The Web API must provide means for the web
application to indicate the type of audio signal
(speech, audio) for audio stream(s)/stream
component(s).
----------------------------------------------------------------
A20 It must be possible for an initiator or a
responder web application to indicate the types
of media it is willing to accept incoming
streams for when setting up a connection (audio,
video, other). The types of media to be accepted
can be a subset of the types of media the browser
is able to accept.
----------------------------------------------------------------
A21 The Web API must provide means for the
application to ask the browser for permission
to the screen, a certain area on the screen
or what a certain application displays on the
screen as input to streams.
----------------------------------------------------------------
A22 The Web API must provide means for the
application to specify several STUN and/or
TURN servers to use.
----------------------------------------------------------------
Holmberg, et al. Expires August 16, 2014 [Page 30]
Internet-Draft RTC-Web February 2014
A23 The Web API must provide means for the
application to specify the priority to
apply for outgoing streams and data.
----------------------------------------------------------------
A24 The Web API must provide a mechanism for sending
and receiving files.
----------------------------------------------------------------
A25 It must be possible for the application to
instruct the browser to refrain from exposing
the host IP address to the application
----------------------------------------------------------------
A26 The Web API must provide means for the
application to obtain the statistics (related
to transport, and collected by the browser)
needed to estimate quality of service.
----------------------------------------------------------------
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
Holmberg, et al. Expires August 16, 2014 [Page 31]
