One document matched: draft-ietf-rtcweb-use-cases-and-requirements-14.xml
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docName="draft-ietf-rtcweb-use-cases-and-requirements-14.txt"
ipr="trust200902" obsoletes="" submissionType="IETF" updates=""
xml:lang="en">
<front>
<title abbrev="RTC-Web">Web Real-Time Communication Use-cases and
Requirements</title>
<author fullname="Christer Holmberg" initials="C.H." surname="Holmberg">
<organization>Ericsson</organization>
<address>
<postal>
<street>Hirsalantie 11</street>
<code>02420</code>
<city>Jorvas</city>
<country>Finland</country>
</postal>
<email>christer.holmberg@ericsson.com</email>
</address>
</author>
<author fullname="Stefan Hakansson" initials="S.H." surname="Hakansson">
<organization>Ericsson</organization>
<address>
<postal>
<street>Laboratoriegrand 11</street>
<code>97128</code>
<city>Lulea</city>
<country>Sweden</country>
</postal>
<email>stefan.lk.hakansson@ericsson.com</email>
</address>
</author>
<author fullname="Goran AP Eriksson" initials="G.E." surname="Eriksson">
<organization>Ericsson</organization>
<address>
<postal>
<street>Farogatan 6</street>
<code>16480</code>
<city>Stockholm</city>
<country>Sweden</country>
</postal>
<email>goran.ap.eriksson@ericsson.com</email>
</address>
</author>
<date year="2014" />
<area>Transport</area>
<workgroup>RTCWEB Working Group</workgroup>
<keyword>webrtc</keyword>
<keyword>browser</keyword>
<keyword>websocket</keyword>
<keyword>real-time</keyword>
<abstract>
<t>
This document describes web based real-time communication use-cases.
Requirements on the browser functionality are derived from the use-cases.
</t>
</abstract>
</front>
<middle>
<section title="Introduction" toc="default">
<t>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,
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).</t>
<t>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
<xref target="browser_reqs"></xref>.</t>
<t>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.</t>
<t>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 <xref target="api_reqs"></xref>-</t>
</section>
<section title="Conventions" toc="default">
<t>
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT",
"RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described
in BCP 14, RFC 2119 <xref target="RFC2119" pageno="false" format="default" />.
</t>
</section>
<section anchor="use_cases" title="Use-cases" toc="default">
<section title="Introduction" toc="default">
<t>
This section describes web based real-time communication use-cases,
from which requirements are derived.
</t>
<t>The following considerations are applicable to all use cases:<list style="symbols">
<t>Clients can be on IPv4-only</t>
<t>Clients can be on IPv6-only</t>
<t>Clients can be on dual-stack</t>
<t>Clients can be connected to networks with different throughput capabilities</t>
<t>Clients can be on variable-media-quality networks (wireless)</t>
<t>Clients can be on congested networks</t>
<t>Clients can be on firewalled networks with no UDP allowed</t>
<t>Clients can be on networks with a NAT using any type of Mapping and Filtering behaviors (as described in RFC4787).</t>
</list></t>
</section>
<section title="Common requirements" toc="default">
<t>
The requirements retrived from the <xref format="title"
pageno="false" target="simple-video-comm-service"></xref> use-case
(<xref target="simple-video-comm-service"></xref>) by
default apply to all other use-cases, and are considred common. For each
individual use-case, only the additional requirements are listed.
</t>
</section>
<section title="Browser-to-browser use-cases" toc="default">
<section anchor="simple-video-comm-service" title="Simple Video Communication Service" toc="default">
<section title="Description" toc="default">
<t>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.</t>
<t>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</t>
<t> 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.</t>
<t>The application gives the users the opportunity to stop it from
exposing the host IP address to the application of the other user.</t>
<t>Any session participant can end the session at any time.</t>
<t>The two users may be using communication devices with
different operating systems and browsers from different vendors.</t>
<t>The web service monitors the quality of the service (focus on
quality of audio and video) the end-users experience.</t>
</section>
<section title="Common Requirements" toc="default">
<figure align="left" alt="" height="" suppress-title="false" title=""
width="">
<artwork align="left" alt="" height="" name="" type="" width=""
xml:space="preserve"><![CDATA[
----------------------------------------------------------------
REQ-ID DESCRIPTION
----------------------------------------------------------------
F1 The browser must be able to use microphones and
cameras as input devices to generate streams.
----------------------------------------------------------------
F2 The browser must be able to send streams 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].
----------------------------------------------------------------
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.
----------------------------------------------------------------
]]></artwork>
</figure>
<t>A1, A2, A3, A4, A5, A6, A7, A8, A9, A10, A11, A12, A25, A26</t>
</section>
</section>
<section title="Simple Video Communication Service, NAT/Firewall that blocks UDP" toc="default">
<section title="Description" toc="default">
<t>This use-case is almost identical to the <xref format="title"
pageno="false" target="simple-video-comm-service"></xref> use-case
(<xref target="simple-video-comm-service"></xref>). The difference
is that one of the users is behind a NAT/Firewall that blocks UDP
traffic.</t>
</section>
<section title="Additional Requirements" toc="default">
<figure align="left" alt="" height="" suppress-title="false" title=""
width="">
<artwork align="left" alt="" height="" name="" type="" width=""
xml:space="preserve"><![CDATA[
----------------------------------------------------------------
REQ-ID DESCRIPTION
----------------------------------------------------------------
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.
----------------------------------------------------------------
]]></artwork>
</figure>
</section>
</section>
<section title="Simple Video Communication Service, Firewall that only allows traffic via a HTTP Proxy" toc="default">
<section title="Description" toc="default">
<t>This use-case is almost identical to the <xref format="title"
pageno="false" target="simple-video-comm-service"></xref> use-case
(<xref target="simple-video-comm-service"></xref>). The difference is that
one of the users is behind a Firewall that only allows traffic via a HTTP Proxy.</t>
</section>
<section title="Additional Requirements" toc="default">
<figure align="left" alt="" height="" suppress-title="false" title=""
width="">
<artwork align="left" alt="" height="" name="" type="" width=""
xml:space="preserve"><![CDATA[
----------------------------------------------------------------
REQ-ID DESCRIPTION
----------------------------------------------------------------
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.
----------------------------------------------------------------
]]></artwork>
</figure>
</section>
</section>
<section title="Simple Video Communication Service, global service provider" toc="default">
<section title="Description" toc="default">
<t>This use-case is almost identical to the <xref format="title"
pageno="false" target="simple-video-comm-service"></xref> use-case
(<xref target="simple-video-comm-service"></xref>).</t>
<t>What is added is that the service provider is operating over
large geographical areas (or even globally).</t>
<t>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.</t>
<t>Note that ICE support being mandatory does not preclude a
WebRTC endpoint from supporting more traversal mechanisms than
ICE using STUN and TURN.</t>
</section>
<section title="Additional Requirements" toc="default">
<figure align="left" alt="" height="" suppress-title="false" title=""
width="">
<artwork align="left" alt="" height="" name="" type="" width=""
xml:space="preserve"><![CDATA[
----------------------------------------------------------------
REQ-ID DESCRIPTION
----------------------------------------------------------------
F19 The browser must be able to use several STUN
and TURN servers
----------------------------------------------------------------
]]></artwork>
</figure>
<t>A22</t>
</section>
</section>
<section title="Simple Video Communication Service, enterprise aspects" toc="default">
<section title="Description" toc="default">
<t>This use-case is similar to the <xref format="title"
pageno="false" target="simple-video-comm-service"></xref> use-case
(<xref target="simple-video-comm-service"></xref>).</t>
<t>What is added is aspects when using the service in enterprises.
ICE is assumed in the further description of this use-case.</t>
<t>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. </t>
<t>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.</t>
</section>
<section title="Additional Requirements" toc="default">
<figure align="left" alt="" height="" suppress-title="false" title=""
width="">
<artwork align="left" alt="" height="" name="" type="" width=""
xml:space="preserve"><![CDATA[
----------------------------------------------------------------
REQ-ID DESCRIPTION
----------------------------------------------------------------
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).
----------------------------------------------------------------
]]></artwork>
</figure>
</section>
</section>
<section anchor="simple-video-comm-service-access-change" title="Simple Video Communication Service, access change" toc="default">
<section title="Description" toc="default">
<t>This use-case is almost identical to the <xref format="title"
pageno="false" target="simple-video-comm-service"></xref> use-case
(<xref target="simple-video-comm-service"></xref>). The difference
is that the user changes network access during the session.</t>
<t>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.</t>
</section>
<section title="Additional Requirements" toc="default">
<figure align="left" alt="" height="" suppress-title="false" title=""
width="">
<artwork align="left" alt="" height="" name="" type="" width=""
xml:space="preserve"><![CDATA[
----------------------------------------------------------------
REQ-ID DESCRIPTION
----------------------------------------------------------------
F17 The communication session must survive across a
change of the network interface used by the
session
----------------------------------------------------------------
]]></artwork>
</figure>
</section>
</section>
<section title="Simple Video Communication Service, QoS" toc="include">
<section title="Description" toc="default">
<t>This use-case is almost identical to the <xref format="title"
pageno="false" target="simple-video-comm-service-access-change"></xref> use-case
(<xref target="simple-video-comm-service-access-change"></xref>).
The use of Quality of Service (QoS) capabilities is added:</t>
<t>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.</t>
</section>
<section title="Additional Requirements" toc="default">
<figure align="left" alt="" height="" suppress-title="false" title=""
width="">
<artwork align="left" alt="" height="" name="" type="" width=""
xml:space="preserve"><![CDATA[
----------------------------------------------------------------
REQ-ID DESCRIPTION
----------------------------------------------------------------
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.
----------------------------------------------------------------
]]></artwork>
</figure>
</section>
</section>
<section title="Simple Video Communication Service with screen sharing" toc="default">
<section title="Description" toc="default">
<t>This use-case has the audio and video communication of the
<xref format="title" pageno="false"
target="simple-video-comm-service"></xref> use-case (<xref
target="simple-video-comm-service"></xref>).</t>
<t>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.</t>
</section>
<section title="Additional Requirements" toc="default">
<figure align="left" alt="" height="" suppress-title="false" title=""
width="">
<artwork align="left" alt="" height="" name="" type="" width=""
xml:space="preserve"><![CDATA[
----------------------------------------------------------------
REQ-ID DESCRIPTION
----------------------------------------------------------------
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.
----------------------------------------------------------------
]]></artwork>
</figure>
<t>A21</t>
</section>
</section>
<section title="Simple Video Communication Service with file exchange" toc="default">
<section title="Description" toc="default">
<t>This use-case has the audio and video communication of the
<xref format="title" pageno="false"
target="simple-video-comm-service"></xref> use-case (<xref
target="simple-video-comm-service"></xref>).</t>
<t>But in addition to this, the users can send and receive files
stored in the file system of the device used.</t>
</section>
<section title="Additional Requirements" toc="default">
<figure align="left" alt="" height="" suppress-title="false" title=""
width="">
<artwork align="left" alt="" height="" name="" type="" width=""
xml:space="preserve"><![CDATA[
----------------------------------------------------------------
REQ-ID DESCRIPTION
----------------------------------------------------------------
F35 The browser must be able to send reliable
data traffic to a peer browser.
----------------------------------------------------------------
]]></artwork>
</figure>
<t>A21, A24</t>
</section>
</section>
<section title="Hockey Game Viewer" toc="default">
<section title="Description" toc="default">
<t>An ice-hockey club uses an application that enables talent
scouts to, in real-time, show and discuss games and players with
the club manager. The talent scouts use a mobile phone with two
cameras, one front facing and one rear facing.</t>
<t>The club manager uses a desktop, equipped with one camera, for
viewing the game and discussing with the talent scout.</t>
<t>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).</t>
<t>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.</t>
</section>
<section title="Additional Requirements" toc="default">
<figure align="left" alt="" height="" suppress-title="false" title=""
width="">
<artwork align="left" alt="" height="" name="" type="" width=""
xml:space="preserve"><![CDATA[
----------------------------------------------------------------
REQ-ID DESCRIPTION
----------------------------------------------------------------
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
----------------------------------------------------------------
]]></artwork>
</figure>
<t>A17, A23</t>
</section>
</section>
<section title="Multiparty video communication" toc="default">
<section title="Description" toc="default">
<t>In this use-case, the <xref format="title" pageno="false"
target="simple-video-comm-service"></xref> use-case (<xref
target="simple-video-comm-service"></xref>) 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.</t>
<t>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.</t>
<t>Another feature intended to enhance the use experience is that
the video window that displays the video of the currently speaking
peer is highlighted.</t>
<t>Each video stream received is by default displayed in a
thumbnail frame within the browser, but users can change the
display size.</t>
<t>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.</t>
</section>
<section title="Additional Requirements" toc="default">
<figure align="left" alt="" height="" suppress-title="false" title=""
width="">
<artwork align="left" alt="" height="" name="" type="" width=""
xml:space="preserve"><![CDATA[
----------------------------------------------------------------
REQ-ID DESCRIPTION
----------------------------------------------------------------
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.
----------------------------------------------------------------
]]></artwork>
</figure>
<t>A13, A14, A15, A16</t>
</section>
</section>
<section title="Multiparty on-line game with voice communication" toc="default">
<section title="Description" toc="default">
<t>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.</t>
<t>Quick updates of the game state is required, and have higher
priority than the voice.</t>
<t>Note: the difference regarding local audio processing compared
to the "Multiparty video communication" use-case is that other
sound objects than the streams must be possible to be included in
the spatialization and mixing. "Other sound objects" could for
example be a file with the sound of the tank; that file could be
stored locally or remotely.</t>
</section>
<section title="Additional Requirements" toc="default">
<figure align="left" alt="" height="" suppress-title="false" title=""
width="">
<artwork align="left" alt="" height="" name="" type="" width=""
xml:space="preserve"><![CDATA[
----------------------------------------------------------------
REQ-ID DESCRIPTION
----------------------------------------------------------------
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].
----------------------------------------------------------------
]]></artwork>
</figure>
<t>A13, A14, A15, A16, A17, A18, A23</t>
</section>
</section>
</section>
<section title="Browser - GW/Server use cases" toc="default">
<section title="Telephony terminal" toc="default">
<section title="Description" toc="default">
<t>A mobile telephony operator allows its customers to use a web
browser to access their services. After a simple log in the user
can place and receive calls in the same way as when using a normal
mobile phone. When a call is received or placed, the identity is
shown in the same manner as when a mobile phone is used.</t>
<t>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.</t>
</section>
<section title="Additional Requirements" toc="default">
<figure align="left" alt="" height="" suppress-title="false" title=""
width="">
<artwork align="left" alt="" height="" name="" type="" width=""
xml:space="preserve"><![CDATA[
----------------------------------------------------------------
REQ-ID DESCRIPTION
----------------------------------------------------------------
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.
----------------------------------------------------------------
]]></artwork>
</figure>
</section>
</section>
<section title="Fedex Call" toc="default">
<section title="Description" toc="default">
<t>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.</t>
</section>
<section title="Additional Requirements" toc="default">
<figure align="left" alt="" height="" suppress-title="false" title=""
width="">
<artwork align="left" alt="" height="" name="" type="" width=""
xml:space="preserve"><![CDATA[
----------------------------------------------------------------
REQ-ID DESCRIPTION
----------------------------------------------------------------
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
----------------------------------------------------------------
]]></artwork>
</figure>
</section>
</section>
<section title="Video conferencing system with central server" toc="default">
<section title="Description" toc="default">
<t>An organization uses a video communication system that supports
the establishment of multiparty video sessions using a central
conference server.</t>
<t>The 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.</t>
<t>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.</t>
<t>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.</t>
<t>Note: This use-case adds requirements on support for fast
stream switches F16. There exist several solutions that
enable the server to forward one high resolution and several low
resolution video streams: a) each browser could send a high
resolution, but scalable stream, and the server could send just
the base layer for the low resolution streams, b) each browser
could in a simulcast fashion send one high resolution and one low
resolution stream, and the server just selects or c) each browser
sends just a high resolution stream, the server transcodes into
low resolution streams as required.</t>
</section>
<section title="Additional Requirements" toc="default">
<figure align="left" alt="" height="" suppress-title="false" title=""
width="">
<artwork align="left" alt="" height="" name="" type="" width=""
xml:space="preserve"><![CDATA[
----------------------------------------------------------------
REQ-ID DESCRIPTION
----------------------------------------------------------------
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
----------------------------------------------------------------
]]></artwork>
</figure>
</section>
</section>
</section>
</section>
<section title="Requirements summary" toc="default">
<section title="General" toc="default">
<t>This section contains the requirements on the browser derived from the use-cases
in <xref target="use_cases"></xref>.</t>
<t>NOTE: It is assumed that the user applications are executed on a
browser. Whether the capabilities to implement specific browser
requirements are implemented by the browser application, or are
provided to the browser application by the underlying operating
system, is outside the scope of this document.</t>
</section>
<section anchor="browser_reqs" title="Browser requirements"
toc="default">
<figure align="left" alt="" height="" suppress-title="false" title=""
width="">
<artwork align="left" alt="" height="" name="" type="" width=""
xml:space="preserve"><![CDATA[
----------------------------------------------------------------
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
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]).
----------------------------------------------------------------
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.
----------------------------------------------------------------
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
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.
----------------------------------------------------------------
]]></artwork>
</figure>
</section>
</section>
<section title="IANA Considerations" toc="default">
<t>There are no IANA actions in this document.</t>
</section>
<section anchor="sec-security" title="Security Considerations"
toc="default">
<section anchor="sec-security-int" title="Introduction" toc="default">
<t>A malicious web application might use the browser to perform Denial
Of Service (DOS) attacks on NAT infrastructure, or on peer devices.
Also, a malicious web application might silently establish outgoing,
and accept incoming, streams on an already established connection.</t>
<t>Based on the identified security risks, this section will describe
security considerations for the browser and web application.</t>
</section>
<section anchor="sec-security-browser" title="Browser Considerations"
toc="default">
<t>The browser is expected to provide mechanisms for getting user
consent to use device resources such as camera and microphone.</t>
<t>The browser is expected to provide mechanisms for informing the
user that device resources such as camera and microphone are in use
("hot").</t>
<t>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.</t>
<t>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.</t>
<t>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").</t>
<t>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.</t>
<t>The browser is expected to provide mechanisms in order to assure
that streams are the ones the recipient intended to receive.</t>
<t>The browser is expected to provide mechanisms that allows the users
to verify that the streams received have not be manipulated (F12).</t>
<t>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.</t>
<t>The browser needs to ensure that the stream negotiation procedures
are not seen as Denial Of Service (DOS) by other entities.</t>
</section>
<section anchor="sec-security-wepapp"
title="Web Application Considerations" toc="default">
<t>The web application is expected to ensure user consent in sending
and receiving media streams.</t>
</section>
</section>
<section anchor="sec-acks" title="Acknowledgements" toc="default">
<t>
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.
</t>
</section>
<section title="Change Log">
<t>[RFC EDITOR NOTE: Please remove this section when publishing]</t>
<t>Changes from draft-ietf-rtcweb-use-cases-and-requirements-10</t>
<t><list style="symbols">
<t>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. </t>
<t>Removed the "Conventions" section with the key-words and reference to RFC2119.
Also changed uppercase MUST's/SHOULD's to lowercase.</t>
<t>Added a note on the proposed use of the document to the introduction.</t>
<t>Removed the note talking about WS from the "Firewall that only allows http" use-case.</t>
<t>Removed the word "Skype" that was used as example in one of the use-cases.</t>
<t>Clarified F3 (the req saying the everything the browser sends must be rate controlled).</t>
<t>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).</t>
<t>Removed F6 ("The browser must be able to handle high loss and
jitter levels in a graceful way."), also after a list discussion.</t>
<t>Clarified F7 (used to say that the browser must support fast stream switches, now says that reference frames must
be inserted when requested).</t>
<t>Removed the questions from F9 (echo cancellation), F10 (synchronization), F21 (telephony codec).</t>
<t>Exchanged "restrictive firewalls" for "limited middleboxes" in F19 (as proposed by Martin).</t>
<t>Expanded DTMF and IVR in F22 (proposed by Martin)</t>
<t>Added ref to RFC5405 in F23 (proposed by Lars Eggert).</t>
<t>Exchanged "service provided" for "web application" in F32.</t>
<t>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.</t>
<t>Changed "quality of service" to "quality of experience" in F38.</t>
<t>Added F39.</t>
<t>Used new formulation of A17 (proposed by Martin).</t>
<t>Updated A20.</t>
<t>Updated A25.</t>
</list>
</t>
<t>Changes from draft-ietf-rtcweb-use-cases-and-requirements-09</t>
<t><list style="symbols">
<t>Changed "video communication session" to "audiovisual communication session.</t>
</list>
</t>
<t>Changes from draft-ietf-rtcweb-use-cases-and-requirements-08</t>
<t><list style="symbols">
<t>Changed "eavesdropping" to "wiretapping" and referenced RFC2804.</t>
<t>Removed informal ref webrtc_req; that document has been
abandoned by the W3C webrtc WG.</t>
<t> Added use-case where one user is behind a Firewall that only
allows http; derived req. F37.</t>
<t>Changed F24 slightly; MUST-> SHOULD, inserted "available".</t>
<t>Added a clause to "Simple video communication service" saying
that the service provider monitors the quality of service, and
derived reqs F38 and A26.</t>
</list>
</t>
<t>Changes from draft-ietf-rtcweb-use-cases-and-requirements-07</t>
<t><list style="symbols">
<t>Added "and data exchange" to 1. Introduction.</t>
<t>Removed cone and symmetric NAT from 4.1 Introduction, refers
to RFC4787 instead.</t>
<t>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</t>
<t>Added text on that the browser should reject media (data) that has
been created/injected/modified by non-trusted party, derived req. F36</t>
<t>Added text on enabling the app to refrain from revealing IP
address to use-case "Simple Video Communication Service", derived
req. A25</t>
<t>Added use-case "Simple Video Communication Service with
file exchange", derived reqs F33 and A24</t>
<t>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</t>
<t> In F22, "the IVR" -> "a DTMF based IVR".</t>
<t>Updated req F23 to clarify that requirements such as NAT traversal,
protection from eavesdropping, rate control applies also to datagram.</t>
</list></t>
<t>Changes from draft-ietf-rtcweb-use-cases-and-requirements-06</t>
<t><list style="symbols">
<t>Renaming of requirements (FaI1 -> F31), (FaI2 -> F32) and (AaI1 -> A22)</t>
</list></t>
<t>Changes from draft-ietf-rtcweb-use-cases-and-requirements-05</t>
<t><list style="symbols">
<t>Added use-case "global service provider", derived reqs associated
with several STUN/TURN servers</t>
<t>Added use-case "enterprise aspects", derived req associated with
enabling the network provider to supply STUN and TURN servers</t>
<t>The requirements from the above are ICE specific and labeled
accordingly</t>
<t>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)</t>
<t>Added use-case on sharing as decided in
http://www.ietf.org/mail-archive/web/rtcweb/current/msg01700.html,
derived reqs F30 and A21</t>
<t>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</t>
</list></t>
<t>Changes from draft-ietf-rtcweb-use-cases-and-requirements-04</t>
<t><list style="symbols">
<t>Most changes based on the input from Dan Burnett
http://www.ietf.org/mail-archive/web/rtcweb/current/msg00948.html</t>
<t>Many editorial changes</t>
<t>4.2.1.1 Clarified</t>
<t>Some clarification added to 4.3.1.1 as a note</t>
<t>F-requirements updated (see reply to Dan's mail).</t>
<t>Almost all A-requirements updated to start "The Web API MUST
provide ..."</t>
<t>A8 removed, A9 rephrased to cover A8 and old A9</t>
<t>A15 rephrased</t>
<t>For more details, and discussion, look at the response to Dan's
mail
http://www.ietf.org/mail-archive/web/rtcweb/current/msg01177.html</t>
</list></t>
<t>Changes from draft-ietf-rtcweb-use-cases-and-requirements-03</t>
<t><list style="symbols">
<t>Editorials</t>
<t>Changed when the self-view is displayed in 4.2.1.1, and added
words about allowing users to remove and re-insert it.</t>
<t>Clarified 4.2.6.1</t>
<t>Removed the "mono" stuff from 4.2.7.1</t>
<t>Added that communication should not be possible to eavesdrop to
most use cases - and req. F17</t>
<t>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.</t>
<t>Re-inserted API requirements after discussion in the W3C webrtc
WG. (Re-phrased A15 and added A18 compared to version -02).</t>
</list></t>
<t>Changes from draft-ietf-rtcweb-use-cases-and-requirements-02</t>
<t><list style="symbols">
<t>Removed description/list of API requirements, instead</t>
<t>Reference to W3C webrtc_reqs document for API requirements</t>
</list></t>
<t>Changes from draft-ietf-rtcweb-ucreqs-01</t>
<t><list style="symbols">
<t>Changed Intended status to Information</t>
<t>Changed "Ipr" to "trust200902"</t>
<t>Added use case "Simple video communication service, NAT/Firewall that
blocks UDP", and derived new req F26</t>
<t>Added use case "Distributed Music Band" and derived new req
A17</t>
<t>Added F24 as requirement derived from use case "Simple video
communication service with inter-operator calling"</t>
<t>Added section "Additional use cases"</t>
<t>Added text about ID handling to multiparty with central server
use case</t>
<t>Re-phrased A1 slightly</t>
</list></t>
<t>Changes from draft-ietf-rtcweb-ucreqs-00</t>
<t><list style="symbols">
<t>- 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)</t>
<t>- Changed the formulation of F19</t>
<t>- Removed the requirement on an API for DTMF</t>
<t>- 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"</t>
<t>- (see
http://www.ietf.org/mail-archive/web/rtcweb/current/msg00227.html
for more details)</t>
<t>-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.</t>
</list>Changes from draft-holmberg-rtcweb-ucreqs-01 <list
style="symbols">
<t>- Draft name changed to draft-ietf-rtcweb-ucreqs</t>
<t>- Use-case grouping introduced</t>
<t>- Additional use-cases added</t>
<t>- Additional reqs added (derived from use cases): F19-F25,
A16-A17</t>
</list></t>
<t>Changes from draft-holmberg-rtcweb-ucreqs-00 <list style="symbols">
<t>- Mapping between use-cases and requirements added (Harald
Alvestrand, 090311)</t>
<t>- Additional security considerations text (Harald Alvestrand,
090311)</t>
<t>- Clarification that user applications are assumed to be executed
by a browser (Ted Hardie, 080311)</t>
<t>- Editorial corrections and clarifications</t>
</list></t>
</section>
</middle>
<back>
<references title="Normative References">
<?rfc include="reference.RFC.2119"?>
<?rfc include="reference.RFC.2804"?>
<?rfc include="reference.RFC.5405"?>
<?rfc include="reference.RFC.5479"?>
</references>
<section anchor="api_reqs" title="API requirements" toc="default">
<t>This section contains the requirements on the API derived from the use-cases
in <xref target="use_cases"></xref>.</t>
<figure align="left" alt="" height="" suppress-title="false" title=""
width="">
<artwork align="left" alt="" height="" name="" type="" width=""
xml:space="preserve"><![CDATA[
REQ-ID DESCRIPTION
----------------------------------------------------------------
A1 The Web 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.
----------------------------------------------------------------
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
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.
----------------------------------------------------------------
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.
----------------------------------------------------------------
]]></artwork>
</figure>
</section>
</back>
</rfc>| PAFTECH AB 2003-2026 | 2026-04-24 04:11:19 |