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<!DOCTYPE html>
<html lang="en" dir="ltr">
<head>
<meta name="generator" content="HTML Tidy for HTML5 for Linux/x86 version 5.3.9">
<meta charset="utf-8">
<link href="webrtc.css" rel="stylesheet">
<title>WebRTC Extended Use Cases</title>
<script class="remove" src="https://www.w3.org/Tools/respec/respec-w3c"></script>
<script src="respec-config.js" class="remove"></script>
</head>
<body>
<section id="abstract">
<p>
This document describes an extended set of use cases motivating the
development of additional WebRTC APIs, as well as the requirements derived
from those use cases.
</p>
</section>
<section id="sotd"></section>
<section id="overview*">
<h2>Scope and Motivation</h2>
<p>
To motivate the development of WebRTC, the IETF RTCWEB WG developed [[?RFC7478]].
This document describes extended use cases motivating the development of additional
WebRTC APIs and the requirements deriving from those use cases. The use cases fall
into one of two categories: enhancements to use cases already covered in [[?RFC7478]],
and new use cases which are not supported in WebRTC [[?WEBRTC]] without extensions.
</p>
</section>
<section id="existingusecases*">
<h2>Existing Use Cases</h2>
<p>The uses cases in this section improve upon use cases described in [[?RFC7478]].</p>
<section id="multipartygame*">
<h3>Multiparty online game with voice communications</h3>
<p>
[[?RFC7478]] Section 2.3.12 describes a use case involving a multiparty online game
with voice communications. In these scenarios, reducing time to join the game and
receive media is important. To minimize this, ICE enhancements are desirable, such as
the ability to control candidate gathering and pruning. Also, allowing a participant
to broadcast a configuration to a “room” abstraction (maintained on a server), with
other room participants responding back directly, avoiding a separate discovery step,
minimizes conference establishment time. Also, managing audio quality and latency in
a fair manner between multiple connections prevents queue buildup. Supporting this
enhancement adds the following requirements:
</p>
<table class="simple">
<thead>
<tr>
<th>Requirement ID</th>
<th>Description</th>
</tr>
</thead>
<tbody>
<tr>
<td>N01</td>
<td>The user agent can control candidate gathering
and pruning, limiting the networks on which candidates
are gathered, the types of candidates, etc.</td>
</tr>
<tr>
<td>N02</td>
<td>The user agent must be capable of establishing multiple
connections to peers without generating a separate
configuration ("offer") for each connection prior to
establishment.</td>
</tr>
<tr>
<td>N03</td>
<td>Congestion control must be able to manage audio
quality and latency in a fair manner between multiple
connections.</td>
</tr>
</tbody>
</table>
<p>Experience: This use case has been implemented by a gaming service utilizing [[?ORTC]].</p>
References:
<ol>
<li><a href="https://github.com/w3c/ortc/issues/54">ORTC Issue 54</a></li>
<li><a href="https://github.com/w3c/ortc/issues/603">ORTC Issue 603</a></li>
</ol>
</section>
<section id="mobility*">
<h3>Mobile calling service</h3>
<p>
[[?RFC7478]] Section 2.3.6 describes a simple communications service where the user changes access network
during the session. This use case is enhanced by being able to ring multiple endpoints simultaneously, as
well as to re-route media over an alternate path (potentially taking network cost into account) without
need for signaling.
</p>
<p>
An additional enhancement is to provide management of the user experience at both ends of a call during
interuptions to the media flow caused by other activities taking higher priority on the smartphone.
</p>
<table class="simple">
<thead>
<tr>
<th>Requirement ID</th>
<th>Description</th>
</tr>
</thead>
<tbody>
<tr>
<td>N02</td>
<td>The user agent must be capable of establishing multiple
connections to peers without generating a separate
configuration ("offer") for each connection prior to
establishment.</td>
</tr>
<tr>
<td>N04</td>
<td>The ICE agent must be able to maintain multiple
candidate pairs and move traffic between them.</td>
</tr>
<tr>
<td>N05</td>
<td>The ICE agent must be able to take the network
cost into account when considering re-routing.</td>
</tr>
<tr>
<td>N30</td>
<td>The user agent must provide the ability to re-establish media
after an interruption.</td>
</tr>
<tr>
<td>N31</td>
<td>The user agent must provide notification of a media interruption
caused by the OS (e.g. GSM incoming call) (c.f. on hold music).
and support the use of a datachannel to inform the remote peer of the interruption.</td>
</tr>
<tr>
<td>N32</td>
<td>The user agent must provide the ability to 'park' a connection such
that it can be retrieved and continued by a newly loaded page to prevent
accidental 'browsing away' from dropping a call irretrievably.</td>
</tr>
</tbody>
</table>
<p>References:</p>
<ol>
<li><a href="https://lists.w3.org/Archives/Public/public-webrtc/2018May/0079.html">Mailing list proposal</a></li>
<li><a href="https://lists.w3.org/Archives/Public/public-webrtc/2018May/0019.html">Mailing list proposal</a></li>
<li><a href="https://github.com/w3c/ortc/issues/583">ORTC Issue 583</a></li>
</ol>
<p>Experience: This use case has been implemented by multiple native smartphone apps with call-kit integration.</p>
</section>
<section id="videoconferencing*">
<h3>Video Conferencing with a Central Server</h3>
<p>[[?RFC7478]] Section 2.4.3.1 describes a use case involving Multiparty Video Communications with
a central conferencing server. In such a use case, clients with disparate capabilities such as
differing bandwidth availability,
screen size and maximum displayable frame rate may participate in the same conference.
In such a situation it is advantageous to support Scalable Video Coding (SVC).
Encoding with temporal scalability is supported by several browsers today and is utilized by most
centralized conferencing services.</p>
<p>It is expected that spatial scalability (supported by VP9 and AV1) will become more popular with time.
In this use case, if the desired video codec is known beforehand and participants are muted by default
(as in a very large meeting), it is desirable to allow new participants to start receiving immediately,
without negotiation. Supporting this enhancement adds the following requirements:</p>
<table class="simple">
<thead>
<tr>
<th>Requirement ID</th>
<th>Description</th>
</tr>
</thead>
<tbody>
<tr>
<td>N06</td>
<td>The user agent must be able to encode and decode
video utilizing temporal scalability and (if supported
by the chosen codec) spatial scalability.</td>
</tr>
<tr>
<td>N07</td>
<td>A user agent can receive audio/video without requiring
construction of a corresponding sender object.</td>
</tr>
<tr>
<td>N08</td>
<td>It is possible to select the sending and/or
receiving codec as well as rtcp parameters and
header extensions without negotiation.</td>
</tr>
<tr>
<td>N09</td>
<td>The user agent must be able to control
robustness (RTX, RED, FEC) applied to individual
simulcast and SVC layers.</td>
</tr>
<tr>
<td>N24</td>
<td>Content Security Policy (CSP) support for WebRTC.</td>
</tr>
</tbody>
</table>
<p>This use case has been implemented by conferencing services utilizing [[?ORTC]],
as well as proprietary additions to [[?WEBRTC]].</p>
</section>
</section>
<section id="newusecases*">
<h2>New Use Cases</h2>
<p>Several new uses cases relate to scenarios that cannot be supported in [[?WEBRTC]] without extensions.</p>
<section id="filesharing*">
<h3>File Sharing</h3>
<p>Participants in a mesh exchange large files without disruption to audio/video sessions.
It is also possible for a participant to send a large file to a user who is not currently online.
Supporting this use case adds the following requirements:</p>
<table class="simple">
<thead>
<tr>
<th>Requirement ID</th>
<th>Description</th>
</tr>
</thead>
<tbody>
<tr>
<td>N10</td>
<td>It must be possible for the user agent to initiate
transfer of a large file with a single API operation.</td>
</tr>
<tr>
<td>N11</td>
<td>The application must be able to signal backpressure
(flow control) when receiving data. It must also
receive a backpressure signal when sending data.</td>
</tr>
<tr>
<td>N12</td>
<td>It must be possible for the user agent to transfer
data utilizing a congestion control algorithm
that does not compete aggressively with
audio/video communications.</td>
</tr>
<tr>
<td>N13</td>
<td>It must be possible to support data exchange
in a web, service, or shared worker. Support for
service workers allows the page to issue a fetch()
which can be resolved in the service worker.</td>
</tr>
<tr>
<td>N24</td>
<td>Content Security Policy (CSP) support for WebRTC.</td>
</tr>
</tbody>
</table>
<p>References:</p>
<ol>
<li><a href="https://lists.w3.org/Archives/Public/public-webrtc/2018May/0079.html">Mailing list discussion</a></li>
<li><a href="https://lists.w3.org/Archives/Public/public-webrtc/2018May/0082.html">Mailing list discussion</a></li>
</ol>
</section>
<section id="low-latency-streaming">
<h3>Low latency streaming</h3>
<section id="game-streaming">
<h4>Game streaming</h4>
<p>
Game streaming involves the sending of audio and video (potentially at high resolution and framerate)
to the recipient, along with data being sent in the opposite direction. Games can be streamed either
from a cloud service (client/server), or from a peer game console (P2P). It is highly desirable that
media flow without interruption, and that game players not reveal their location to each other. Even in
the case of games streamed from a cloud service, it can be desirable for players to be able to communicate
with each other directly (via chat, audio or video).
</p>
<table class="simple">
<thead>
<tr>
<th>Requirement ID</th>
<th>Description</th>
</tr>
</thead>
<tbody>
<tr>
<td>N15</td>
<td>The application must be able to take steps to ensure a low and consistent
latency for audio, video and data under varying network conditions. This may
include tweaking of transport parameters for both media and data.</td>
</tr>
<tr>
<td>N36</td>
<td>An application that is only receiving but not sending media or data can operate
efficiently without access to camera or microphone.</td>
</tr>
<tr>
<td>N37</td>
<td>It must be possible for the user agent's receive pipeline to process
video at high resolution and framerate (e.g. by controlling hardware
acceleration if necessary).</td>
</tr>
<tr>
<td>N38</td>
<td>The application must be able to control the jitter buffer and rendering
delay. This requirement is addressed by jitterBufferTarget, defined in
[[?WebRTC-Extensions]] Section 6.</td>
</tr>
</tbody>
</table>
<p>Experience: Microsoft's Xbox Cloud Gaming and NVIDIA's GeForce NOW are examples of this use case, with media
transported using RTP or RTCDataChannel.</p>
</section>
<section id="auction">
<h4>Ultra Low latency Broadcast with Fanout</h4>
<p>
There are streaming applications that require large scale as well as ultra low latency
such as auctions. Live audio, video and data is sent to thousands of recipients.
Limited interactivity may be supported, such as capturing audio/video from
auction bidders. Both the media senders and receivers may be behind a NAT.
Peer-to-peer relays are used to improve scalability, with ingestion, distribution
and fanout requiring unreliable/unordered transport to reduce latency, and support
for retransmission and forward error correction to provide robustness. In this use
case, low latency is more important than media quality, so that low latency congestion
control algorithms are required, and latency-enducing effects such as head of line
blocking are to be avoided. Reception of media via events is problematic, due to
lack of coupling between the event loop and the receive window.
</p>
<table class="simple">
<thead>
<tr>
<th>Requirement ID</th>
<th>Description</th>
</tr>
</thead>
<tbody>
<tr>
<td>N15</td>
<td>The application must be able to take steps to ensure a low and consistent
latency for audio, video and data under varying network conditions. This may
include tweaking of transport parameters for both media and data.</td>
</tr>
<tr>
<td>N39</td>
<td>A user-agent must be able to forward media received from a peer
to another peer. Applications require access to encoded chunk metadata
as well as information from the RTP header to provide for timing, media
configuration and congestion control. This includes a mechanism
for a relaying peer to obtain a bandwidth estimate.</td>
</tr>
</tbody>
</table>
<p>Experience: |pipe| and Dolby are examples of this use case, with media
transported via RTP.</p>
</section>
</section>
<section id="iot*">
<h3>Internet of Things</h3>
<p>An IoT sensor maintains a long-term connection and seeks to minimize power consumption.
Some of the sensor’s data may need to be sent reliable and ordered while other sensors may
provide data that can be sent unreliable and unordered or in a partially reliable manner.
Such IoT sensors may also produce realtime video or audio data for remote users which are
privacy sensitive and may only be accessed by selected devices. This use case adds the
following requirements:</p>
<table class="simple">
<thead>
<tr>
<th>Requirement ID</th>
<th>Description</th>
</tr>
</thead>
<tbody>
<tr>
<td>N14</td>
<td>The application must be able to minimize ICE
connectivity checks.</td>
</tr>
<tr>
<td>N15</td>
<td>The application must be able to take steps to ensure a low and consistent
latency for audio, video and data under varying network conditions. This may
include tweaking of transport parameters for both media and data.</td>
</tr>
<tr>
<td>N16</td>
<td>It must be possible to send arbitrary data
reliable, unreliable or partially reliable with
a specific maximum number of retransmissions
or a specific maximum timeout.</td>
</tr>
<tr>
<td>N17</td>
<td>It must be possible to send arbitrary data
ordered or unordered.</td>
</tr>
<tr>
<td>N24</td>
<td>Content Security Policy (CSP) support for WebRTC.</td>
</tr>
<tr>
<td>N33</td>
<td>A pre-existing peers must be able to be (re) establish a connection
without access to external services in the event of the local
network becoming isolated from the wider network. Without
compromising e2e security but possibly leveraging pre-shared tokens from a previous connection.</td>
</tr>
</tbody>
</table>
<p>Reference</p>
<a href="https://lists.w3.org/Archives/Public/public-webrtc/2018May/0079.html">Mailing list discussion</a>
<p>Experience: Building a respiration monitor that works locally during an internet outage
but is also available remotely when the internet connection returns.</p>
</section>
<section id="decent">
<h3>Decentralized messaging</h3>
<p>New decentralized applications provide P2P services and client-server services for consumption in a browser.</p>
<p>The differences in transport layer semantics make it difficult to share code between the two modes.</p>
<p class="note">This use case has not completed a Call for Consensus (CfC).</p>
<table class="simple">
<thead>
<tr>
<th>Requirement ID</th>
<th>Description</th>
</tr>
</thead>
<tbody>
<tr>
<td>N34</td>
<td>Ability to intercept the fetch API and service it over a P2P link.
One way to do this would be to support data exchange in service workers
which can already intercept fetch.</td>
</tr>
</tbody>
</table>
<p>Experience: Both |pipe| and [Matrix] have implemented systems of this sort.</p>
</section>
<section id="vr*">
<h3>Virtual Reality Gaming</h3>
<p>A virtual reality gaming service utilizing a centralized conferencing server wants
to synchronize data with media, using an existing Selective Forwarding Unit (SFU)
to distribute the data. This use case adds the following requirements:</p>
<table class="simple">
<thead>
<tr>
<th>Requirement ID</th>
<th>Description</th>
</tr>
</thead>
<tbody>
<tr>
<td>N23</td>
<td>The user agent must be able to send data synchronized
with audio and video.</td>
</tr>
<tr>
<td>N24</td>
<td>Content Security Policy (CSP) support for WebRTC.</td>
</tr>
</tbody>
</table>
<p>References:</p>
<a href="https://lists.w3.org/Archives/Public/public-webrtc/2018May/0063.html">Mailing list discussion</a>
</section>
<section id="funnyhats*">
<h3>Funny Hats</h3>
<p>A communications service that manipulates captured media prior to encoding and after
decoding to provide effects including:</p>
<ol>
<li>Captioning</li>
<li>Transcription</li>
<li>Language translation</li>
<li>Funny hats</li>
<li>Background removal or blurring</li>
<li>In-browser compositing</li>
<li>Voice effects</li>
<li>Stress detection</li>
</ol>
<p>This use case requires manipulation of raw media from both local and remote sources. Since
media processing can be CPU intensive, enabling it to occur off the main thread is important,
as is enabling the processing to take advantage of the GPU. This use case adds the following
requirements:</p>
<p>This usecase also requires that user agent provide non-discriminatory implementations of
facetracking and body tracking algorithms that can be efficiently used
by the application. This however is outside the scope of this document.</p>
<table class="simple">
<thead>
<tr>
<th>Requirement ID</th>
<th>Description</th>
</tr>
</thead>
<tbody>
<tr>
<td>N18</td>
<td>The application must be able to obtain raw media
from the capture device in desired formats.</td>
</tr>
<tr>
<td>N19</td>
<td>The application must be able to insert processed
frames into the outgoing media path.</td>
</tr>
<tr>
<td>N20</td>
<td>The application must be able to obtain decoded
media from the remote party.</td>
</tr>
<tr>
<td>N21</td>
<td>It must be possible to efficiently share media
between the main thread and worker threads.</td>
</tr>
<tr>
<td>N22</td>
<td>It must be possible to do efficient media manipulation
in worker threads.</td>
</tr>
<tr>
<td>N24</td>
<td>Content Security Policy (CSP) support for WebRTC.</td>
</tr>
</tbody>
</table>
<p>References:</p>
<ol>
<li><a href="https://lists.w3.org/Archives/Public/public-webrtc/2018May/0037.html">Mailing list discussion</a></li>
<li><a href="https://lists.w3.org/Archives/Public/public-webrtc/2018Jun/0006.html">Mailing list discussion</a></li>
<li><a href="https://ai.googleblog.com/2016/11/enhance-raisr-sharp-images-with-machine.html">Sharper Image Research</a></li>
</ol>
</section>
<section id="no-trust-webex">
<h3>Don't Pown My Video Conferencing </h3>
<p>
Cloud video conferencing systems have no need to be able to access
the cleartext media and text flowing through their servers.
Some of these conferencing services desire to be able to promote trust
by explicitly showing they do not have access to contents of their
users' calls. They are trusted to connect the right people to the
conference and to route the packets but they are not trusted to access
the audio and video media or text in the call.
</p>
<p>Solutions to this problem fall into two major categories: one where
the JavaScript comes from a source trusted to see the media contents,
and one where it does not.
</p>
<section id="untrusted*">
<h4>Untrusted JavaScript Cloud Conferencing</h4>
<p>
There are many cases where a system such as WebEx is trusted to
connect the members of a conference but has no need to access the
contents of the conference. This is true of the majority of
conferencing systems on the web today. Just to highlight the
scope of this requirement, there are more minutes of WebRTC
that are used in conferences where the servers have no need
to access the contents (e.g. where audio is forwarded rather than
mixed) than any other use of WebRTC audio by orders of magnitude.
This is one of the primary use case for WebRTC audio and accounts
for billions of minutes per month of potential use of WebRTC.
</p>
<p>
In this use case, the JavaScript comes from the operator of the
conference bridge. The isolated media features of WebRTC can
prevent the JavaScript from accessing the media and the identity
features are used to provide a user interface that allows the
user to know it connected to the correct conference. The goal is
for the end users to be able to see the contents, but the web
service that provides the JS and the media switching bridges
and Selective Forwarding Units (SFUs) cannot access the contents
(audio, video, text). The browser may choose to reveal some
metadata, such as the audio power level, to the media server,
in order to support functions like speaker switching.
</p>
<p>
For small groups (fewer than 20 participants) the SFU could also
run within the browser, further reducing the dependency on costly
centralized servers with management functions running within a
web or service worker.
</p>
<p>
A possible solution this problem is the browser to negotiate
end-to-end encryption keys which are not revealed to the
JavaScript.
</p>
<p>
Security requirements relating to this use case are discussed
in [[?MLS-ARCH]], and include the following:
</p>
<table class="simple">
<thead>
<tr>
<th>Requirement ID</th>
<th>Description</th>
</tr>
</thead>
<tbody>
<tr>
<td>N13</td>
<td>It must be possible to support data exchange
in a web, service, or shared worker. Support for
service workers allows the page to issue a fetch()
which can be resolved in the service worker.</td>
</tr>
<tr>
<td>N25</td>
<td>Only current group members can receive media or
text sent to the group.</td>
</tr>
<tr>
<td>N26</td>
<td>A group member cannot send media or text that
appears to be from another group member.</td>
</tr>
<tr>
<td>N27</td>
<td>The conference server must not have access to
cleartext media or text or to the identity of
group members.</td>
</tr>
<tr>
<td>N28</td>
<td>Perfect Forward Secrecy (FCS): access to encrypted
traffic as well as all current keying material does
not compromise the secrecy of media or text older
than the oldest key of a compromised client.</td>
</tr>
<tr>
<td>N29</td>
<td>Post Compromise Security (PCS). Protection against
past or future device compromise.</td>
</tr>
<tr>
<td>N35</td>
<td>A group member can encrypt and send copies of the realtime encoded media directly
to multiple group members without re-encoding for each recipient (to reduce resource usage).</td>
</tr>
<tr><td>note that the requirements from the Funny Hats usecase are also required here.</td></tr>
</tbody>
</table>
</section>
</section>
<section id="one-way-media">
<h3>One-way media</h3>
<section id="live-encoded-media">
<h3>Live encoded non-WebRTC media</h3>
<p>
There are use cases where it is desirable to transmit live
encoded media from a non-WebRTC source over an RTP connection
to a WebRTC-compatible endpoint. An example is traffic cameras
that serve video over HTTP using "long poll" or similar mechanisms,
but do not support WebRTC.
</p>
<p>
The source may permit dynamic reconfiguration of its resolution or frame
rate in order to produce an outgoing video stream with the desired
characteristics.
</p>
<p class="note">This use case has completed a Call for Consensus (CfC) [[?CFC-One-Way]] but has unresolved issues.</p>
<table class="simple">
<thead>
<tr>
<th>Requirement ID</th>
<th>Description</th>
</tr>
</thead>
<tbody>
<tr>
<td>N40</td>
<td>An application can create an outgoing WebRTC connection without
activating an encoder.</td>
</tr>
<tr>
<td>N41</td>
<td>An application can create encoded video frames from encoded data
and metadata, and enqueue them on an outgoing WebRTC connection</td>
</tr>
<tr>
<td>N42</td>
<td>The WebRTC connection can generate signals indicating the desired
bandwidth, and surface those to the application.</td>
</tr>
</tbody>
</table>
</section>
<section id="stored-encoded-media">
<h3>Transmitting stored encoded media</h3>
<p>
There are use cases with stored pre-encoded media where transmission as part of a
WebRTC RTP session is desirable. These include:
<ul>
<li>
"Wait signals" or "your message is important to us, please stay on the line", inserted
prior to switching to a live interactive stream.
</li>
<li>
Insertion of announcements or alarm signals in otherwise live streams.
</li>
<li>
Insertion of static content (such as profile pictures) when the sender temporarily disables a camera.
</li>
</ul>
</p>
<p>
The pre-encoded media may be available in multiple bandwidths, and switching between them may be
possible at certain points in the media.
</p>
<p class="note">This use case has completed a Call for Consensus (CfC) [[?CFC-One-Way]] but has unresolved issues.</p>
<table class="simple">
<thead>
<tr>
<th>Requirement ID</th>
<th>Description</th>
</tr>
</thead>
<tbody>
<tr>
<td>N41</td>
<td>An application can create encoded video frames from encoded data
and metadata, and enqueue them on an outgoing WebRTC connection</td>
</tr>
<tr>
<td>N42</td>
<td>The WebRTC connection can generate signals indicating the desired
bandwidth, and surface those to the application.</td>
</tr>
<tr>
<td>N43</td>
<td>The application can modify metadata on outgoing frames so that
they fit smoothly within the expected sequence of timestamps and
sequence numbers.</td>
</tr>
<tr>
<td>N44</td>
<td>The application can signal the WebRTC encoder when resuming live
transmission in such a way that generated frames fit smoothly within
the expected sequence of timestamps and sequence numbers.
</td>
</tr>
</tbody>
</table>
</section>
<section id="decode-encoded-media">
<h3>Decoding pre-encoded media</h3>
<p>
There are use cases when we have pre-encoded media (either dynamically generated or stored)
that we wish to process in the same way as one processes media coming in over a PeerConnection.
This will typically involve decoding with a WebRTC decoder and generating a MediaStreamTrack.
</p>
<p>
Some use cases that need this functionality are:
<ul>
<li>
Integration of media-over-datachannels with media-over-RTP (uniform treatment)
</li>
<li>
Feeding media coming from a DRM enforcement system into a WebRTC media pipeline
</li>
<li>
File playback using the WebRTC media pipeline
</li>
</ul>
</p>
<p>
Alternative implementations for these use cases may involve decoding data using WebCodecs
and generating a MediaStreamTrack using VideoTrackGenerator. However, this means that Javascript
will have to handle buffers containing raw media, which may not be optimal for speedy processing.
</p>
<p class="note">This use case has completed a Call for Consensus (CfC) [[?CFC-One-Way]] but has unresolved issues.</p>
<table class="simple">
<thead>
<tr>
<th>Requirement ID</th>
<th>Description</th>
</tr>
</thead>
<tbody>
<tr>
<td>N45</td>
<td>An application can create an incoming WebRTC connection to accept
frames as if they were coming in over RTP, without creating
an RTP transprort.</td>
</tr>
<tr>
<td>N46</td>
<td>An application can create encoded video frames from encoded data
and metadata, and enqueue them on an incoming WebRTC connection.</td>
</tr>
<tr>
<td>N47</td>
<td>The WebRTC connection can generate signals indicating demands
for keyframes, and surface those to the application.</td>
</tr>
</tbody>
</table>
</section>
</section>
</section>
<section id="requirements*">
<h3>Requirements Summary</h3>
<p>This section summarizes the requirements arising from
the use-cases included in this document.</p>
<table class="simple">
<thead>
<tr>
<th>Requirement ID</th>
<th>Description</th>
</tr>
</thead>
<tbody>
<tr id="N01">
<td>N01</td>
<td>The user agent can control candidate gathering
and pruning, limiting the networks on which candidates
are gathered, the types of candidates, etc.</td>
</tr>
<tr id="N02">
<td>N02</td>
<td>The user agent must be capable of establishing multiple
connections to peers without generating a separate
configuration ("offer") for each connection prior to
establishment.</td>
</tr>
<tr id="N03">
<td>N03</td>
<td>Congestion control must be able to manage audio
quality and latency in a fair manner between multiple
connections.</td>
</tr>
<tr id="N04">
<td>N04</td>
<td>The ICE agent must be able to maintain multiple
candidate pairs and move traffic between them.</td>
</tr>
<tr id="N05">
<td>N05</td>
<td>The ICE agent must be able to take the network
cost into account when considering re-routing.</td>
</tr>
<tr id="N06">
<td>N06</td>
<td>The user agent must be able to encode and decode
video utilizing temporal scalability and (if supported
by the chosen codec) spatial scalability.</td>
</tr>
<tr id="N07">
<td>N07</td>
<td>A user agent can receive audio/video without requiring
construction of a corresponding sender object.</td>
</tr>
<tr id="N08">
<td>N08</td>
<td>It is possible to select the sending and/or
receiving codec as well as rtcp parameters and
header extensions without negotiation.</td>
</tr>
<tr id="N09">
<td>N09</td>
<td>The user agent must be able to control
robustness (RTX, RED, FEC) applied to individual
simulcast and SVC layers.</td>
</tr>
<tr id="N10">
<td>N10</td>
<td>It must be possible for the user agent to initiate
transfer of a large file with a single API operation.</td>
</tr>
<tr id="N11">
<td>N11</td>
<td>The application must be able to signal backpressure
(flow control) when receiving data. It must also
receive a backpressure signal when sending data.</td>
</tr>
<tr id="N12">
<td>N12</td>
<td>It must be possible for the user agent to transfer
data utilizing a congestion control algorithm
that does not compete aggressively with
audio/video communications.</td>
</tr>
<tr id="N13">
<td>N13</td>
<td>It must be possible to support data exchange
in a web, service, or shared worker. Support for
service workers allows the page to issue a fetch()
which can be resolved in the service worker.</td>
</tr>
<tr id="N14">
<td>N14</td>
<td>The application must be able to minimize ICE
connectivity checks.</td>
</tr>
<tr id="N15">
<td>N15</td>
<td>The application must be able to take steps to ensure a low and consistent
latency for audio, video and data under varying network conditions. This may
include tweaking of transport parameters for both media and data.</td>
</tr>
<tr id="N16">
<td>N16</td>
<td>It must be possible to send arbitrary data
reliable, unreliable or partially reliable with
a specific maximum number of retransmissions
or a specific maximum timeout.</td>
</tr>
<tr id="N17">
<td>N17</td>
<td>It must be possible to send arbitrary data
ordered or unordered.</td>
</tr>
<tr id="N18">
<td>N18</td>
<td>The application must be able to obtain raw media
from the capture device in desired formats.</td>
</tr>
<tr id="N19">
<td>N19</td>
<td>The application must be able to insert processed
frames into the outgoing media path.</td>
</tr>
<tr id="N20">
<td>N20</td>
<td>The application must be able to obtain decoded
media from the remote party.</td>
</tr>
<tr id="N21">
<td>N21</td>
<td>It must be possible to efficiently share media
between the main thread and worker threads.</td>
</tr>
<tr id="N22">
<td>N22</td>
<td>It must be possible to do efficient media
manipulation in worker threads.</td>
</tr>
<tr id="N23">
<td>N23</td>
<td>The user agent must be able to send data synchronized
with audio and video.</td>
</tr>
<tr id="N24">
<td>N24</td>
<td>Content Security Policy (CSP) support for WebRTC.</td>
</tr>
<tr id="N25">
<td>N25</td>
<td>Only current group members can receive media or
text sent to the group.</td>
</tr>
<tr id="N26">
<td>N26</td>
<td>A group member cannot send media or text that
appears to be from another group member.</td>
</tr>
<tr id="N27">
<td>N27</td>
<td>The conference server must not have access to
cleartext media or text or to the identity of
group members.</td>
</tr>
<tr id="N28">
<td>N28</td>
<td>Perfect Forward Secrecy (FCS): access to encrypted
traffic as well as all current keying material does
not compromise the secrecy of media or text older
than the oldest key of a compromised client.</td>
</tr>
<tr id="N29">
<td>N29</td>
<td>Post Compromise Security (PCS). Protection against
past or future device compromise.</td>
</tr>
</tr id="N30">
<td>N30</td>
<td>The user agent must provide the ability to re-establish
media after an interruption.</td>
</tr>
</tr id="N31">
<td>N31</td>
<td>The user agent must provide notification of a media interruption
caused by the OS (e.g. GSM incoming call) (c.f. on hold music).
and support the use of a datachannel to inform the remote peer
of the interruption.</td>
</tr>
</tr id="N32">
<td>N32</td>
<td>The user agent must provide the ability to 'park' a connection such
that it can be retrieved and continued by a newly loaded page to prevent
accidental 'browsing away' from dropping a call irretrievably.</td>
</tr id="N33">
<td>N33</td>
<td>Pre-existing peers must be able to be (re-)establish a connection
without access to external services in the event of the local
network becoming isolated from the wider network. Without
compromising e2e security but possibly leveraging pre-shared tokens
from a previous connection.</td>
</tr>
</tr id="N34">
<td>N34</td>
<td>Ability to intercept the fetch API and service it over a P2P link.
One way to do this would be to support data channels in Service Workers
which can already intercept fetch.</td>
</tr>
</tr id="N35">
<td>N35</td>
<td>A group member can encrypt and send copies of the realtime encoded media directly
to multiple group members without re-encoding for each recipient (to reduce resource usage).</td>