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<?xml version="1.0" encoding="utf-8"?>
<!--
To publish this document, see instructions in README
-->
<!DOCTYPE html>
<html lang="en">
<head>
<title>WebRTC 1.0: Real-time Communication Between Browsers</title>
<meta content="text/html; charset=us-ascii" http-equiv="Content-Type">
<script class="remove" src="http://www.w3.org/Tools/respec/respec-w3c-common"
type="text/javascript">
// // keep this comment //
</script>
<script class="remove" src="webrtc.js" type="text/javascript">
// // keep
this comment //
</script>
</head>
<body>
<section id="abstract">
<p>This document defines a set of ECMAScript APIs in WebIDL to allow media
to be sent to and received from another browser or device implementing the
appropriate set of real-time protocols. This specification is being
developed in conjunction with a protocol specification developed by the
IETF RTCWEB group and an API specification to get access to local media
devices developed by the Media Capture Task Force.</p>
</section>
<section id="sotd">
<p>This document is neither complete nor stable, and as such is not yet
suitable for commercial implementation. However, early experimentation is
encouraged. The API is based on preliminary work done in the WHATWG. The
Web Real-Time Communications Working Group expects this specification to
evolve significantly based on:</p>
<ul>
<li>The outcome of ongoing exchanges in the companion RTCWEB group at
IETF to define the set of protocols that, together with this document,
will enable real-time communications in Web browsers.</li>
<li>Privacy issues that arise when exposing local capabilities and local
streams.</li>
<li>Technical discussions within the group.</li>
<li>Experience gained through early experimentations.</li>
<li>Feedback received from other groups and individuals.</li>
</ul>
</section>
<section class="informative" id="intro">
<h2>Introduction</h2>
<p>There are a number of facets to video-conferencing in HTML covered by
this specification:</p>
<ul>
<li>Connecting to remote peers using NAT-traversal technologies such as
ICE, STUN, and TURN.</li>
<li>Sending the locally-produced streams to remote peers and receiving
streams from remote peers.</li>
<li>Sending arbitrary data directly to remote peers.</li>
</ul>
<p>This document defines the APIs used for these features. This
specification is being developed in conjunction with a protocol
specification developed by the <a href=
"http://datatracker.ietf.org/wg/rtcweb/">IETF RTCWEB group</a> and an API
specification to get access to local media devices
[[!GETUSERMEDIA]]developed by the <a href=
"http://www.w3.org/2011/04/webrtc/">Media Capture Task Force</a>. An
overview of the system can be found in [[RTCWEB-OVERVIEW]] and
[[RTCWEB-SECURITY]].</p>
</section>
<section id="conformance">
<p>This specification defines conformance criteria that apply to a single
product: the <dfn>user agent</dfn> that implements the interfaces that it
contains.</p>
<p>Conformance requirements phrased as algorithms or specific steps may be
implemented in any manner, so long as the end result is equivalent. (In
particular, the algorithms defined in this specification are intended to be
easy to follow, and not intended to be performant.)</p>
<p>Implementations that use ECMAScript to implement the APIs defined in
this specification must implement them in a manner consistent with the
ECMAScript Bindings defined in the Web IDL specification [[!WEBIDL]], as
this specification uses that specification and terminology.</p>
</section>
<section>
<h2>Terminology</h2>
<p>The <code><a href=
"http://dev.w3.org/html5/spec/webappapis.html#eventhandler">EventHandler</a></code>
interface represents a callback used for event handlers as defined in
[[!HTML5]].</p>
<p>The concepts <dfn><a href=
"http://dev.w3.org/html5/spec/webappapis.html#queue-a-task">queue a
task</a></dfn> and <dfn><a href=
"http://dev.w3.org/html5/spec/webappapis.html#fire-a-simple-event">fires a
simple event</a></dfn> are defined in [[!HTML5]].</p>
<p>The terms <dfn>event</dfn>, <dfn><a href=
"http://dev.w3.org/html5/spec/webappapis.html#event-handlers">event
handlers</a></dfn> and <dfn><a href=
"http://dev.w3.org/html5/spec/webappapis.html#event-handler-event-type">event
handler event types</a></dfn> are defined in [[!HTML5]].</p>
<p>The terms <dfn>MediaStream</dfn>, <dfn>MediaStreamTrack</dfn>,
<dfn>Constraints</dfn>, and <dfn>Consumer</dfn> are defined in
[[!GETUSERMEDIA]].</p>
</section>
<section>
<h2>Peer-to-peer connections</h2>
<section>
<h3>Introduction</h3>
<p>An <code><a>RTCPeerConnection</a></code> allows two users to
communicate directly, browser to browser. Communications are coordinated
via a signaling channel which is provided by unspecified means, but
generally by a script in the page via the server, e.g. using
<code>XMLHttpRequest</code>.</p>
</section>
<section>
<h3>Configuration</h3>
<section>
<h4>RTCConfiguration Type</h4>
<dl class="idl" title="dictionary RTCConfiguration">
<dt>sequence<RTCIceServer> iceServers</dt>
<dd>
<p>An array containing URIs of servers available to be used by ICE,
such as STUN and TURN server.</p>
</dd>
<dt>RTCIceTransports iceTransports = "all"</dt>
<dd>
<p>Indicates which candidates the ICE engine is allowed to use.</p>
</dd>
<dt>DOMString peerIdentity</dt>
<dd>
<p>Sets the <a href="#target-peer-identity">target peer
identity</a> for the <a>RTCPeerConnection</a>. The
<a>RTCPeerConnection</a> will establish a connection to a remote
peer unless it can be successfully authenticated with the provided
name.</p>
</dd>
</dl>
</section>
<section>
<h4>RTCIceServer Type</h4>
<dl class="idl" title="dictionary RTCIceServer">
<dt>(DOMString or sequence<DOMString> urls</dt>
<dd>
<p>STUN or TURN URI(s) as defined in [[!RFC7064]] and [[!RFC7065]]
or other URI types.</p>
</dd>
<dt>DOMString username</dt>
<dd>
<p>If this <code><a>RTCIceServer</a></code> object represents a
TURN server, then this attribute specifies the username to use with
that TURN server.</p>
</dd>
<dt>DOMString credential</dt>
<dd>
<p>If this <code><a>RTCIceServer</a></code> object represents a
TURN server, then this attribute specifies the credential to use
with that TURN server.</p>
</dd>
</dl>
<p>In network topologies with multiple layers of NATs, it is desirable
to have a STUN server between every layer of NATs in addition to the
TURN servers to minimize the peer to peer network latency.</p>
<p>An example array of RTCIceServer objects is:</p>
<p><code>[ { "urls": "stun:stun1.example.net" }, { "urls":
"turn:turn.example.org", "username": "user", "credential": "myPassword"
} ]</code></p>
</section>
<section>
<h4>RTCIceTransports Enum</h4>
<dl class="idl" title="enum RTCIceTransports">
<dt>none</dt>
<dd>The ICE engine MUST not send or receive any packets at this
point.</dd>
<dt>relay</dt>
<dd>The ICE engine MUST only use media relay candidates such as
candidates passing through a TURN server. This can be used to reduce
leakage of IP addresses in certain use cases.</dd>
<dt>all</dt>
<dd>The ICE engine may use any type of candidates when this value is
specified.</dd>
</dl>
</section>
<section>
<h4>Offer/Answer Options</h4>
<p>These dictionaries describe the options that can be used to control
the offer/answer creation process.</p>
<dl class="idl" title="dictionary RTCOfferOptions">
<dt>long offerToReceiveVideo</dt>
<dd>
<p>In some cases, an <code>RTCPeerConnection</code> may wish to
receive video but not send any video. The
<code>RTCPeerConnection</code> needs to know if it should signal to
the remote side whether it wishes to receive video or not. This
option allows an application to indicate its preferences for the
number of video streams to receive when creating an offer.</p>
</dd>
<dt>long offerToReceiveAudio</dt>
<dd>
<p>In some cases, an <code>RTCPeerConnection</code> may wish to
receive audio but not send any audio. The
<code>RTCPeerConnection</code> needs to know if it should signal to
the remote side whether it wishes to receive audio. This option
allows an application to indicate its preferences for the number of
audio streams to receive when creating an offer.</p>
</dd>
<dt>boolean voiceActivityDetection = true</dt>
<dd>
<p>Many codecs and system are capable of detecting "silence" and
changing their behavior in this case by doing things such as not
transmitting any media. In many cases, such as when dealing with
emergency calling or sounds other than spoken voice, it is
desirable to be able to turn off this behavior. This option allows
the application to provide information about whether it wishes this
type of processing enabled or disabled.</p>
</dd>
<dt>boolean iceRestart = false</dt>
<dd>
<p>When the value of this dictionary member is true, the generated
description will have ICE credentials that are different from the
current credentials (as visible in the
<code><a>localDescription</a></code> attribute's SDP). Applying the
generated description will restart ICE.</p>
<p>When the value of this dictionary member is false, and the
<code><a>localDescription</a></code> attribute has valid ICE
credentials, the generated description will have the same ICE
credentials as the current value from the
<code><a>localDescription</a></code> attribute.</p>
</dd>
</dl>
<dl class="idl" title="enum RTCIdentityOption">
<dt>yes</dt>
<dd>An identity MUST be requested.</dd>
<dt>no</dt>
<dd>No identity is to be requested.</dd>
<dt>ifconfigured</dt>
<dd>The value "ifconfigured" means that an identity will be requested
if either the user has configured an identity in the browser or if
the <code>setIdentityProvider()</code> call has been made in
JavaScript. As this is the default value, an identity will be
requested if and only if the user has configured an IdP in some
way.</dd>
</dl>
</section>
</section>
<section>
<h3>RTCPeerConnection Interface</h3>
<p>The general operation of the RTCPeerConnection is described in
[[!RTCWEB-JSEP]].</p>
<section>
<h4>Operation</h4>
<p>Calling <code>new <a>RTCPeerConnection</a>(<var>configuration</var>
)</code> creates an <code><a>RTCPeerConnection</a></code> object.</p>
<p>The <var>configuration</var> has the information to find and access
the servers used by ICE. There may be multiple servers of each type and
any TURN server also acts as a STUN server.</p>
<p>An <code><a>RTCPeerConnection</a></code> object has an associated
<dfn id="rtcpeerconnection-ice-agent">ICE agent</dfn> [[!ICE]],
RTCPeerConnection signaling state, ICE gathering state, and ICE
connection state. These are initialized when the object is created.</p>
<p>When the <dfn id=
"dom-peerconnection"><code>RTCPeerConnection()</code></dfn> constructor
is invoked, the user agent MUST run the following steps:</p>
<ol>
<li>
<p>Validate the <code><a>RTCConfiguration</a></code> argument by
running the steps defined by the <a href=
"#dom-peerconnection-updateice">updateIce()</a> method.</p>
</li>
<li>
<p>Let <var>connection</var> be a newly created
<code><a>RTCPeerConnection</a></code> object.</p>
</li>
<li>
<p>Create an ICE Agent as defined in [[!ICE]] and let
<var>connection</var>'s <code>RTCPeerConnection</code> ICE Agent be
that ICE Agent and provide it the the <a href=
"#ice-servers-list">ICE servers list</a>. The ICE Agent will
proceed with gathering as soon as the <a href=
"#ice-transports-setting">ICE transports setting</a> is not set to
<code>none</code>. At this point the ICE Agent does not know how
many ICE components it needs (and hence the number of candidates to
gather), but it can make a reasonable assumption such as 2. As the
<code>RTCPeerConnection</code> object gets more information, the
ICE Agent can adjust the number of components.</p>
</li>
<li>
<p>Set <var>connection</var>'s <a href=
"#dom-peerconnection-signaling-state"><code>RTCPeerConnection</code>
signalingState</a> to <code>stable</code>.</p>
</li>
<li>
<p>Set <var>connection</var>'s <a href=
"#dom-peerconnection-ice-connection-state"><code>RTCPeerConnection</code>
ice connection state</a> to <code>new</code>.</p>
</li>
<li>
<p>Set <var>connection</var>'s <a href=
"#dom-peerconnection-ice-gathering-state"><code>RTCPeerConnection</code>
ice gathering state</a> to <code>new</code>.</p>
</li>
<li>
<p>Initialize an internal variable to represent a queue of
<code>operations</code> with an empty set.</p>
</li>
<li>
<p>Return <var>connection</var>.</p>
</li>
</ol>
<p>Once the RTCPeerConnection object has been initialized, for every
call to <code>createOffer</code>, <code>setLocalDescription</code>,
<code>createAnswer</code> and <code>setRemoteDescription</code>;
execute the following steps:</p>
<ol>
<li>
<p>Append an object representing the current call being handled
(i.e. function name and corresponding arguments) to the
<code>operations</code> array.</p>
</li>
<li>
<p>If the length of the <code>operations</code> array is exactly 1,
execute the function from the front of the queue
asynchronously.</p>
</li>
<li>
<p>When the asynchronous operation completes (either successfully
or with an error), remove the corresponding object from the
<code>operations</code> array. After removal, if the array is
non-empty, execute the first object queued asynchronously and
repeat this step on completion.</p>
</li>
</ol>
<p>The general idea is to have only one among <code>createOffer</code>,
<code>setLocalDescription</code>, <code>createAnswer</code> and
<code>setRemoteDescription</code> executing at any given time. If
subsequent calls are made while one of them is still executing, they
are added to a queue and processed when the previous operation is fully
completed. It is valid, and expected, for normal error handling
procedures to be applied.</p>
<p>Additionally, during the lifetime of the RTCPeerConnection object,
the following procedures are followed when an ICE event occurs:</p>
<ol>
<li>
<p>If the <a href=
"#dom-peerconnection-ice-gathering-state"><code>RTCPeerConnection</code>
ice gathering state</a> is <code>new</code> and the <a href=
"#ice-transports-setting">ICE transports setting</a> is not set to
<code>none</code>, the user agent MUST queue a task to start
gathering ICE addresses and set the <a href=
"#dom-peerconnection-ice-gathering-state">ice gathering state</a>
to <code>gathering</code>.</p>
</li>
<li>
<p>If the ICE Agent has found one or more candidate pairs for each
MediaStreamTrack that forms a valid connection, the ICE connection
state is changed to "connected".</p>
</li>
<li>
<p>When the ICE Agent finishes checking all candidate pairs, if at
least one connection has been found for each MediaStreamTrack, the
<a href=
"#dom-peerconnection-ice-connection-state"><code>RTCPeerConnection</code>
ice connection state</a> is changed to "completed"; otherwise
"failed".</p>
</li>
</ol>
<p>When the ICE Agent needs to notify the script about the candidate
gathering progress, the user agent must queue a task to run the
following steps:</p>
<ol>
<li>
<p>Let <var>connection</var> be the
<code><a>RTCPeerConnection</a></code> object associated with this
ICE Agent.</p>
</li>
<li>
<p>If <var>connection</var>'s <a href=
"#dom-peerconnection-signaling-state"><code>RTCPeerConnection</code>
signalingState</a> is <code>closed</code>, abort these steps.</p>
</li>
<li>
<p>If the intent of the ICE Agent is to notify the script that:</p>
<ul>
<li>
<p>A new candidate is available.</p>
<p>Add the candidate to <var>connection</var>'s
<code><a>localDescription</a></code> and create a
<code><a>RTCIceCandidate</a></code> object to represent the
candidate. Let <var>newCandidate</var> be that object.</p>
</li>
<li>
<p>The gathering process is done.</p>
<p>Set <var>connection</var>'s <a href=
"#dom-peerconnection-ice-gathering-state">ice gathering
state</a> to <code>completed</code> and let
<var>newCandidate</var> be null.</p>
</li>
</ul>
</li>
<li>
<p>Fire a icecandidate event named <code><a href=
"#event-icecandidate">icecandidate</a></code> with
<var>newCandidate</var> at <var>connection</var>.</p>
</li>
</ol>
<p>User agents negotiate the codec resolution, bitrate, and other media
parameters. It is RECOMMENDED that user agents initially negotiate for
the maximum resolution of a video stream. For streams that are then
rendered (using a <code>video</code> element), it is RECOMMENDED that
user agents renegotiate for a resolution that matches the rendered
display size.</p>
<p>The word "components" in this context refers to an RTP media flow
and does not have anything to do with how [[ICE]] uses the term
"component".</p>
<p>When a user agent has reached the point where a
<code><a>MediaStreamTrack</a></code> can be created to represent an incoming
component, the user agent MUST run the following steps:</p>
<ol>
<li>
<p>Let <var>connection</var> be the
<code><a>RTCPeerConnection</a></code> expecting this media.</p>
</li>
<li>
<p>Run the following steps to create a track
representing the incoming component:</p>
<ol>
<li>
<p>Create a <code><a>MediaStreamTrack</a></code> object
<var>track</var> to represent the component.</p>
</li>
<li>
<p>Initialize <var>track's</var> <code><a href=
"getusermedia.html#dom-mediastreamtrack-kind">kind</a></code>
attribute to "<code>audio</code>" or "<code>video</code>"
depending on the media type of the incoming component.</p>
</li>
<li>
<p>Initialize <var>track's</var> <code><a href=
"getusermedia.html#dom-mediastreamtrack-id">id</a></code>
attribute to the component track id.</p>
</li>
<li>
<p>Initialize <var>track's</var> <code><a href=
"getusermedia.html#dom-mediastreamtrack-label">label</a></code>
attribute to "<code>remote audio</code>" or "<code>remote
video</code>" depending on the media type of the incoming
component.</p>
</li>
<li>
<p>Initialize <var>track's</var> <code><a href=
"getusermedia.html#dom-mediastreamtrack-readystate">readyState</a></code>
attribute to <code>muted</code>.</p>
</li>
</ol>
</li></p>
<p class="note">The creation of new incoming
<code>MediaStreamTrack</code>s may be triggered either by SDP
negotiation or by the receipt of media on a given flow.
<!-- [[OPEN ISSUE: How many <code>MediaStream</code>s are created
when you receive multiple conflicting pranswers?]] --></p>
</li>
<li>
Add <var>track</var> to the <code>MediaStream</code>(s) <var>streams</var>
referenced by the SDP negotiation, creating new ones if they do not
yet exist. If no <code>MediaStream</code> is specified,
a default <code>MediaStream</code> is created and used.</p>
</li>
<li>
<p>Queue a task to run the following substeps:</p>
<ol>
<li>
<p>If the <var>connection</var>'s <a href=
"#dom-peerconnection-signaling-state"><code>RTCPeerConnection</code>
signalingState</a> is <code>closed</code>, abort these
steps.</p>
</li>
<li>
<p>Create a new RTCRtpReceiver object <var>receiver</var>
for <var>track</track>, and add it
to <var>connection</var>'s <a href=
"#receivers-set">set of receivers</a>.</p>
</li>
<li>
<p><a href="#fire-a-track-event">Fire an event</a> named
<code title="event-track"><a href=
"#event-track">track</a></code> with
<var>receiver</var>, <var>track</var>, and <var>streams</var>
at the <var title="">connection</var> object.</p>
</li>
</ol>
</li>
</ol>
<p>When a user agent has negotiated media for a component that belongs
to a media track that is already represented by an existing
<code><a>MediaStreamTrack</a></code> object, the user agent MUST associate
the component with that <code><a>MediaStreamTrack</a></code> object.</p>
<p>When an <code><a>RTCPeerConnection</a></code> finds that a track
from the remote peer has been removed, the user agent MUST follow these
steps:</p>
<ol>
<li>
<p>Let <var>connection</var> be the
<code><a>RTCPeerConnection</a></code> associated with the track
being removed.</p>
</li>
<li>
<p>Let <var>track</var> be the <code><a>MediaStreamTrack</a></code>
object that represents the track being removed, if any. If
there isn't one, then abort these steps.</p>
</li>
<li>
<p>By definition, <var>track</var> is now ended.</p>
<p class="note">A <span title="concept-task">task</span> is thus
<span title="queue a task">queued</span> to update
<var>track</var> and fire an event.</p>
</li>
<li>
<p>Queue a task to run the following substeps:</p>
<ol>
<li>
<p>If the <var>connection</var>'s <a href=
"#dom-peerconnection-signaling-state"><code>RTCPeerConnection</code>
signalingState</a> is <code>closed</code>, abort these
steps.</p>
</li><!-- close() was probably called just before this
task ran -->
<li>
<p>Remove the RTCRtpReceiver associated with <var>track</var> from
<var>connection</var>'s <a href="#receivers-set">set of receivers</a>.</p>
</li>
<li>
<p>End the <var>track</var> object.</p>
</li>
</ol>
</li>
</ol>
<p>The task source for the <span title="concept-task">tasks</span>
listed in this section is the networking task source.</p>
<p>If something in the browser changes that causes the
<code><a>RTCPeerConnection</a></code> object to need to initiate a new
session description negotiation, a <code><a href=
"#event-negotiation">negotiationneeded</a></code> event is fired at the
<code><a>RTCPeerConnection</a></code> object.</p>
<p>In particular, if a <code><a>MediaStreamTrack</a></code> object
is added to (using the <a href="#dom-peerconnection-addtrack"><code>
addTrack()</code> </a> method) or removed from (using the
<a href="#dom-peerconnection-removetrack"><code>removeTrack()</code>
</a> method) an <code><a>RTCPeerConnection</a></code> object, the
<code><a>RTCPeerConnection</a></code> object MUST fire the
"negotiationneeded" event (except for the cases when the added track
is already present in the <a href="senders-set">set of RTCRtpSenders</a>,
or the removed track is not present in the <a href="senders-set">set
of RTCRtpSenders</a>).</p>
<p class="warning">To prevent network sniffing from allowing a fourth
party to establish a connection to a peer using the information sent
out-of-band to the other peer and thus spoofing the client, the
configuration information SHOULD always be transmitted using an
encrypted connection.</p>
</section>
<section>
<h3>Interface Definition</h3>
<dl class="idl" title="interface RTCPeerConnection : EventTarget ">
<dt>Constructor (RTCConfiguration configuration)</dt>
<dd>
See the <a href="#dom-peerconnection">RTCPeerConnection constructor
algorithm</a>.
</dd><!--
<dt>void getCapabilities ( RTCSessionDescriptionCallback
successCallback )</dt>
<dd>
<p> The getCapabilities method generates a blob of SDP that
contains a RFC 3264 offer that represets the most optimist view on
the capabilities of the media system. It does not reserver any
resources, ports, or other state but is meant to provide a way
to discover the types of capabilities of the browser including
which codecs may be supported. The SDP should have any ports set
to 0 (Open Issue: should this be 9?). Other values that would
allocate state should be set to static, unusable values. It
should include the SDP for media stream for each media type the
browser supports along with all the codecs that are supported.
It does not matter if any streams have been added to the
RTCPeerConnection object. </p>
<p> TODO - discuss privacy implications. </p>
</dd>
-->
<dt>void createOffer (RTCSessionDescriptionCallback successCallback,
RTCPeerConnectionErrorCallback failureCallback, optional
RTCOfferOptions options)</dt>
<dd>
<p>The createOffer method generates a blob of SDP that contains an
RFC 3264 offer with the supported configurations for the session,
including descriptions of the local <code>MediaStream</code>s
attached to this <code>RTCPeerConnection</code>, the codec/RTP/RTCP
options supported by this implementation, and any candidates that
have been gathered by the ICE Agent. The options parameter may be
supplied to provide additional control over the offer
generated.</p>
<p>As an offer, the generated SDP will contain the full set of
capabilities supported by the session (as opposed to an answer,
which will include only a specific negotiated subset to use); for
each SDP line, the generation of the SDP must follow the
appropriate process for generating an offer. In the event
createOffer is called after the session is established, createOffer
will generate an offer that is compatible with the current session,
incorporating any changes that have been made to the session since
the last complete offer-answer exchange, such as addition or
removal of streams. If no changes have been made, the offer will
include the capabilities of the current local description as well
as any additional capabilities that could be negotiated in an
updated offer.</p>
<p>Session descriptions generated by createOffer MUST be
immediately usable by setLocalDescription without causing an error
as long as setLocalDescription is called within the successCallback
function. If a system has limited resources (e.g. a finite number
of decoders), createOffer needs to return an offer that reflects
the current state of the system, so that setLocalDescription will
succeed when it attempts to acquire those resources. The session
descriptions MUST remain usable by setLocalDescription without
causing an error until at least end of the successCallback
function. Calling this method is needed to get the ICE user name
fragment and password.</p>
<p>If the <code>RTCPeerConnection</code> is configured to generate
Identity assertions, then the session description SHALL contain an
appropriate assertion.</p>
<p>If this <code>RTCPeerConnection</code> object is closed before
the SDP generation process completes, the USER agent MUST suppress
the result and not call any of the result callbacks.</p>
<p>If the SDP generation process completed successfully, the user
agent MUST queue a task to invoke <var>successCallback</var> with a
newly created <code><a>RTCSessionDescription</a></code> object,
representing the generated offer, as its argument.</p>
<p>If the SDP generation process failed for any reason, the user
agent MUST queue a task to invoke <var>failureCallback</var> with
an <code>DOMError</code> object of type TBD as its argument.</p>
<p>To Do: Discuss privacy aspects of this from a fingerprinting
point of view - it's probably around as bad as access to a canvas
:-)</p>
</dd>
<dt>void createAnswer (RTCSessionDescriptionCallback successCallback,
RTCPeerConnectionErrorCallback failureCallback)</dt>
<dd>
<p>The createAnswer method generates an [[!SDP]] answer with the
supported configuration for the session that is compatible with the
parameters in the remote configuration. Like createOffer, the
returned blob contains descriptions of the local MediaStreams
attached to this RTCPeerConnection, the codec/RTP/RTCP options
negotiated for this session, and any candidates that have been
gathered by the ICE Agent. The options parameter may be supplied to
provide additional control over the generated answer.</p>
<p>As an answer, the generated SDP will contain a specific
configuration that, along with the corresponding offer, specifies
how the media plane should be established. The generation of the
SDP must follow the appropriate process for generating an
answer.</p>
<p>Session descriptions generated by createAnswer must be
immediately usable by setLocalDescription without generating an
error if setLocalDescription is called from the successCallback
function. Like createOffer, the returned description should reflect
the current state of the system. The session descriptions MUST
remain usable by setLocalDescription without causing an error until
at least the end of the successCallback function. Calling this
method is needed to get the ICE user name fragment and
password.</p>
<p>An answer can be marked as provisional, as described in
[[!RTCWEB-JSEP]], by setting the <code><a href=
"#widl-RTCSessionDescription-type">type</a></code> to
<code>"pranswer"</code>.</p>
<p>If the <code>RTCPeerConnection</code> is configured to generate
Identity assertions, then the session description SHALL contain an
appropriate assertion.</p>
<p>If this <code>RTCPeerConnection</code> object is closed before
the SDP generation process completes, the USER agent MUST suppress
the result and not call any of the result callbacks.</p>
<p>If the SDP generation process completed successfully, the user
agent MUST queue a task to invoke <var>successCallback</var> with a
newly created <code><a>RTCSessionDescription</a></code> object,
representing the generated answer, as its argument.</p>
<p>If the SDP generation process failed for any reason, the user
agent MUST queue a task to invoke <var>failureCallback</var> with
an <code>DOMError</code> object of type TBD as its argument.</p>
</dd>
<dt>void setLocalDescription (RTCSessionDescription description,
VoidFunction successCallback, RTCPeerConnectionErrorCallback
failureCallback)</dt>
<dd>
<p>The <dfn id=
"dom-peerconnection-setlocaldescription"><code>setLocalDescription()</code></dfn>
method instructs the <code><a>RTCPeerConnection</a></code> to apply
the supplied <code><a>RTCSessionDescription</a></code> as the local
description.</p>
<p>This API changes the local media state. In order to successfully
handle scenarios where the application wants to offer to change
from one media format to a different, incompatible format, the
<code><a>RTCPeerConnection</a></code> must be able to
simultaneously support use of both the old and new local
descriptions (e.g. support codecs that exist in both descriptions)
until a final answer is received, at which point the
<code><a>RTCPeerConnection</a></code> can fully adopt the new local
description, or rollback to the old description if the remote side
denied the change.</p>
<p class="issue">ISSUE: how to indicate to rollback?</p>
<p>To Do: specify what parts of the SDP can be changed between the
createOffer and setLocalDescription</p>
<p>When the method is invoked, the user agent must follow the
<dfn id="set-description-model">processing model</dfn> described by
the following list:</p>
<ul>
<li>
<p>If this <code><a>RTCPeerConnection</a></code> object's
<a href="#dom-peerconnection-signaling-state">signaling
state</a> is <code>closed</code>, the user agent MUST throw an
<code>InvalidStateError</code> exception and abort this
operation.</p>
</li>
<li>
<p>If a local description contains a different set of ICE
credentials, then the ICE Agent MUST trigger an ICE restart.
When ICE restarts, the gathering state will be changed back to
"gathering", if it was not already gathering. If the <a href=
"#dom-peerconnection-ice-connection-state"><code>RTCPeerConnection</code>
ice connection state</a> was "completed", it will be changed
back to "connected".</p>
</li>
<li>
<p>If the process to apply the
<code><a>RTCSessionDescription</a></code> argument fails for
any reason, then user agent must queue a task runs the
following steps:</p>
<ol>
<li>
<p>Let <var>connection</var> be the
<code><a>RTCPeerConnection</a></code> object on with this
method was invoked.</p>
</li>
<li>
<p>If <var>connection</var>'s <a href=
"#dom-peerconnection-signaling-state">signaling state</a>
is <code>closed</code>, then abort these steps.</p>
</li>
<li>
<p>If the reason for the failure is:</p>
<ul>
<li>
<p>The content of the
<code><a>RTCSessionDescription</a></code> argument is
invalid or the <code><a href=
"#widl-RTCSessionDescription-type">type</a></code> is
wrong for the current <a href=
"#dom-peerconnection-signaling-state">signaling
state</a> of <var>connection</var>.</p>
<p>Let <var>errorType</var> be
<code>InvalidSessionDescriptionError</code>.</p>
</li>
<li>
<p>The <code><a>RTCSessionDescription</a></code> is a
valid description but cannot be applied at the media
layer.</p>
<p>TODO ISSUE - next few points are probably wrong.
Make sure to check this in setRemote too.</p>
<p>This can happen, e.g., if there are insufficient
resources to apply the SDP. The user agent MUST then
rollback as necessary if the new description was
partially applied when the failure occurred.</p>
<p>If rollback was not necessary or was completed
successfully, let <var>errorType</var> be
<code>IncompatibleSessionDescriptionError</code>. If
rollback was not possible, let <var>errorType</var> be
<code>InternalError</code> and set
<var>connection</var>'s <a href=
"#dom-peerconnection-signaling-state">signaling
state</a> to <code>closed</code>.</p>
</li>
</ul>
</li>
<li>
<p>Invoke the <var>failureCallback</var> with an
<code>DOMError</code> object, whose <code>name</code>
attribute is <var>errorType</var>, as its argument.</p>
</li>
</ol>
</li>
<li>
<p>If the <code><a>RTCSessionDescription</a></code> argument is
applied successfully, then user agent must queue a task runs
the following steps:</p>
<ol>
<li>
<p>Let <var>connection</var> be the
<code><a>RTCPeerConnection</a></code> object on with this
metod was invoked.</p>
</li>
<li>
<p>If <var>connection</var>'s <a href=
"#dom-peerconnection-signaling-state">signaling state</a>
is <code>closed</code>, then abort these steps.</p>
</li>
<li>
<p>Set <var>connection</var>'s description attribute (
<code><a>localDescription</a></code> or
<code><a>remoteDescription</a></code> depending on the
setting operation) to the
<code><a>RTCSessionDescription</a></code> argument.</p>
</li>
<li>
<p>If the local description was set,
<var>connection</var>'s <a href=
"#dom-peerconnection-ice-gathering-state">ice gathering
state</a> is <code>new</code>, and the local description
contains media, then set <var>connection</var>'s <a href=
"#dom-peerconnection-ice-gathering-state">ice gathering
state</a> to <code>gathering</code>.</p>
</li>
<li>
<p>If the local description was set with content that
caused an ICE restart, then set <var>connection</var>'s
<a href="#dom-peerconnection-ice-gathering-state">ice
gathering state</a> to <code>gathering</code>.</p>
</li>
<li>
<p>Set <var>connection</var>'s <a href=
"#dom-peerconnection-signaling-state">signalingState</a>
accordingly.</p>
</li>
<li>
<p>If <var>connection</var>'s <a href=