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<!DOCTYPE html>
<html lang="en">
<head>
<meta charset="utf-8">
<link href="webrtc.css" rel="stylesheet">
<title>
WebRTC 1.0: Real-time Communication Between Browsers
</title>
<script class="remove" src="respec-w3c-common.js" type="text/javascript">
// // keep this comment //
</script>
<script class="remove" src="webrtc.js" type="text/javascript">
// // keep
this comment //
</script>
</head>
<body>
<p class='copyright'>
Initial Author of this Specification was Ian Hickson, Google Inc., with
the following copyright statement:<br>
© Copyright 2004-2011 Apple Computer, Inc., Mozilla Foundation, and Opera
Software ASA. You are granted a license to use, reproduce and create
derivative works of this document. All subsequent changes since 26 July
2011 done by the W3C WebRTC Working Group are under the following
<a href='https://www.w3.org/Consortium/Legal/ipr-notice#Copyright'>Copyright</a>:<br>
© 2011-2018 <a href='https://www.w3.org/'><abbr title=
'World Wide Web Consortium'>W3C</abbr></a><sup>®</sup> (<a href=
'https://www.csail.mit.edu/'><abbr title=
'Massachusetts Institute of Technology'>MIT</abbr></a>, <a href=
'https://www.ercim.eu/'><abbr title=
'European Research Consortium for Informatics and Mathematics'>ERCIM</abbr></a>,
<a href='https://www.keio.ac.jp/'>Keio</a>, <a href=
'https://ev.buaa.edu.cn/'>Beihang</a>). <a href=
'https://www.w3.org/Consortium/Legal/copyright-documents'>Document
use</a> rules apply. For the entire publication on the W3C site the
<a href=
'https://www.w3.org/Consortium/Legal/ipr-notice#Legal_Disclaimer'>liability</a>
and <a href=
'https://www.w3.org/Consortium/Legal/ipr-notice#W3C_Trademarks'>trademark</a>
rules apply.
</p>
<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.
</p>
</section>
<section id="sotd">
<p>
The API is based on preliminary work done in the WHATWG.
</p>
<p>
The specification is feature complete and is expected to be stable with
no further substantive change. Since the <a href=
"https://www.w3.org/TR/2019/CR-webrtc-20191213/">previous Candidate
Recommendation</a>, the following substantive changes have been brought
to the specification:
</p>
<ul>
<li>the specification allows administrative prohibiting of ICE
candidates to help prevent WebRTC to be used for portscanning
</li>
<li>the support for setting multiple DTLS certificates has been
deprecated
</li>
<li>
<code>voiceActivityFlag</code> has been marked at risk for lack of
implementation
</li>
<li>the state machine when closing a <code>PeerConnection</code> has
been clarified
</li>
<li>Align with latest changes in <a href=
"https://w3c.github.io/webrtc-stats/">WebRTC Statistics</a>
</li>
</ul>
<p>
Its <a href=
"https://github.com/web-platform-tests/wpt/tree/master/webrtc">associated
test suite</a> will be used to build an <a href=
"https://w3c.github.io/webrtc-interop-reports/webrtc-pc-report.html">implementation
report</a> of the API.
</p>
<p>
To go into Proposed Recommendation status, the group expects to
demonstrate implementation of each feature in at least two deployed
browsers, and at least one implementation of each optional feature.
Mandatory feature with only one implementation may be marked as
optional in a revised Candidate Recommendation where applicable.
</p>
</section>
<section class="informative" id="intro">
<h2>
Introduction
</h2>
<p>
There are a number of facets to peer-to-peer communications and
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 tracks to remote peers and receiving
tracks 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=
"https://datatracker.ietf.org/wg/rtcweb/">IETF RTCWEB group</a> and an
API specification to get access to local media devices [[GETUSERMEDIA]]
developed by the WebRTC Working Group. 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 {{EventHandler}} interface, representing a callback used for event
handlers, is defined in [[!HTML]].
</p>
<p>
The concepts [= queue a task =] and [= networking task source =] are
defined in [[!HTML]].
</p>
<p>
The concept [= fire an event =] is defined in [[!DOM]].
</p>
<p>
The terms [= event =], [= event handlers =] and [= event handler event
types =] are defined in [[!HTML]].
</p>
<p>
{{Performance.timeOrigin}} and {{Performance.now()}} are defined in
[[!hr-time]].
</p>
<p>
The terms [= serializable objects =], [= serialization steps =], and [=
deserialization steps =] are defined in [[!HTML]].
</p>
<p>
The terms {{MediaStream}}, {{MediaStreamTrack}}, and
{{MediaStreamConstraints}} are defined in [[!GETUSERMEDIA]]. Note that
{{MediaStream}} is extended in <a href="#mediastream-network-use"></a>
in this document while {{MediaStreamTrack}} is extended in <a href=
"#mediastreamtrack-network-use"></a> in this document.
</p>
<p>
The term {{Blob}} is defined in [[!FILEAPI]].
</p>
<p>
The term <dfn>media description</dfn> is defined in [[!RFC4566]].
</p>
<p>
The term <dfn>media transport</dfn> is defined in [[!RFC7656]].
</p>
<p>
The term <dfn>generation</dfn> is defined in [[!TRICKLE-ICE]] Section
2.
</p>
<p>
The terms <dfn data-cite="!WEBRTC-STATS#dfn-stats-object">stats
object</dfn> and <dfn data-cite=
"!WEBRTC-STATS#dfn-monitored-object">monitored object</dfn> are defined
in [[!WEBRTC-STATS]].
</p>
<p>
When referring to exceptions, the terms [= exception/throw =] and [=
exception/created =] are defined in [[!WEBIDL]].
</p>
<p>
The callback {{VoidFunction}} is defined in [[!WEBIDL]].
</p>
<p>
The term "throw" is used as specified in [[!INFRA]]: it terminates the
current processing steps.
</p>
<p>
The terms <dfn data-lt="fulfill|fulfillment">fulfilled</dfn>,
<dfn data-lt="reject|rejection|rejecting">rejected</dfn>, <dfn data-lt=
"resolve|resolves">resolved</dfn>, <dfn>pending</dfn> and
<dfn>settled</dfn> used in the context of Promises are defined in
[[!ECMASCRIPT-6.0]].
</p>
<p>
The terms <dfn>bundle</dfn>, <dfn>bundle-only</dfn> and
<dfn>bundle-policy</dfn> are defined in [[!JSEP]].
</p>
<p>
The <dfn data-cite=
"WebCryptoAPI/#dfn-AlgorithmIdentifier">AlgorithmIdentifier</dfn> is
defined in [[!WebCryptoAPI]].
</p>
<p class="note">
The general principles for Javascript APIs apply, including the
principle of <a href=
"https://w3ctag.github.io/design-principles/#js-rtc">run-to-completion</a>
and no-data-races as defined in [[API-DESIGN-PRINCIPLES]]. That is,
while a task is running, external events do not influence what's
visible to the Javascript application. For example, the amount of data
buffered on a data channel will increase due to "send" calls while
Javascript is executing, and the decrease due to packets being sent
will be visible after a task checkpoint.<br>
It is the responsibility of the user agent to make sure the set of
values presented to the application is consistent - for instance that
getContributingSources() (which is synchronous) returns values for all
sources measured at the same time.
</p>
</section>
<section>
<h2>
Peer-to-peer connections
</h2>
<section>
<h3>
Introduction
</h3>
<p data-tests="RTCPeerConnection-remote-track-mute.https.html">
An {{RTCPeerConnection}} instance allows an application to establish
peer-to-peer communications with another {{RTCPeerConnection}}
instance in another browser, or to another endpoint implementing the
required protocols. Communications are coordinated by the exchange of
control messages (called a signaling protocol) over a signaling
channel which is provided by unspecified means, but generally by a
script in the page via the server, e.g. using <a data-cite="html">Web
Sockets</a> or <code>XMLHttpRequest</code> [[?xhr]].
</p>
</section>
<section>
<h3>
Configuration
</h3>
<section>
<h4>
<dfn>RTCConfiguration</dfn> Dictionary
</h4>
<p>
The {{RTCConfiguration}} defines a set of parameters to configure
how the peer-to-peer communication established via
{{RTCPeerConnection}} is established or re-established.
</p>
<div>
<pre class="idl">dictionary RTCConfiguration {
sequence<RTCIceServer> iceServers;
RTCIceTransportPolicy iceTransportPolicy;
RTCBundlePolicy bundlePolicy;
RTCRtcpMuxPolicy rtcpMuxPolicy;
sequence<RTCCertificate> certificates;
[EnforceRange] octet iceCandidatePoolSize = 0;
};</pre>
<section>
<h2>
Dictionary {{RTCConfiguration}} Members
</h2>
<dl data-link-for="RTCConfiguration" data-dfn-for=
"RTCConfiguration" class="dictionary-members">
<dt data-tests="RTCConfiguration-iceServers.html">
<dfn data-idl="">iceServers</dfn> of type <span class=
"idlMemberType">sequence<{{RTCIceServer}}></span>
</dt>
<dd>
<p>
An array of objects describing servers available to be used
by ICE, such as STUN and TURN servers.
</p>
</dd>
<dt data-tests="RTCConfiguration-iceTransportPolicy.html">
<dfn data-idl="">iceTransportPolicy</dfn> of type
<span class="idlMemberType">{{RTCIceTransportPolicy}}</span>.
</dt>
<dd>
<p>
Indicates which candidates the [= ICE Agent =] is allowed
to use.
</p>
</dd>
<dt data-tests=
"RTCConfiguration-bundlePolicy.html,RTCPeerConnection-iceConnectionState.https.html">
<dfn data-idl="">bundlePolicy</dfn> of type <span class=
"idlMemberType">{{RTCBundlePolicy}}</span>.
</dt>
<dd>
<p>
Indicates which <a data-lt="RTCBundlePolicy">media-bundling
policy</a> to use when gathering ICE candidates.
</p>
</dd>
<dt data-tests="RTCConfiguration-rtcpMuxPolicy.html">
<dfn data-idl="">rtcpMuxPolicy</dfn> of type <span class=
"idlMemberType">{{RTCRtcpMuxPolicy}}</span>.
</dt>
<dd>
<p>
Indicates which <a data-lt="RTCRtcpMuxPolicy">rtcp-mux
policy</a> to use when gathering ICE candidates.
</p>
</dd>
<dt data-tests=
"RTCPeerConnection-constructor.html,RTCCertificate.html">
<dfn data-idl="">certificates</dfn> of type <span class=
"idlMemberType">sequence<{{RTCCertificate}}></span>
</dt>
<dd>
<p>
A set of certificates that the {{RTCPeerConnection}} uses
to authenticate.
</p>
<p>
Valid values for this parameter are created through calls
to the {{RTCPeerConnection/generateCertificate()}}
function.
</p>
<p>
Although any given DTLS connection will use only one
certificate, this attribute allows the caller to provide
multiple certificates that support different algorithms.
The final certificate will be selected based on the DTLS
handshake, which establishes which certificates are
allowed. The {{RTCPeerConnection}} implementation selects
which of the certificates is used for a given connection;
how certificates are selected is outside the scope of this
specification.
</p>
<p class="note">
Existing implementations only utilize the first certificate
provided; the others are ignored.
</p>
<p>
If this value is absent, then a default set of certificates
is generated for each {{RTCPeerConnection}} instance.
</p>
<p>
This option allows applications to establish key
continuity. An {{RTCCertificate}} can be persisted in
[[?INDEXEDDB]] and reused. Persistence and reuse also
avoids the cost of key generation.
</p>
<p>
The value for this configuration option cannot change after
its value is initially selected.
</p>
</dd>
<dt data-tests=
"RTCPeerConnection-constructor.html,RTCConfiguration-iceCandidatePoolSize.html">
<dfn data-idl="">iceCandidatePoolSize</dfn> of type
<span class="idlMemberType">octet</span>, defaulting to
<code>0</code>
</dt>
<dd>
<p>
Size of the prefetched ICE pool as defined in
<span data-jsep=
"ice-candidate-pool constructor">[[!JSEP]]</span>.
</p>
</dd>
</dl>
</section>
</div>
</section>
<section>
<h4>
<dfn>RTCIceCredentialType</dfn> Enum
</h4>
<div>
<pre class="idl">enum RTCIceCredentialType {
"password"
};</pre>
<table data-link-for="RTCIceCredentialType" data-dfn-for=
"RTCIceCredentialType" class="simple">
<thead>
<tr>
<th colspan="2">
Enumeration description
</th>
</tr>
</thead>
<tbody>
<tr>
<td data-tests="RTCConfiguration-iceServers.html">
<dfn data-idl="">password</dfn>
</td>
<td>
The credential is a long-term authentication username and
password, as described in [[!RFC5389]], Section 10.2.
</td>
</tr>
</tbody>
</table>
</div>
</section>
<section>
<h4>
<dfn>RTCIceServer</dfn> Dictionary
</h4>
<p>
The {{RTCIceServer}} dictionary is used to describe the STUN and
TURN servers that can be used by the [= ICE Agent =] to establish a
connection with a peer.
</p>
<div>
<pre class="idl">dictionary RTCIceServer {
required (DOMString or sequence<DOMString>) urls;
DOMString username;
DOMString credential;
RTCIceCredentialType credentialType = "password";
};</pre>
<section>
<h2>
Dictionary {{RTCIceServer}} Members
</h2>
<dl data-link-for="RTCIceServer" data-dfn-for="RTCIceServer"
class="dictionary-members">
<dt data-tests="RTCConfiguration-iceServers.html">
<dfn data-idl="">urls</dfn> of type <span class=
"idlMemberType">(DOMString or
sequence<DOMString>)</span>, required
</dt>
<dd>
<p>
STUN or TURN URI(s) as defined in [[!RFC7064]] and
[[!RFC7065]] or other URI types.
</p>
</dd>
<dt data-tests="RTCConfiguration-iceServers.html">
<dfn data-idl="">username</dfn> of type <span class=
"idlMemberType">DOMString</span>
</dt>
<dd>
<p>
If this {{RTCIceServer}} object represents a TURN server,
and {{credentialType}} is
{{RTCIceCredentialType/"password"}}, then this attribute
specifies the username to use with that TURN server.
</p>
</dd>
<dt data-tests="RTCConfiguration-iceServers.html">
<dfn data-idl="">credential</dfn> of type {{DOMString}}
</dt>
<dd>
<p>
If this {{RTCIceServer}} object represents a TURN server,
then this attribute specifies the credential to use with
that TURN server.
</p>
<p>
If {{credentialType}} is
{{RTCIceCredentialType/"password"}}, {{credential}}
represents a long-term authentication password, as
described in [[!RFC5389]], Section 10.2.
</p>
<p class="note">
To support additional values of {{credentialType}},
{{credential}} may evolve in future as a union.
</p>
</dd>
<dt data-tests="RTCConfiguration-iceServers.html">
<dfn data-idl="">credentialType</dfn> of type <span class=
"idlMemberType">{{RTCIceCredentialType}}</span>, defaulting
to {{RTCIceCredentialType/"password"}}
</dt>
<dd>
<p>
If this {{RTCIceServer}} object represents a TURN server,
then this attribute specifies how <var>credential</var>
should be used when that TURN server requests
authorization.
</p>
</dd>
</dl>
</section>
</div>
<p>
An example array of {{RTCIceServer}} objects is:
</p>
<pre class="example highlight">
[
{urls: 'stun:stun1.example.net'},
{urls: ['turns:turn.example.org', 'turn:turn.example.net'],
username: 'user',
credential: 'myPassword',
credentialType: 'password'},
];
</pre>
</section>
<section data-tests="RTCConfiguration-iceTransportPolicy.html">
<h4>
<dfn>RTCIceTransportPolicy</dfn> Enum
</h4>
<p>
As described in <span data-jsep="constructor">[[!JSEP]]</span>, if
the {{RTCConfiguration/iceTransportPolicy}} member of the
{{RTCConfiguration}} is specified, it defines the <span data-jsep=
"ice-candidate-policy">ICE candidate policy [[!JSEP]]</span> the
browser uses to surface the permitted candidates to the
application; only these candidates will be used for connectivity
checks.
</p>
<div>
<pre class="idl">enum RTCIceTransportPolicy {
"relay",
"all"
};</pre>
<table data-link-for="RTCIceTransportPolicy" data-dfn-for=
"RTCIceTransportPolicy" class="simple">
<thead>
<tr>
<th colspan="2">
Enumeration description (non-normative)
</th>
</tr>
</thead>
<tbody>
<tr>
<td>
<dfn data-idl="">relay</dfn>
</td>
<td>
<p>
The [= ICE Agent =] uses only media relay candidates such
as candidates passing through a TURN server.
</p>
<div class="note">
This can be used to prevent the remote endpoint from
learning the user's IP addresses, which may be desired in
certain use cases. For example, in a "call"-based
application, the application may want to prevent an
unknown caller from learning the callee's IP addresses
until the callee has consented in some way.
</div>
</td>
</tr>
<tr>
<td>
<dfn data-idl="">all</dfn>
</td>
<td>
<p>
The [= ICE Agent =] can use any type of candidate when
this value is specified.
</p>
<div class="note">
The implementation can still use its own candidate
filtering policy in order to limit the IP addresses
exposed to the application, as noted in the description
of {{RTCIceCandidate}}.{{RTCIceCandidate/address}}.
</div>
</td>
</tr>
</tbody>
</table>
</div>
</section>
<section data-tests="RTCConfiguration-bundlePolicy.html">
<h4>
<dfn>RTCBundlePolicy</dfn> Enum
</h4>
<p>
As described in <span data-jsep="constructor">[[!JSEP]]</span>,
bundle policy affects which media tracks are negotiated if the
remote endpoint is not bundle-aware, and what ICE candidates are
gathered. If the remote endpoint is bundle-aware, all media tracks
and data channels are bundled onto the same transport.
</p>
<div>
<pre id="target-bundle-policy" class="idl">enum RTCBundlePolicy {
"balanced",
"max-compat",
"max-bundle"
};</pre>
<table data-link-for="RTCBundlePolicy" data-dfn-for=
"RTCBundlePolicy" class="simple">
<thead>
<tr>
<th colspan="2">
Enumeration description (non-normative)
</th>
</tr>
</thead>
<tbody>
<tr>
<td data-tests=
"RTCRtpSender-transport.https.html,RTCPeerConnection-iceConnectionState.https.html">
<dfn data-idl="">balanced</dfn>
</td>
<td>
Gather ICE candidates for each media type in use (audio,
video, and data). If the remote endpoint is not
bundle-aware, negotiate only one audio and video track on
separate transports.
</td>
</tr>
<tr>
<td data-tests=
"RTCRtpSender-transport.https.html,RTCPeerConnection-iceConnectionState.https.html">
<dfn data-idl="">max-compat</dfn>
</td>
<td>
Gather ICE candidates for each track. If the remote
endpoint is not bundle-aware, negotiate all media tracks on
separate transports.
</td>
</tr>
<tr>
<td data-tests=
"RTCRtpSender-transport.https.html,RTCPeerConnection-iceConnectionState.https.html">
<dfn data-idl="">max-bundle</dfn>
</td>
<td>
Gather ICE candidates for only one track. If the remote
endpoint is not bundle-aware, negotiate only one media
track.
</td>
</tr>
</tbody>
</table>
</div>
</section>
<section data-tests="RTCConfiguration-rtcpMuxPolicy.html">
<h4>
<dfn>RTCRtcpMuxPolicy</dfn> Enum
</h4>
<p>
As described in <span data-jsep="constructor">[[!JSEP]]</span>, the
{{RTCRtcpMuxPolicy}} affects what ICE candidates are gathered to
support non-multiplexed RTCP. The only value defined in this spec
is {{RTCRtcpMuxPolicy/"require"}}.
</p>
<div>
<pre id="target-rtcp-mux-policy" class="idl">
enum RTCRtcpMuxPolicy {
"require"
};</pre>
<table data-link-for="RTCRtcpMuxPolicy" data-dfn-for=
"RTCRtcpMuxPolicy" class="simple">
<thead>
<tr>
<th colspan="2">
Enumeration description (non-normative)
</th>
</tr>
</thead>
<tbody>
<tr>
<td>
<dfn data-idl="">require</dfn>
</td>
<td>
Gather ICE candidates only for RTP and multiplex RTCP on
the RTP candidates. If the remote endpoint is not capable
of rtcp-mux, session negotiation will fail.
</td>
</tr>
</tbody>
</table>
</div>
</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>
<div>
<pre class="idl">dictionary RTCOfferAnswerOptions {};</pre>
<section>
<h2>
Dictionary <dfn>RTCOfferAnswerOptions</dfn> Members
</h2>
<dl data-link-for="RTCOfferAnswerOptions" data-dfn-for=
"RTCOfferAnswerOptions" class="dictionary-members"></dl>
</section>
</div>
<div>
<pre class="idl">
dictionary RTCOfferOptions : RTCOfferAnswerOptions {
boolean iceRestart = false;
};</pre>
<section>
<h2>
Dictionary <dfn>RTCOfferOptions</dfn> Members
</h2>
<dl data-link-for="RTCOfferOptions" data-dfn-for=
"RTCOfferOptions" class="dictionary-members">
<dt data-tests="RTCPeerConnection-restartIce.https.html">
<dfn data-idl="">iceRestart</dfn> of type <span class=
"idlMemberType">boolean</span>, defaulting to
<code>false</code>
</dt>
<dd>
<p>
When the value of this dictionary member is
<code>true</code>, or the relevant {{RTCPeerConnection}}
object's <a>[[\LocalIceCredentialsToReplace]]</a> slot is
not empty, then the generated description will have ICE
credentials that are different from the current credentials
(as visible in the
{{RTCPeerConnection/currentLocalDescription}} attribute's
SDP). Applying the generated description will restart ICE,
as described in section 9.1.1.1 of [[!ICE]].
</p>
<p>
When the value of this dictionary member is
<code>false</code>, and the relevant {{RTCPeerConnection}}
object's <a>[[\LocalIceCredentialsToReplace]]</a> slot is
empty, and the
{{RTCPeerConnection/currentLocalDescription}} attribute has
valid ICE credentials, then the generated description will
have the same ICE credentials as the current value from the
{{RTCPeerConnection/currentLocalDescription}} attribute.
</p>
<p class="note">
Performing an ICE restart is recommended when
{{RTCPeerConnection/iceConnectionState}} transitions to
{{RTCIceConnectionState/"failed"}}. An application may
additionally choose to listen for the
{{RTCPeerConnection/iceConnectionState}} transition to
{{RTCIceConnectionState/"disconnected"}} and then use other
sources of information (such as using
{{RTCPeerConnection/getStats}} to measure if the number of
bytes sent or received over the next couple of seconds
increases) to determine whether an ICE restart is
advisable.
</p>
</dd>
</dl>
</section>
</div>
<div>
<p>
The <dfn>RTCAnswerOptions</dfn> dictionary describe options
specific to session description of type {{RTCSdpType/"answer"}}
(none in this version of the specification).
</p>
<pre class="idl">
dictionary RTCAnswerOptions : RTCOfferAnswerOptions {};</pre>
</div>
</section>
</section>
<section>
<h3>
State Definitions
</h3>
<section>
<h4>
<dfn>RTCSignalingState</dfn> Enum
</h4>
<div>
<pre class="idl">enum RTCSignalingState {
"stable",
"have-local-offer",
"have-remote-offer",
"have-local-pranswer",
"have-remote-pranswer",
"closed"
};</pre>
<table data-link-for="RTCSignalingState" data-dfn-for=
"RTCSignalingState" class="simple">
<thead>
<tr>
<th colspan="2">
Enumeration description
</th>
</tr>
</thead>
<tbody>
<tr>
<td data-tests=
"RTCPeerConnection-onsignalingstatechanged.https.html,RTCPeerConnection-setRemoteDescription.html">
<dfn data-idl="">stable</dfn>
</td>
<td>
There is no offer/answer exchange in progress. This is also
the initial state, in which case the local and remote
descriptions are empty.
</td>
</tr>
<tr>
<td data-tests=
"RTCPeerConnection-createOffer.html, RTCPeerConnection-setLocalDescription-offer.html,RTCPeerConnection-setLocalDescription.html,RTCPeerConnection-setRemoteDescription.html">
<dfn data-idl="">have-local-offer</dfn>
</td>
<td>
A local description, of type {{RTCSdpType/"offer"}}, has
been successfully applied.
</td>
</tr>
<tr>
<td data-tests=
"RTCPeerConnection-setLocalDescription-answer.html,RTCPeerConnection-setRemoteDescription-offer.html,RTCPeerConnection-setRemoteDescription-rollback.html,RTCPeerConnection-setRemoteDescription.html">
<dfn data-idl="">have-remote-offer</dfn>
</td>
<td>
A remote description, of type {{RTCSdpType/"offer"}}, has
been successfully applied.
</td>
</tr>
<tr>
<td data-tests=
"RTCPeerConnection-setLocalDescription-pranswer.html">
<dfn data-idl="">have-local-pranswer</dfn>
</td>
<td>
A remote description of type {{RTCSdpType/"offer"}} has
been successfully applied and a local description of type
{{RTCSdpType/"pranswer"}} has been successfully applied.
</td>
</tr>
<tr>
<td data-tests=
"RTCPeerConnection-setRemoteDescription-pranswer.html">
<dfn data-idl="">have-remote-pranswer</dfn>
</td>
<td>
A local description of type {{RTCSdpType/"offer"}} has been
successfully applied and a remote description of type
{{RTCSdpType/"pranswer"}} has been successfully applied.
</td>
</tr>
<tr>
<td data-tests=
"RTCPeerConnection-onsignalingstatechanged.https.html">
<dfn data-idl="">closed</dfn>
</td>
<td>
The {{RTCPeerConnection}} has been closed; its
<a>[[\IsClosed]]</a> slot is <code>true</code>.
</td>
</tr>
</tbody>
</table>
</div>
<figure>
<img alt="signaling state transition diagram" src=
"images/peerstates.svg" width="600">
<figcaption>
Non-normative signaling state transitions diagram. Method calls
abbreviated.
</figcaption>
</figure>
<p>
An example set of transitions might be:
</p>
<dl>
<dt>
Caller transition:
</dt>
<dd>
<ul>
<li>new RTCPeerConnection(): {{RTCSignalingState/"stable"}}
</li>
<li>setLocalDescription(offer):
{{RTCSignalingState/"have-local-offer"}}
</li>
<li>setRemoteDescription(pranswer):
{{RTCSignalingState/"have-remote-pranswer"}}
</li>
<li>setRemoteDescription(answer):
{{RTCSignalingState/"stable"}}
</li>
</ul>
</dd>
<dt>
Callee transition:
</dt>
<dd>
<ul>
<li>new RTCPeerConnection(): {{RTCSignalingState/"stable"}}
</li>
<li>setRemoteDescription(offer):
{{RTCSignalingState/"have-remote-offer"}}
</li>
<li>setLocalDescription(pranswer):
{{RTCSignalingState/"have-local-pranswer"}}
</li>
<li>setLocalDescription(answer): {{RTCSignalingState/"stable"}}
</li>
</ul>
</dd>
</dl>
</section>
<section>
<h4>
<dfn>RTCIceGatheringState</dfn> Enum
</h4>
<div>
<pre class="idl">enum RTCIceGatheringState {
"new",
"gathering",
"complete"
};</pre>
<table data-link-for="RTCIceGatheringState" data-dfn-for=
"RTCIceGatheringState" class="simple">
<thead>
<tr>
<th colspan="2">
Enumeration description
</th>
</tr>
</thead>
<tbody>
<tr>
<td data-tests="RTCPeerConnection-iceGatheringState.html">
<dfn data-idl="">new</dfn>
</td>
<td>
Any of the {{RTCIceTransport}}s are in the
{{RTCIceGathererState/"new"}} gathering state and none of
the transports are in the
{{RTCIceGathererState/"gathering"}} state, or there are no
transports.
</td>
</tr>
<tr>
<td data-tests="RTCPeerConnection-iceGatheringState.html">
<dfn data-idl="">gathering</dfn>
</td>
<td>
Any of the {{RTCIceTransport}}s are in the
{{RTCIceGathererState/"gathering"}} state.
</td>
</tr>
<tr>
<td data-tests=
"RTCPeerConnection-iceGatheringState.html,protocol/candidate-exchange.https.html">
<dfn data-idl="">complete</dfn>
</td>
<td>
At least one {{RTCIceTransport}} exists, and all
{{RTCIceTransport}}s are in the
{{RTCIceGathererState/"complete"}} gathering state.
</td>
</tr>
</tbody>
</table>
</div>
</section>
<section>
<h4>
<dfn>RTCPeerConnectionState</dfn> Enum
</h4>
<div>
<pre class="idl">enum RTCPeerConnectionState {
"closed",
"failed",
"disconnected",
"new",
"connecting",
"connected"
};</pre>
<table data-link-for="RTCPeerConnectionState" data-dfn-for=
"RTCPeerConnectionState" class="simple">
<thead>
<tr>
<th colspan="2">
Enumeration description
</th>
</tr>
</thead>
<tbody>
<tr>
<td data-tests=
"RTCPeerConnection-connectionState.https.html">
<dfn data-idl="">closed</dfn>
</td>
<td>
The {{RTCPeerConnection}} object's <a>[[\IsClosed]]</a>
slot is <code>true</code>.
</td>