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<!DOCTYPE html>
<html lang="en" dir="ltr" xmlns="http://www.w3.org/1999/xhtml">
<head>
<title>Microsoft Object RTC (ORTC) Implementation</title>
<meta charset='utf-8'>
<script src="respec-w3c-common.js" async class="remove"></script>
<script class="remove">
var respecConfig = {
"specStatus": "CG-DRAFT",
"shortName": "ms-ortc",
"publishDate": "2017-10-13",
"editors": [
{ "name": "Microsoft Corporation",
"company": "Microsoft",
"companyURL": "https://www.microsoft.com/" }
],
"authors": [
],
"wg": "Microsoft Corporation",
"wgPatentURI": "",
"localBiblio": {
"IANA-RTP-2": {
title: "RTP Payload Format media types"
, href: "https://www.iana.org/assignments/rtp-parameters/rtp-parameters.xhtml#rtp-parameters-2"
, publisher: "IANA"
},
"IANA-RTP-10": {
title: "RTP Compact Header Extensions"
, href: "https://www.iana.org/assignments/rtp-parameters/rtp-parameters.xhtml#rtp-parameters-10"
, publisher: "IANA"
},
"IANA-SDP-14": {
title: "'rtcp-fb' Attribute Values"
, href: "https://www.iana.org/assignments/sdp-parameters/sdp-parameters.xhtml#sdp-parameters-14"
, publisher: "IANA"
},
"IANA-SDP-15": {
title: "'ack' and 'nack' Attribute Values"
, href: "https://www.iana.org/assignments/sdp-parameters/sdp-parameters.xhtml#sdp-parameters-15"
, publisher: "IANA"
},
"IANA-SDP-19": {
title: "Codec Control Messages"
, href: "https://www.iana.org/assignments/sdp-parameters/sdp-parameters.xhtml#sdp-parameters-19"
, publisher: "IANA"
},
"MUX-FIXES": {
title: "Multiplexing Scheme Updates for Secure Real-time Transport Protocol (SRTP) Extension for Datagram Transport Layer Security (DTLS)"
, href: "https://tools.ietf.org/html/draft-ietf-avtcore-rfc5764-mux-fixes"
, authors: [
"M. Petit-Huguenin"
, "G. Salgueiro"
]
, status: "24 March 2015. Internet draft (work in progress)"
, publisher: "IETF"
},
"OPUS-RTP": {
title: "RTP Payload Format for Opus Speech and Audio Codec"
, href: "https://tools.ietf.org/html/draft-ietf-payload-rtp-opus"
, authors: [
"J. Spittka"
, "K. Vos"
, "JM. Valin"
]
, status: "14 April 2015. Internet Draft (work in progress)"
, publisher: "IETF"
},
"RFC3264": {
title: "An Offer/Answer Model with the Session Description Protocol"
, href: "https://tools.ietf.org/html/rfc3264"
, authors: [
"J. Rosenberg"
, "H. Schulzrinne"
]
, status: "July 2002. RFC"
, publisher: "IETF"
},
"RFC4568": {
title: "SDP Security Descriptions for Media Streams"
, href: "https://tools.ietf.org/html/rfc4568"
, authors: [
"F. Andreasen"
, "M. Baugher"
, "D. Wing"
]
, status: "July 2006. RFC"
, publisher: "IETF"
},
"RFC4585": {
title: "Extended RTP Profile for RTCP-Based Feedback (RTP/AVPF)"
, href: "https://tools.ietf.org/html/rfc4585"
, authors: [
"J. Ott"
, "S. Wenger"
, "N. Sato"
, "C. Burmeister"
, "J. Rey"
]
, status: "July 2006. RFC"
, publisher: "IETF"
},
"RFC4733": {
title: "RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals"
, href: "https://tools.ietf.org/html/rfc4733"
, authors: [
"H. Schulzrinne"
, "T. Taylor"
]
, status: "December 2006. RFC"
, publisher: "IETF"
},
"RFC4961": {
title: "Symmetric RTP/RTC Control Protocol (RTCP)"
, href: "https://tools.ietf.org/html/rfc4961"
, authors: [
"D. Wing"
]
, status: "July 2007. RFC"
, publisher: "IETF"
},
"RFC5104": {
title: "Codec Control Messages in the RTP Audio-Visual Profile with Feedback (AVPF)"
, href: "https://tools.ietf.org/html/rfc5104"
, authors: [
"S. Wenger"
, "U. Chandra"
, "M. Westerlund"
, "B. Burman"
]
, status: "February 2008. RFC"
, publisher: "IETF"
},
"RFC5124": {
title: "Extended Secure RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/SAVPF)"
, href: "https://tools.ietf.org/html/rfc5124"
, authors: [
"J. Ott"
, "E. Carrara"
]
, status: "February 2008. RFC"
, publisher: "IETF"
},
"RFC5450": {
title: "Transmission Time Offsets in RTP Streams"
, href: "https://tools.ietf.org/html/rfc5450"
, authors: [
"D. Singer"
, "H. Desineni"
]
, status: "March 2009. RFC"
, publisher: "IETF"
},
"RFC5506": {
title: "Support for Reduced-Size Real-Time Transport Control Protocol (RTCP): Opportunities and Consequences"
, href: "https://tools.ietf.org/html/rfc5506"
, authors: [
"I. Johansson"
, "M. Westerlund"
]
, status: "April 2009. RFC"
, publisher: "IETF"
},
"RFC5583": {
title: "Signaling Media Decoding Dependency in the SDP"
, href: "https://tools.ietf.org/html/rfc5583"
, authors: [
"T. Schierl"
, "S. Wenger"
]
, status: "July 2009. RFC"
, publisher: "IETF"
},
"RFC5761": {
title: "Multiplexing RTP Data and Control Packets on a Single Port"
, href: "https://tools.ietf.org/html/rfc5761"
, authors: [
"C. Perkins"
, "M. Westerlund"
]
, status: "April 2010. RFC"
, publisher: "IETF"
},
"RFC5764": {
title: "Datagram Transport Layer Security (DTLS) Extension to Establish Keys for the Secure Real-time Transport Protocol (SRTP)"
, href: "https://tools.ietf.org/html/rfc5764"
, authors: [
"D. McGrew"
, "E. Rescorla"
]
, status: "May 2010. RFC"
, publisher: "IETF"
},
"RFC6184": {
title: "RTP Payload Format for H.264 Video"
, href: "https://tools.ietf.org/html/rfc6184"
, authors: [
"Y.-K.. Wang"
, "R. Even"
, "T. Kristensen"
, "R. Jesup"
]
, status: "May 2011. RFC"
, publisher: "IETF"
},
"RFC6190": {
title: "RTP Payload Format for Scalable Video Coding"
, href: "https://tools.ietf.org/html/rfc6190"
, authors: [
"S. Wenger"
, "Y.-K. Wang"
, "T. Schierl"
, "A. Eleftheriadis"
]
, status: "May 2011. RFC"
, publisher: "IETF"
},
"RFC6455": {
title: "The WebSocket Protocol"
, href: "https://tools.ietf.org/html/rfc6455"
, authors: [
"I. Fette"
, "A. Melnikov"
]
, status: "December 2011. RFC"
, publisher: "IETF"
},
"RFC6544": {
title: "TCP Candidates with Interactive Connectivity Establishment (ICE)"
, href: "https://tools.ietf.org/html/rfc6544"
, authors: [
"J. Rosenberg"
, "A. Keranen"
, "B. B. Lowekamp"
, "A. B. Roach"
]
, status: "March 2012. RFC"
, publisher: "IETF"
},
"RFC6716": {
title: "Definition of the Opus Audio Codec"
, href: "https://tools.ietf.org/html/rfc6716"
, authors: [
"JM. Valin",
"K. Vos",
"T. Terriberry"
]
, status: "September 2012. RFC"
, publisher: "IETF"
},
"RFC7022": {
title: "Guidelines for Choosing RTP Control Protocol (RTCP) Canonical Names (CNAMEs)"
, href: "https://tools.ietf.org/html/rfc7022"
, authors: [
"A. Begen",
"C. Perkins",
"D. Wing",
"E. Rescorla"
]
, status: "September 2013. RFC"
, publisher: "IETF"
},
"RFC7064": {
title: "URI Scheme for Session Traversal Utilities for NAT (STUN) Protocol"
, href: "https://tools.ietf.org/html/rfc7064"
, authors: [
"S. Nandakumar",
"G. Salgueiro",
"P. Jones",
"M. Petit-Huguenin"
]
, status: "November 2013. RFC"
, publisher: "IETF"
},
"RFC7065": {
title: "Traversal Using Relays around NAT (TURN) Uniform Resource Identifiers"
, href: "https://tools.ietf.org/html/rfc7065"
, authors: [
"M. Petit-Huguenin",
"S. Nandakumar",
"G. Salgueiro",
"P. Jones"
]
, status: "November 2013. RFC"
, publisher: "IETF"
},
"BUNDLE": {
"title": "Negotiating Media Multiplexing Using the Session Description Protocol (SDP)",
"href": "https://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation",
"authors": [
"C. Holmberg",
"H. Alvestrand",
"C. Jennings"
],
"status": "12 April 2017. Internet Draft (work in progress)",
"publisher": "IETF"
},
"CONSENT": {
title: "STUN Usage for Consent Freshness"
, href: "https://tools.ietf.org/html/draft-ietf-rtcweb-stun-consent-freshness"
, authors: [
"M. Perumal",
"D. Wing",
"T. Reddy",
"M. Thomson"
]
, status: "8 June 2015. Internet Draft (work in progress)"
, publisher: "IETF"
},
"GROUPING": {
title: "RTP Grouping Taxonomy"
, href: "https://tools.ietf.org/html/draft-ietf-avtext-rtp-grouping-taxonomy"
, authors: [
"J. Lennox",
"K. Gross",
"S. Nandakumar",
"G. Salgueiro",
"B. Burman"
]
, status: "05 March 2015. Internet Draft (work in progress)"
, publisher: "IETF"
},
"MS-DTMF": {
title: "RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals Extensions"
, href: "https://msdn.microsoft.com/en-us/library/office/cc431502(v=office.12).aspx"
, authors: [
"Microsoft"
]
, status: "30 March 2015."
, publisher: "Microsoft"
},
"MS-H264PF": {
title: "RTP Payload Format for H.264 Video Streams Extensions"
, href: "https://msdn.microsoft.com/en-us/library/hh659565(v=office.12).aspx"
, authors: [
"Microsoft"
]
, status: "30 March 2015."
, publisher: "Microsoft"
},
"MS-ICE": {
title: "Interactive Connectivity Establishment (ICE) Extensions"
, href: "https://msdn.microsoft.com/en-us/library/office/Cc431495(v=office.12).aspx"
, authors: [
"Microsoft"
]
, status: "30 March 2015."
, publisher: "Microsoft"
},
"MS-ICE2": {
title: "Interactive Connectivity Establishment (ICE) Extensions 2.0"
, href: "https://msdn.microsoft.com/en-us/library/office/cc431504(v=office.12).aspx"
, authors: [
"Microsoft"
]
, status: "30 March 2015."
, publisher: "Microsoft"
},
"MS-QoE": {
title: "Quality of Experience Monitoring Server Protocol"
, href: "https://msdn.microsoft.com/en-us/library/office/cc431512(v=office.12).aspx"
, authors: [
"Microsoft"
]
, status: "14 September 2016."
, publisher: "Microsoft"
},
"MS-ICE": {
title: "[MS-ICE]: Interactive Connectivity Establish (ICE) Extensions"
, href: "https://msdn.microsoft.com/en-us/library/dd922095%28v=office.12%29.aspx"
, authors: [
"Microsoft"
]
, status: "30 October 2014."
, publisher: "Microsoft"
},
"MS-RTP": {
title: "Real-time Transport Protocol (RTP) Extensions"
, href: "https://msdn.microsoft.com/en-us/library/office/cc431492(v=office.12).aspx"
, authors: [
"Microsoft"
]
, status: "30 March 2015."
, publisher: "Microsoft"
},
"MS-RTPRADEX": {
title: "RTP Payload for Redundant Audio Data Extensions"
, href: "https://msdn.microsoft.com/en-us/library/office/cc431513(v=office.12).aspx"
, authors: [
"Microsoft"
]
, status: "30 March 2015."
, publisher: "Microsoft"
},
"MS-SDPEXT": {
title: "Session Description Protocol (SDP) Version 2.0 Extensions"
, href: "https://msdn.microsoft.com/en-us/library/cc431514(v=office.12).aspx"
, authors: [
"Microsoft"
]
, status: "30 March 2015."
, publisher: "Microsoft"
},
"MS-TURN": {
title: "Traversal Using Relay NAT (TURN) Extensions"
, href: "https://msdn.microsoft.com/en-us/library/cc431507(v=office.12).aspx"
, authors: [
"Microsoft"
]
, status: "30 March 2015."
, publisher: "Microsoft"
},
"RTCWEB-SECURITY": {
title: "Security Considerations for WebRTC"
, href: "https://tools.ietf.org/html/draft-ietf-rtcweb-security"
, authors: [
"E. Rescorla"
]
, status: "26 February 2015. Internet Draft (work in progress)"
, publisher: "IETF"
},
"RTCWEB-SECURITY-ARCH": {
title: "WebRTC Security Architecture"
, href: "https://tools.ietf.org/html/draft-ietf-rtcweb-security-arch"
, authors: [
"E. Rescorla"
]
, status: "07 March 2015. Internet Draft (work in progress)"
, publisher: "IETF"
},
"RTCWEB-AUDIO": {
title: "WebRTC Audio Codec and Processing Requirements"
, href: "https://tools.ietf.org/html/draft-ietf-rtcweb-audio"
, authors: [
"JM. Valin"
,"C. Bran"
]
, status: "30 April 2015. Internet Draft (work in progress)"
, publisher: "IETF"
},
"RTCWEB-VIDEO": {
title: "WebRTC Video Processing and Codec Requirements"
, href: "https://tools.ietf.org/html/draft-ietf-rtcweb-video"
, authors: [
"A.B. Roach"
]
, status: "12 June 2015. Internet Draft (work in progress)"
, publisher: "IETF"
},
"RTP-MULTI-STREAM": {
title: "Sending Multiple Types of Media in a Single RTP Session"
, href: "https://tools.ietf.org/html/draft-ietf-avtcore-multi-media-rtp-session"
, authors: [
"M. Westerlund"
, "C. Perkins"
, "J. Lennox"
]
, status: "09 March 2015. Internet Draft (work in progress)"
, publisher: "IETF"
},
"RTP-USAGE": {
title: "Web Real-Time Communication (WebRTC): Media Transport and Use of RTP"
, href: "https://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage"
, authors: [
"C. Perkins"
, "M. Westerlund"
, "J. Ott"
]
, status: "12 June 2015. Internet Draft (work in progress)"
, publisher: "IETF"
},
"SILK": {
title: "SILK Speech Codec"
, href: "https://tools.ietf.org/html/draft-vos-silk"
, authors: [
"K. Vos"
, "S. Jensen"
, "K. Soerensen"
]
, status: "8 March 2010. Internet Draft (work in progress)"
, publisher: "IETF"
},
"VP8-RTP": {
title: "RTP Payload Format for VP8 Video"
, href: "https://tools.ietf.org/html/draft-ietf-payload-vp8"
, authors: [
"P. Westin",
"H. Lundin",
"M. Glover",
"J. Uberti",
"F. Galligan"
]
, status: "05 June 2015. Internet Draft (work in progress)"
, publisher: "IETF"
},
"WEBRTC-STATS": {
title: "Identifiers for WebRTC's Statistics API"
, href: "https://www.w3.org/TR/webrtc-stats/"
, authors: [
"Harald Alvestrand",
"Varun Singh"
]
, status: "06 February 2015 (work in progress)"
, publisher: "W3C"
}
}
};
</script>
</head>
<body>
<section id="abstract">
<p>
This document describes the Microsoft Edge browser's implementation of the ORTC API within the
Windows 10 Anniversary Update, as of August 2016.
The Microsoft Edge implementation differs from the most recent version of the
<a href="http://draft.ortc.org/">ORTC API</a> since ORTC has continued to evolve since the development of
Microsoft Edge got underway. In addition, Microsoft Edge does not implement every object or method
within the ORTC API, and includes extensions not currently incorporated within the specification.
Since both Microsoft Edge and this document represent works-in-progress, the behavior of Microsoft Edge can
change substantially from build-to-build. Please bring questions, errors and omissions to the attention of the
Edge RTC team (Bernard Aboba, bernarda@microsoft and Angelina Gambo, [email protected]).
</p>
</section>
<section id='sotd'>
</section>
<section id="overview*">
<h2><span class="secno"> </span>Overview</h2>
<p>Microsoft's Object Real-Time Communications (ORTC) API implementation enables the development of WebRTC applications.
<p>
Within Microsoft's implementation,
the relationship between the application and the objects, as well
as between the objects themselves is shown below.
Horizontal or slanted arrows denote the flow of media or data,
whereas vertical arrows denote interactions via methods and events.
</p>
<img alt="The non-normative ORTC Big Picture Diagram" src="images/ms-ortc.png" style="width:100%" />
<p>In the figure above, the <code><a>RTCRtpSender</a></code> (<a href="#rtcrtpsender*">Section 6</a>) encodes the
track provided as input, which is transported over a <code><a>RTCDtlsTransport</a></code>
(<a href="#rtcdtlstransport*">Section 4</a>) or an <code><a>RTCSrtpSdesTransport</a></code>
(<a href="#rtcsrtpsdestransport*">Section 5</a>).
Sending of Dual Tone Multi Frequency (DTMF) tones is supported via the
<code><a>RTCDtmfSender</a></code> (<a href="#rtcdtmfsender*">Section 9</a>).</p>
<p>The <code><a>RTCDtlsTransport</a></code> and <code><a>RTCSrtpSdesTransport</a></code> utilize an <code><a>RTCIceTransport</a></code>
(<a href="#rtcicetransport*">Section 3</a>) to select a communication path to reach the
receiving peer's <code><a>RTCIceTransport</a></code>, which is in turn associated with
an <code><a>RTCDtlsTransport</a></code> or <code><a>RTCSrtpSdesTransport</a></code> which de-multiplexes
media to the <code><a>RTCRtpReceiver</a></code> (<a href="#rtcrtpreceiver*">Section 7</a>).
The <code><a>RTCRtpReceiver</a></code> then decodes media, producing a track which is rendered
by an audio or video tag.</p>
<p>Several other objects also play a role.
The <code><a>RTCIceGatherer</a></code> (<a href="#rtcicegatherer*">Section 2</a>) gathers local ICE candidates for use by a single
<code><a>RTCIceTransport</a></code> object.</p>
<div class="note">
<p>Edge Interop Note: Since Microsoft Edge does not implement the data channel, the <code>RTCDataChannel</code> and <code>RTCSctpTransport</code> objects are not supported.
</p>
</div>
<p>Remaining sections of the specification fill in details relating to RTP capabilities and parameters,
operational statistics and compatibility with the
WebRTC 1.0 API.
RTP dictionaries are described in <a href="#rtcrtpdictionaries*">Section 8</a>,
the Statistics API is described in <a href="#statistics-api">Section 10</a>,
Microsoft extensions related to the H.264UC codec [[MS-H264PF]] and statistics are described in <a href="#msextensions*">Section 11</a>,
an event summary is provided in <a href="#event-summary">Section 12</a>,
a list of error codes is provided in <a href="#error-codes*">Section 13</a>
and WebRTC 1.0 compatibility issues are discussed in <a href="#webrtc-compat*">Section 14</a>.
</p>
<p>Complete examples are provided in <a href="#examples*">Section 15</a>.</p>
<section>
<h3>Terminology</h3>
<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.w2.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> and <dfn>MediaStreamTrack</dfn>
are defined in
[[!GETUSERMEDIA]].</p>
<p>For Scalable Video Coding (SVC), the terms single-session transmission (<dfn>SST</dfn>) and multi-session transmission (<dfn>MST</dfn>)
are defined in [[RFC6190]]. This specification only supports <a>SST</a> but not <a>MST</a>.
The term Single Real-time transport protocol stream Single Transport (<dfn>SRST</dfn>),
defined in [[GROUPING]] Section 3.7, refers to an SVC implementation that transmits all layers within a
single transport, using a single Real-time Transport Protocol (RTP) stream and synchronization source (SSRC).
The term Multiple RTP stream Single Transport (<dfn>MRST</dfn>), also defined in [[GROUPING]] Section 3.7, refers
to an implementation that transmits all layers within a single transport, using multiple RTP streams with
a distinct SSRC for each layer.</p>
</section>
</section>
<section id="rtcicegatherer*">
<h2>The RTCIceGatherer Object</h2>
<p>
The <dfn>RTCIceGatherer</dfn> gathers local host, server reflexive and relay candidates, as
well as enabling the retrieval of local Interactive Connectivity Establishment (ICE) parameters which can be exchanged in signaling.
</p>
<section id="rtcicegatherer-overview*">
<h3>Overview</h3>
<p>The Microsoft Edge implementation of the <code><a>RTCIceGatherer</a></code> does not support
the <var>component</var> or <var>state</var> attributes, nor does it support the
<code>close()</code> method or the <code>ongatherstatechange</code> event handler.
</p>
<div class="note">
<p>Edge Interop Note: In the Microsoft Edge implementation, an <code><a>RTCIceGatherer</a></code> instance can only
be associated to a single <code><a>RTCIceTransport</a></code>
object. As a result, forking scenarios are not supported.
The <code><a>RTCIceGatherer</a></code> prunes local candidates when its
associated <code><a>RTCIceTransport</a></code> object enters the "completed" state.
</p>
</div>
<p>As noted in [[!RFC5245]] Section 7.1.2.2, an incoming connectivity check contains an <code>ICE-CONTROLLING</code>
or <code>ICE-CONTROLLED</code> attribute, depending on the role of the ICE agent initiating the check.
Since an <code><a>RTCIceGatherer</a></code> object does not have a role, it cannot determine whether
to respond to an incoming connectivity check with a 487 (Role Conflict) error; however, it can validate
that an incoming connectivity check utilizes the correct local username fragment and password,
and if not, can respond with an
401 (Unauthorized) error, as described in [[!RFC5389]] Section 10.1.2.</p>
<p>For incoming connectivity checks that pass validation, the <code><a>RTCIceGatherer</a></code>
<em class="rfc2119" title="MUST">MUST</em>
buffer incoming connectivity checks so as to be able to provide them to associated
<code><a>RTCIceTransport</a></code> objects so that they can respond.</p>
</section>
<section id="rtcicegatherer-operation*">
<h3>Operation</h3>
<p>An <code><a>RTCIceGatherer</a></code> instance is constructed from an
<code><a>RTCIceGatherOptions</a></code> object.
</p>
</section>
<section id="rtcicegatherer-interface-definition*">
<h3>Interface Definition</h3>
<dl class="idl" title="[Constructor(RTCIceGatherOptions options)] interface RTCIceGatherer : RTCStatsProvider">
<dt>readonly attribute RTCIceComponent component</dt>
<dd><p>The component-id of the <code><a>RTCIceGatherer</a></code>.</p></dd>
<dt>RTCIceParameters getLocalParameters()</dt>
<dd><p>Obtain the ICE parameters of the <code><a>RTCIceGatherer</a></code>.</p></dd>
<dt>sequence<RTCIceCandidate> getLocalCandidates()</dt>
<dd><p>Retrieve the sequence of valid local candidates associated with the
<code><a>RTCIceGatherer</a></code>.
This retrieves all unpruned local candidates currently known (except for peer reflexive candidates),
even if an <code><a>onlocalcandidate</a></code>
event hasn't been processed yet. In Microsoft Edge, if <code>getLocalCandidates()</code> is called prior to emitting of the
first candidate, an exception is thrown.
</p></dd>
<dt>RTCIceGatherer createAssociatedGatherer()</dt>
<dd><p>Create an associated <code><a>RTCIceGatherer</a></code> for RTCP, with the same
<code><a>RTCIceParameters</a></code> and <code><a>RTCIceGatherOptions</a></code>.
If an <code><a>RTCIceGatherer</a></code> calls the method more than once,
or if <var>component</var> is "RTCP", throw
an <code>InvalidStateError</code> exception.
</p></dd>
<dt>attribute EventHandler? onerror</dt>
<dd><p>This event handler, of event handler type <code>error</code>,
<em class="rfc2119" title="MUST">MUST</em> be supported by all objects
implementing the <code><a>RTCIceGatherer</a></code> interface.
If TURN credentials are invalid, then this event <em class="rfc2119" title="MUST">MUST</em>
be fired.
</p></dd>
<dt>attribute EventHandler? onlocalcandidate</dt>
<dd><p>This event handler, of event handler event type <code>icecandidate</code>, uses
the <code><a>RTCIceGathererEvent</a></code> interface.
It <em class="rfc2119" title="MUST">MUST</em> be supported by all objects implementing the
<code><a>RTCIceGatherer</a></code> interface.
It receives events when a new local ICE candidate is available. Since ICE candidate gathering begins once
an <code><a>RTCIceGatherer</a></code> object is created,
<code>candidate</code> events are queued until an <code>onlocalcandidate</code> event handler is assigned.
When the final candidate is gathered, a <code>candidate</code> event occurs with an <code>RTCIceCandidateComplete</code> emitted.
</p></dd>
</dl>
</section>
<section id="rtciceparameters*">
<h3>The RTCIceParameters Object</h3>
<p>
The <dfn>RTCIceParameters</dfn> object includes the ICE username fragment and password.
</p>
<dl class="idl" title="dictionary RTCIceParameters">
<dt>DOMString usernameFragment</dt>
<dd><p>ICE username.</p></dd>
<dt>DOMString password</dt>
<dd><p>ICE password.</p></dd>
</dl>
</section>
<section id="rtcicecandidate*">
<h3>The RTCIceCandidate Object</h3>
<p>
The <dfn>RTCIceCandidate</dfn> object includes information relating to an ICE candidate.
</p>
<pre xml:space="preserve" class='example highlight'>
foundation: "abcd1234",
priority: 1694498815,
ip: "192.0.2.33",
protocol: "udp",
port: 10000,
type: "host"
};
</pre>
<dl class="idl" title="typedef (RTCIceCandidate or RTCIceCandidateComplete) RTCIceGatherCandidate">
</dl>
<dl class="idl" title="dictionary RTCIceCandidate">
<dt>DOMString foundation</dt>
<dd>
<p>A unique identifier that allows ICE to correlate candidates that appear on multiple <code><a>RTCIceTransport</a></code>s.</p>
</dd>
<dt>unsigned long priority</dt>
<dd><p>The assigned priority of the candidate. This is automatically populated by the browser.</p></dd>
<dt>DOMString ip</dt>
<dd><p>The IP address of the candidate.</p></dd>
<dt>RTCIceProtocol protocol</dt>
<dd><p>The protocol of the candidate (UDP/TCP).</p></dd>
<dt>unsigned short port</dt>
<dd><p>The port for the candidate.</p></dd>
<dt>DOMString interfaceType</dt>
<dd><p>The interface type for the candidate. Types include "lan", "wlan", "wwan" and "vpn".
Not supported in Microsoft Edge.</p></dd>
<dt>RTCIceCandidateType type</dt>
<dd><p>The type of candidate.</p></dd>
<dt>RTCIceTcpCandidateType tcpType=null</dt>
<dd><p>The type of TCP candidate.</p></dd>
<dt>DOMString relatedAddress=""</dt>
<dd>
<p>For candidates that are derived from others, such as relay or reflexive candidates, the <dfn>relatedAddress</dfn>
refers to the candidate that these are derived from. For host candidates, the <var>relatedAddress</var>
is set to the empty string.</p>
</dd>
<dt>unsigned short relatedPort=null</dt>
<dd>
<p>For candidates that are derived from others, such as relay or reflexive candidates, the <dfn>relatedPort</dfn>
refers to the host candidate that these are derived from. For host candidates, the <var>relatedPort</var>
is null.</p>
</dd>
</dl>
<section>
<h4>The RTCIceProtocol</h4>
<p>
The <dfn>RTCIceProtocol</dfn> includes the protocol of the ICE candidate.
</p>
<dl class="idl" title="enum RTCIceProtocol">
<dt>udp</dt>
<dd><p>A UDP candidate, as described in [[!RFC5245]].</p></dd>
<dt>tcp</dt>
<dd><p>A TCP candidate, as described in [[!RFC6544]].</p></dd>
</dl>
</section>
<section>
<h4>The RTCIceTcpCandidateType</h4>
<p>
The <dfn>RTCIceTcpCandidateType</dfn> includes the type of the ICE TCP candidate, as described in [[!RFC6544]].
Browsers MUST gather active TCP candidates and only active TCP candidates.
Servers and other endpoints MAY gather active, passive or so candidates.
</p>
<dl class="idl" title="enum RTCIceTcpCandidateType">
<dt>active</dt>
<dd><p>An active TCP candidate is one for which the transport will
attempt to open an outbound connection but will not receive incoming
connection requests.</p></dd>
<dt>passive</dt>
<dd><p>A passive TCP candidate is one for which the transport
will receive incoming connection attempts but not attempt a
connection.</p></dd>
<dt>so</dt>
<dd><p>An so candidate is one for which the transport will attempt
to open a connection simultaneously with its peer.</p></dd>
</dl>
</section>
<section>
<h4>The RTCIceCandidateType</h4>
<p>
The <dfn>RTCIceCandidateType</dfn> includes the type of the ICE candidate.
</p>
<dl class="idl" title="enum RTCIceCandidateType">
<dt>host</dt>
<dd><p>A host candidate.</p></dd>
<dt>srflx</dt>
<dd><p>A server reflexive candidate.</p></dd>
<dt>prflx</dt>
<dd><p>A peer reflexive candidate.</p></dd>
<dt>relay</dt>
<dd><p>A relay candidate.</p></dd>
</dl>
</section>
</section>
<section id="rtcicecandidatecomplete*">
<h3>dictionary RTCIceCandidateComplete</h3>
<p>
<dfn>RTCIceCandidateComplete</dfn> is a dictionary signifying that all <code>RTCIceCandidate</code>s are gathered.
<p>
<dl class="idl" title="dictionary RTCIceCandidateComplete">
</dl>
</section>
<section>
<h4>RTCIceGathererEvent</h4>
<p>The <code>icecandidate</code> event of the <code><a>RTCIceGatherer</a></code> object uses
the <code><a>RTCIceGathererEvent</a></code> interface.</p>
<p>Firing an
<code><a>RTCIceGathererEvent</a></code> event named
<var>e</var> with an <code><a>RTCIceGatherCandidate</a></code>
<var>candidate</var> means that an event with the name <var>e</var>,
which does not bubble (except where otherwise stated) and is not
cancelable (except where otherwise stated), and which uses the
<code><a>RTCIceGathererEvent</a></code> interface with the
<var>candidate</var> attribute set to the new ICE candidate,
<em class="rfc2119" title="MUST">MUST</em> be
created and dispatched at the given target.</p>
<dl class="idl" data-merge="RTCIceGathererEventInit" title=
"[Constructor(DOMString type, RTCIceGathererEventInit eventInitDict)] interface RTCIceGathererEvent : Event">
<dt>readonly attribute RTCIceGatherCandidate candidate</dt>
<dd>
<p>The <var>candidate</var> attribute is the
<code><a>RTCIceGatherCandidate</a></code> object with the new ICE
candidate that caused the event.
If <var>candidate</var> is of type <code>RTCIceCandidateComplete</code>,
there are no additional candidates.
</p>
</dd>
</dl>
<dl class="idl" title=
"dictionary RTCIceGathererEventInit : EventInit">
<dt>RTCIceGatherCandidate candidate</dt>
<dd>
<p>The ICE candidate that caused the event.</p>
</dd>
</dl>
</section>
<section id="rtcicegatheroptions*">
<h3>dictionary RTCIceGatherOptions</h3>
<p><dfn>RTCIceGatherOptions</dfn> provides options relating to the gathering of ICE candidates.</p>
<dl class="idl" title=
"dictionary RTCIceGatherOptions">
<dt>RTCIceGatherPolicy gatherPolicy</dt>
<dd><p>The ICE gather policy.</p></dd>
<dt>sequence<RTCIceServer> iceservers</dt>
<dd><p>The ICE servers to be configured.</p></dd>
</dl>
</section>
<section id="rtcicegatherpolicy*">
<h3>enum RTCIceGatherPolicy</h3>
<p><dfn>RTCIceGatherPolicy</dfn> denotes the policy relating to the gathering of ICE candidates.</p>
<dl class="idl" title="enum RTCIceGatherPolicy">
<dt>all</dt>
<dd><p>The ICE gatherer gathers all types of candidates when this value is specified.</p></dd>
<dt>nohost</dt>
<dd><p>The ICE gatherer gathers all ICE candidate types except for host candidates.</p></dd>
<dt>relay</dt>
<dd><p>The ICE gatherer <em class="rfc2119" title="MUST">MUST</em> only gather 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.</p></dd>
</dl>
</section>
<section id="rtciceserver*">
<h3>The RTCIceServer Object</h3>
<p>
The <dfn>RTCIceServer</dfn> is used to provide STUN or TURN server configuration.
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 of an array of <code<<a>RTCIceServer</a></code> objects supporting the STUN and TURN standards:
</p>
<pre xml:space="preserve" class='example highlight'>
[ { urls: "stun:stun1.example.net" } ,
{ urls:"turn:turn.example.org", username: "user", credential:"myPassword"}
]
</pre>
</p>
An example of an array of <code<<a>RTCIceServer</a></code> objects supporting [[MS-TURN]]:
</p>
<pre xml:space="preserve" class='example highlight'>
[ { urls: "stun:stun1.example.net" } ,
{ urls:"msturn:turn.example.org", username: "user", credential:"myPassword"}
]
</pre>
<div class="note">
<p>Edge Interop Note: Inclusion of the MSTURN URI triggers use of a proprietary TURN variant described in [[MS-TURN]], as well as proprietary ICE behavior
described in [[MS-ICE]] and [[MS-ICE2]].
The MSTURN URI should therefore not be used in scenarios requiring interoperability with other browsers.
The STUN URI is not currently supported.
</p>
</div>
<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 (such as the MSTURN URI, described in [[MS-TURN]]).</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> represents a TURN server, then this attribute specifies the credential to use with that TURN server.</p></dd>
</dl>
</section>
<section id="rtcicegatherer-initial-example*">
<h3>Example</h3>
<pre xml:space="preserve" class='example highlight'>
// Include some helper functions
import "helper.js";
// Create ICE gather options
var gatherOptions = new RTCIceGatherOptions();
gatherOptions.gatherPolicy = RTCIceGatherPolicy.relay;
gatherOptions.iceservers = [ { urls: "stun:stun1.example.net" } ,
{ urls:"turn:turn.example.org", username: "user", credential:"myPassword"} ];
// Create ICE gatherer objects
var iceGatherer = new RTCIceGatherer(gatherOptions);
// Prepare to signal local candidates as well as "end of candidates"
iceGatherer.onlocalcandidate = function (event) {
mySendLocalCandidate(event.candidate);
};
// Set up response function
mySignaller.onResponse = function(responseSignaller,response) {
// The Microsoft Edge ORTC implementation only supports a single response.
};
mySignaller.send({
"ice": iceGatherer.getLocalParameters()
});
</pre>
<pre xml:space="preserve" class='example highlight'>
// Helper functions used in all the examples (helper.js)
function trace(text) {
// This function is used for logging.
if (text[text.length - 1] === '\n') {
text = text.substring(0, text.length - 1);
}
if (window.performance) {
var now = (window.performance.now() / 1000).toFixed(3);
console.log(now + ': ' + text);
} else {
console.log(text);
}
}
function errorHandler (error) {
trace('Error encountered: ' + error.name);
}
function mySendLocalCandidate(candidate, component, kind){
// Set default values
kind = kind || "all";
component = component || RTCIceComponent.RTP;
parameters = parameters || null;