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zenhack committed Sep 3, 2021
2 parents c63021b + 33329ae commit d9779cc
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2 changes: 1 addition & 1 deletion ext-respec-config.js
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Expand Up @@ -4,7 +4,7 @@ var respecConfig = {
repoURL: "https://github.com/w3c/webrtc-extensions/",
branch: "master"
},
"xref": ["webidl", "webrtc", "hr-time", "mediacapture-streams", "webrtc-stats"],
"xref": ["html", "webidl", "webrtc", "hr-time", "mediacapture-streams", "webrtc-stats"],
"shortName": "webrtc-extensions",
"specStatus": "ED",
"subjectPrefix": "[webrtc-extensions]",
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133 changes: 103 additions & 30 deletions index.html
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Expand Up @@ -177,7 +177,7 @@ <h3>
};
</pre>
<p>
Let <dfn>[[\HeaderExtensionsToOffer]]</dfn> be an internal slot of the {{RTCRtpTransceiver}},
Let <dfn data-dfn-for=RTCRtpTransceiver>[[\HeaderExtensionsToOffer]]</dfn> be an internal slot of the {{RTCRtpTransceiver}},
initialized to the platform-specific list of implemented RTP header extensions.
The direction
attribute for all extensions that are mandatory to use MUST be initialized to
Expand All @@ -190,41 +190,41 @@ <h3>
BUNDLE is in use, per [[BUNDLE]] section 9.1.
</p>
<p>
Let <dfn>[[\HeaderExtensionsNegotiated]]</dfn> be an internal slot of the RTCRtpTransceiver,
Let <dfn data-dfn-for=RTCRtpTransceiver>{{RTCRtpTransceiver/[[HeaderExtensionsNegotiationed]]}}</dfn> be an internal slot of the {{RTCRtpTransceiver}},
initialized to an empty list.
</p>
<section>
<h2>Modifications to existing procedures</h2>
<p>
In the <a data-cite="WEBRTC#set-description">set a session description</a> algorithm, add a step
right after the step that sets transceiver.[[\Sender]].[[\SendCodecs]],
saying "For each transciever, set <a>[[\HeaderExtensionsNegotiated]]</a> to
saying "For each transciever, set {{RTCRtpTransceiver/[[HeaderExtensionsNegotiationed]]}} to
an empty list, and then
for each <code>"a=hdrext"</code> line, add an appropriate
{{RTCRtpHeaderExtensionCapability}} to the list.
</p>
<p>
<p>\
In the algorithms for generating initial offers in [[RTCWEB-JSEP]] section 5.2.1,
replace "for each supported RTP header extension, an "a=extmap" line, as specified in
[[RFC5285]], section 5" " with "For each RTP header extension "e"
listed in <a>[[\HeaderExtensionsToOffer]]</a> where {{RTCRtpHeaderExtensionCapability/direction}} is not {{RTCRtpTransceiverDirection/"stopped"}}, an "a=extmap"
listed in {{RTCRtpTransceiver/[[HeaderExtensionsToOffer]]}} where {{RTCRtpHeaderExtensionCapability/direction}} is not {{RTCRtpTransceiverDirection/"stopped"}}, an "a=extmap"
line, as specified in [[RFC5285]], section 5, with direction taken from "e"'s {{RTCRtpHeaderExtensionCapability/direction}}
attribute."
</p>
<p>
In the algorithm for generating subsequent offers in [[RTCWEB-JSEP]] section 5.2.2, replace "The
RTP header extensions MUST only include those that are present in the most recent answer"
with "For each RTP header extension listed in <a>[[\HeaderExtensionsToOffer]]</a>,
with "For each RTP header extension listed in {{RTCRtpTransceiver/[[HeaderExtensionsToOffer]]}},
and where {{RTCRtpHeaderExtensionCapability/direction}} is not {{RTCRtpTransceiverDirection/"stopped"}}, generate
an appropriate "a=extmap" line with "direction" set according to the rules of [[RFC5285]]
section 6, considering the {{RTCRtpHeaderExtensionCapability/direction}} in [[\HeaderExtensionsToOffer]] to indicate the
section 6, considering the {{RTCRtpHeaderExtensionCapability/direction}} in {{RTCRtpTransceiver/[[HeaderExtensionsToOffer]]}} to indicate the
answerer's desired usage".
</p>
<p>
In the algorithm for generating initial answers in [[RTCWEB-JSEP]] section 5.3.1, replace "For
each supported RTP header extension that is present in the offer" with "For each
supported RTP header extension that is present in the offer and is also present in
<a>[[\HeaderExtensionsToOffer]]</a> with a {{RTCRtpHeaderExtensionCapability/direction}} different from {{ RTCRtpTransceiverDirection/"stopped"}}".
{{RTCRtpTransceiver/[[HeaderExtensionsToOffer]]}} with a {{RTCRtpHeaderExtensionCapability/direction}} different from {{ RTCRtpTransceiverDirection/"stopped"}}".
</p>
<p class="note">
Since JSEP does not know about WebRTC internal slots, merging this change requires
Expand All @@ -246,11 +246,11 @@ <h2>Methods</h2>
If <var>ext</var>.{{RTCRtpHeaderExtensionCapability/uri}} value is missing, [=exception/throw=] a {{TypeError}}.
</li>
<li>If <var>ext</var>.{{RTCRtpHeaderExtensionCapability/uri}} value is not present in
[[\HeaderExtensionsToOffer]], [=exception/throw=] a {{NotSupportedError}}.
{{RTCRtpTransceiver/[[HeaderExtensionsToOffer]]}}, [=exception/throw=] a {{NotSupportedError}}.
</li>
<li>
Let <var>entry</var> be the entry with the same {{RTCRtpHeaderExtensionCapability/uri}} value in
[[\HeaderExtensionsToOffer]].
{{RTCRtpTransceiver/[[HeaderExtensionsToOffer]]}}.
</li>
<li>If <var>ext</var>.{{RTCRtpHeaderExtensionCapability/direction}} is not {{ RTCRtpTransceiverDirection/"sendrecv"}}, and {{RTCRtpHeaderExtensionParameters/uri}}
indicates a mandatory-to-use attribute that is required to be both
Expand All @@ -273,9 +273,9 @@ <h2>Methods</h2>
<h2>Attributes</h2>
<dl data-dfn-for=RTCRtpTransceiver>
<dt><dfn>headerExtensionsToOffer</dfn>, readonly</dt>
<dd>Returns the value of the [[\HeaderExtensionsToOffer]] slot.</dd>
<dd>Returns the value of the {{RTCRtpTransceiver/[[HeaderExtensionsToOffer]]}} slot.</dd>
<dt><dfn>headerExtensionsNegotiated</dfn>, readonly</dt>
<dd>Returns the value of the [[\HeaderExtensionsNegotiated]] slot.</dd>
<dd>Returns the value of the {{RTCRtpTransceiver/[[HeaderExtensionsNegotiationed]]}} slot.</dd>
</dl>
</section>
</section>
Expand Down Expand Up @@ -341,8 +341,7 @@ <h3>
by adding an additional internal slot and attribute.
</p>
<p>
Let {{RTCRtpReceiver}} objects have a
<dfn>[[\PlayoutDelay]]</dfn> internal slot initially
Let {{RTCRtpReceiver}} objects have a <dfn data-dfn-for=RTCRtpReceiver>[[\PlayoutDelay]]</dfn> internal slot initially
initialized to <code>null</code>.
</p>
<pre class="idl">partial interface RTCRtpReceiver {
Expand Down Expand Up @@ -380,10 +379,10 @@ <h2>Attributes</h2>
[[\AssociatedRemoteMediaStreams]]</a> internal slot, then it will
be synchronized with other tracks (for e.g. audio video
synchronization). This means that even if one of the paired tracks
is delayed through [[\PlayoutDelay]] then the User Agent
is delayed through {{RTCRtpReceiver/[[PlayoutDelay]]}} then the User Agent
synchronization mechanism will automatically delay all others
paired tracks. If multiple such paired tracks are delayed through
<a>[[\PlayoutDelay]]</a> by different amounts then the largest
{{RTCRtpReceiver/[[PlayoutDelay]]}} by different amounts then the largest
of those hints will take precedence in synchronization mechanism.
</p>
<p class="note">
Expand All @@ -392,7 +391,7 @@ <h2>Attributes</h2>
{{RTCInboundRtpStreamStats/jitterBufferEmittedCount}}.
</p>
<p>On getting, this attribute MUST return the value of the
<a>[[\PlayoutDelay]]</a> internal slot.</p>
{{RTCRtpReceiver/[[PlayoutDelay]]}} internal slot.</p>
<p>On setting, the User Agent MUST run the following steps:</p>
<ol>
<li>
Expand All @@ -408,7 +407,7 @@ <h2>Attributes</h2>
[=exception/throw=] a {{RangeError}} and abort these steps.</p>
</li>
<li>
<p>Set the value of <var>receiver</var>'s <a>[[\PlayoutDelay]]</a>
<p>Set the value of <var>receiver</var>'s {{RTCRtpReceiver/[[PlayoutDelay]]}}
internal slot to <var>delay</var>.</p>
</li>
<li>
Expand All @@ -427,7 +426,7 @@ <h2>Attributes</h2>
If User Agent chooses delay different from requested one,
for e.g. due to network conditions or physical memory
constraints, this is not reflected in the
<a>[[\PlayoutDelay]]</a> internal slot.
{{RTCRtpReceiver/[[PlayoutDelay]]}} internal slot.
</p>
</li>
<li>
Expand Down Expand Up @@ -919,26 +918,82 @@ <h3>
</dd>
</dl>
</section>
<section>
<h3>
{{RTCDataChannel}} extensions
</h3>
<p>This document extends {{RTCDataChannel}} by making it <a data-cite="!HTML/#transferable-objects">transferable</a>.</p>
This allows sending and receiving messages outside the context the connection was created, for instance in workers or third-party iframes.</p>
<div>
<p>The WebIDL changes are the following:
<pre class="idl">
[Exposed=(Window,Worker), Transferable]
partial interface RTCDataChannel {
};</pre>
</div>
<div>
<p>The <a data-cite="WEBRTC#dfn-create-an-rtcdatachannel">create an RTCDataChannel</a> algorithm is updated by adding the following steps after step 4 of the original algorithm:</p>
<ol>
<li><p>Initialize <var>channel</var>.`[[IsTransferable]]` to <code>true</code>.</p></li>
<li><p>Queue a task to run the following step:</p>
<ol>
<li><p>Set <var>channel</var>.`[[IsTransferable]]` to <code>false</code>.</p></li>
</ol>
</li>
</ol>
<p>This task needs to run before any task enqueued by the <a data-cite="WEBRTC#dfn-receive-an-rtcdatachannel-message">receiving messages on a data channel</a> algorithm for <var>channel</var>.
This ensures that no message is lost during the transfer of a {{RTCDataChannel}}.</p>
</div>
<div>The <a data-cite="WEBRTC#datachannel-send">send() algorithm</a> is modified by adding the following step after step 1 of the original algorithm:
<ol>
<li><p>Set <var>channel</var>.`[[IsTransferable]]` to <code>false</code>.</p></li>
</ol>
</div>
<div>
<p>The {{RTCDataChannel}} <a data-cite="!HTML/#transfer-steps">transfer steps</a>, given <var>value</var> and <var>dataHolder</var>, are:</p>
<ol>
<li><p>If <var>value</var>.`[[IsTransferable]]` is <code>false</code>, throw a "DataCloneError" DOMException.</p></li>
<li><p>Set <var>dataHolder</var>.`[[ReadyState]]` to <var>value</var>.<a href="WEBRTC#dfn-readystate">`[[ReadyState]]`</a>.</p></li>
<li><p>Set <var>dataHolder</var>.`[[DataChannelLabel]]` to <var>value</var>.<a href="WEBRTC#dfn-datachannellabel">`[[DataChannelLabel]]`</a>.</p></li>
<li><p>Set <var>dataHolder</var>.`[[Ordered]]` to <var>value</var>.<a href="WEBRTC#dfn-ordered">`[[Ordered]]`</a>.</p></li>
<li><p>Set <var>dataHolder</var>.`[[MaxPacketLifeTime]]` to <var>value</var>.<a href="WEBRTC#dfn-maxpacketlifetime"></a>.`[[MaxPacketLifeTime]]`</p></li>
<li><p>Set <var>dataHolder</var>.`[[MaxRetransmits]]` to <var>value</var>.<a href="WEBRTC#dfn-maxretransmits">`[[MaxRetransmits]]`</a>.</p></li>
<li><p>Set <var>dataHolder</var>.`[[DataChannelProtocol]]` to <var>value</var>.<a href="WEBRTC#dfn-datachannelprotocol">`[[DataChannelProtocol]]`</a>.</p></li>
<li><p>Set <var>dataHolder</var>.`[[Negotiated]]` to <var>value</var>.<a href="WEBRTC#dfn-negotiated">`[[Negotiated]]`</a>.</p></li>
<li><p>Set <var>dataHolder</var>.`[[DataChannelId]]` to <var>value</var>.<a href="WEBRTC#dfn-datachannelid">`[[DataChannelId]]`</a>.</p></li>
<li><p>Set <var>dataHolder</var>.`[[underlyingDataTransport]]` to <var>value</var> <a href="WEBRTC#dfn-data-transport">underlying data transport</a>.</p></li>
<li><p>Set <var>value</var>.`[[IsTransferable]]` to <code>false</code>.</p></li>
<li><p>Set <var>value</var>.`[[ReadyState]]` to <a data-cite="WEBRTC/#dom-rtcdatachannelstate-closed">"closed"</a>.</p></li>
</ol>
</div>
<div><p>The {{RTCDataChannel}} <a data-cite="!HTML/#transfer-receiving-steps">transfer-receiving steps</a>, given <var>dataHolder</var> and <var>channel</var>, are:</p>
<ol>
<li><p>Initialize <var>channel</var>.<a href="WEBRTC#dfn-readystate">`[[ReadyState]]`</a> to <var>dataHolder</var>.`[[ReadyState]]`.</p></li>
<li><p>Initialize <var>channel</var>.<a href="WEBRTC#dfn-datachannellabel">`[[DataChannelLabel]]`</a> to <var>dataHolder</var>.`[[\DataChannelLabel]]`.</p></li>
<li><p>Initialize <var>channel</var>.<a href="WEBRTC#dfn-ordered">`[[Ordered]]`</a> to <var>dataHolder</var>.`[[Ordered]]`.</p></li>
<li><p>Initialize <var>channel</var>.<a href="WEBRTC#dfn-maxpacketlifetime">`[[MaxPacketLifeTime]]`</a> to <var>dataHolder</var>.`[[MaxPacketLifeTime]]`.</p></li>
<li><p>Initialize <var>channel</var>.<a href="WEBRTC#dfn-maxretransmits">`[[MaxRetransmits]]`</a> to <var>dataHolder</var>.`[[MaxRetransmits]]`.</p></li>
<li><p>Initialize <var>channel</var>.<a href="WEBRTC#dfn-datachannelprotocol">`[[DataChannelProtocol]]`</a> to <var>dataHolder</var>.`[[DataChannelProtocol]]`.</p></li>
<li><p>Initialize <var>channel</var>.<a href="WEBRTC#dfn-negotiated">`[[Negotiated]]`</a> to <var>dataHolder</var>.`[[Negotiated]]`.</p></li>
<li><p>Initialize <var>channel</var>.<a href="WEBRTC#dfn-datachannelid">`[[DataChannelId]]`</a> to <var>dataHolder</var>.`[[DataChannelId]]`.</p></li>
<li><p>Initialize <var>channel</var> <a href="WEBRTC#dfn-data-transport">underlying data transport</a> to <var>dataHolder</var>.`[[underlyingDataTransport]]`.</p></li>
</ol>
<p>The above steps do not need to transfer <a href="WEBRTC#dfn-readystate">`[[BufferedAmount]]`</a> as its value will always be equal to <code>0</code>.
The reason is an {{RTCDataChannel}} can be transferred only if its <a data-cite="WEBRTC#datachannel-send">send() algorithm</a> was not called prior the transfer.</p>
<p>If the <a href="WEBRTC#dfn-data-transport">underlying data transport</a> is closed at the time of the <a data-cite="!HTML/#transfer-receiving-steps">transfer-receiving steps</a>,
the {{RTCDataChannel}} object will be closed by running the <a href="WEBRTC#announcing-a-data-channel-as-closed">announcing a data channel as closed</a> algorithm immediately after the <a data-cite="!HTML/#transfer-receiving-steps">transfer-receiving steps</a>.</p>
</div>
</section>
<section class="informative">
<h2>
Privacy and Security Considerations
Security Considerations
</h2>
<p>
This section is non-normative; it specifies no new behaviour.
The overall security considerations of the general set
of APIs and protocols used in WebRTC are described in
[[?RFC8827]].
</p>
<section>
<h2>
Revealing IP addresses
</h2>
<p>
The extensions defined in this document do not reveal additional
information on IP addresses beyond that already described in
[[WEBRTC]] Section 13.2.
</p>
</section>
<section>
<h2>
Impact on local network
Expand All @@ -959,6 +1014,24 @@ <h2>
IDs, as well as CSRCs.
</p>
</section>
</section>
<section class="informative">
<h2>
Privacy Considerations
</h2>
<p>
This section is non-normative; it specifies no new behaviour.
</p>
<section>
<h2>
Revealing IP addresses
</h2>
<p>
The extensions defined in this document do not reveal additional
information on IP addresses beyond that already described in
[[WEBRTC]] Section 13.2.
</p>
</section>
<section>
<h2>
Persistent information exposed by WebRTC
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