rfc8829xml2.original.xml   rfc8829.xml 
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ipr="trust200902">
<front> <front>
<title abbrev="JSEP">JavaScript Session Establishment <title abbrev="JSEP">JavaScript Session Establishment Protocol (JSEP)</title
Protocol</title> >
<seriesInfo name="RFC" value="8829"/>
<!-- [rfced] Please insert any keywords (beyond those that appear in the
title) for use on https://www.rfc-editor.org/search -->
<!-- [rfced] We have updated the title to include "JSEP"; please let us know
any objections.
Original:
JavaScript Session Establishment Protocol
Currently:
JavaScript Session Establishment Protocol (JSEP) -->
<author fullname="Justin Uberti" initials="J." surname="Uberti"> <author fullname="Justin Uberti" initials="J." surname="Uberti">
<organization>Google</organization> <organization>Google</organization>
<address> <address>
<postal> <postal>
<street>747 6th St S</street> <street>747 6th Street South</street>
<city>Kirkland</city> <city>Kirkland</city>
<region>WA</region> <region>WA</region>
<code>98033</code> <code>98033</code>
<country>USA</country> <country>United States of America</country>
</postal> </postal>
<email>justin@uberti.name</email> <email>justin@uberti.name</email>
</address> </address>
</author> </author>
<author fullname="Cullen Jennings" initials="C." <author fullname="Cullen Jennings" initials="C." surname="Jennings">
surname="Jennings">
<organization>Cisco</organization> <organization>Cisco</organization>
<address> <address>
<postal> <postal>
<street>400 3rd Avenue SW</street> <street>400 3rd Avenue SW</street>
<city>Calgary</city> <city>Calgary</city>
<region>AB</region> <region>AB</region>
<code>T2P 4H2</code> <code>T2P 4H2</code>
<country>Canada</country> <country>Canada</country>
</postal> </postal>
<email>fluffy@iii.ca</email> <email>fluffy@iii.ca</email>
</address> </address>
</author> </author>
<author fullname="Eric Rescorla" initials="E.K." surname="Rescorla" <author fullname="Eric Rescorla" initials="E." surname="Rescorla" role="edit
role="editor"> or">
<organization>Mozilla</organization> <organization>Mozilla</organization>
<address> <address>
<postal> <postal>
<street>331 Evelyn Ave</street> <street>331 E. Evelyn Ave.</street>
<city>Mountain View</city> <city>Mountain View</city>
<region>CA</region> <region>CA</region>
<code>94041</code> <code>94041</code>
<country>USA</country> <country>United States of America</country>
</postal> </postal>
<email>ekr@rtfm.com</email> <email>ekr@rtfm.com</email>
</address> </address>
</author> </author>
<date />
<area>RAI</area>
<abstract>
<!-- [rfced] Eric, we note that Mozilla and RTFM are listed as your
affiliation in various documents within the cluster. Please review and let us
know if any updates are needed.
-->
<date month="July" year="2020"/>
<abstract>
<t>This document describes the mechanisms for allowing a <t>This document describes the mechanisms for allowing a
JavaScript application to control the signaling plane of a JavaScript application to control the signaling plane of a
multimedia session via the interface specified in the W3C multimedia session via the interface specified in the W3C
RTCPeerConnection API, and discusses how this relates to existing RTCPeerConnection API and discusses how this relates to existing
signaling protocols.</t> signaling protocols.</t>
</abstract> </abstract>
</front> </front>
<middle> <middle>
<section title="Introduction" anchor="sec.introduction">
<t>This document describes how the W3C WEBRTC RTCPeerConnection <!-- [rfced] Adam Roach previously requested that "All other documents in
interface Cluster 238 that currently reference RFC 5245 should be updated to
<xref target="W3C.webrtc"></xref> is used to control the setup, reference RFC 8445. ... any other references to RFC 5245 that I may
management and teardown of a multimedia session.</t> have overlooked should also be updated."
<section title="General Design of JSEP"
anchor="sec.general-design-of-jsep"> We believe this document intentionally refers to both RFCs. Please let us
know if any updates are needed.
-->
<section anchor="sec.introduction" numbered="true" toc="default">
<name>Introduction</name>
<t>This document describes how the W3C Web Real-Time Communication (WebRTC
) RTCPeerConnection
interface
<xref target="W3C.webrtc" format="default"/> is used to control the setup,
management, and teardown of a multimedia session.</t>
<section anchor="sec.general-design-of-jsep" numbered="true" toc="default"
>
<name>General Design of JSEP</name>
<t>WebRTC call setup has been designed to focus on controlling <t>WebRTC call setup has been designed to focus on controlling
the media plane, leaving signaling plane behavior up to the the media plane, leaving signaling-plane behavior up to the
application as much as possible. The rationale is that application as much as possible. The rationale is that
different applications may prefer to use different protocols, different applications may prefer to use different protocols,
such as the existing SIP call signaling protocol, or something such as the existing SIP call signaling protocol, or something
custom to the particular application, perhaps for a novel use custom to the particular application, perhaps for a novel use
case. In this approach, the key information that needs to be case. In this approach, the key information that needs to be
exchanged is the multimedia session description, which exchanged is the multimedia session description, which
specifies the necessary transport and media configuration specifies the transport and media configuration
information necessary to establish the media plane.</t> information necessary to establish the media plane.</t>
<t>With these considerations in mind, this document describes <t>With these considerations in mind, this document describes
the JavaScript Session Establishment Protocol (JSEP) that the JavaScript Session Establishment Protocol (JSEP), which
allows for full control of the signaling state machine from allows for full control of the signaling state machine from
JavaScript. As described above, JSEP assumes a model in which a JavaScript. As described above, JSEP assumes a model in which a
JavaScript application executes inside a runtime containing JavaScript application executes inside a runtime containing
WebRTC APIs (the "JSEP implementation"). The JSEP WebRTC APIs (the "JSEP implementation"). The JSEP
implementation is almost entirely divorced from the core implementation is almost entirely divorced from the core
signaling flow, which is instead handled by the JavaScript signaling flow, which is instead handled by the JavaScript
making use of two interfaces: (1) passing in local and remote making use of two interfaces: (1) passing in local and remote
session descriptions and (2) interacting with the ICE state session descriptions and (2) interacting with the Interactive
machine. The combination of the JSEP implementation and the Connectivity Establishment (ICE) state
machine <xref target="RFC8445"/>. The combination of the JSEP implementa
tion and the
JavaScript application is referred to throughout this document JavaScript application is referred to throughout this document
as a "JSEP endpoint".</t> as a "JSEP endpoint".</t>
<t>In this document, the use of JSEP is described as if it <t>In this document, the use of JSEP is described as if it
always occurs between two JSEP endpoints. Note though in many always occurs between two JSEP endpoints. Note, though, that in many
cases it will actually be between a JSEP endpoint and some kind cases it will actually be between a JSEP endpoint and some kind
of server, such as a gateway or MCU. This distinction is of server, such as a gateway or Multipoint Control Unit (MCU). This dist inction is
invisible to the JSEP endpoint; it just follows the invisible to the JSEP endpoint; it just follows the
instructions it is given via the API.</t> instructions it is given via the API.</t>
<t>JSEP's handling of session descriptions is simple and <t>JSEP's handling of session descriptions is simple and
straightforward. Whenever an offer/answer exchange is needed, straightforward. Whenever an offer/answer exchange is needed,
the initiating side creates an offer by calling a createOffer() the initiating side creates an offer by calling a createOffer()
API. The application then uses that offer to set up its local API. The application then uses that offer to set up its local
config via the setLocalDescription() API. The offer is finally config via the setLocalDescription() API. The offer is finally
sent off to the remote side over its preferred signaling sent off to the remote side over its preferred signaling
mechanism (e.g., WebSockets); upon receipt of that offer, the mechanism (e.g., WebSockets); upon receipt of that offer, the
remote party installs it using the setRemoteDescription() remote party installs it using the setRemoteDescription()
API.</t> API.</t>
<t>To complete the offer/answer exchange, the remote party uses <t>To complete the offer/answer exchange, the remote party uses
the createAnswer() API to generate an appropriate answer, the createAnswer() API to generate an appropriate answer,
applies it using the setLocalDescription() API, and sends the applies it using the setLocalDescription() API, and sends the
answer back to the initiator over the signaling channel. When answer back to the initiator over the signaling channel. When
the initiator gets that answer, it installs it using the the initiator gets that answer, it installs it using the
setRemoteDescription() API, and initial setup is complete. This setRemoteDescription() API, and initial setup is complete. This
process can be repeated for additional offer/answer process can be repeated for additional offer/answer
exchanges.</t> exchanges.</t>
<t>Regarding ICE <t>Regarding ICE
<xref target="RFC8445"></xref>, JSEP decouples the ICE state <xref target="RFC8445" format="default"/>, JSEP decouples the ICE state
machine from the overall signaling state machine, as the ICE machine from the overall signaling state machine, as the ICE
state machine must remain in the JSEP implementation, because state machine must remain in the JSEP implementation, because
only the implementation has the necessary knowledge of only the implementation has the necessary knowledge of
candidates and other transport information. Performing this candidates and other transport information. Performing this
separation provides additional flexibility in protocols that separation provides additional flexibility in protocols that
decouple session descriptions from transport. For instance, in decouple session descriptions from transport. For instance, in
traditional SIP, each offer or answer is self-contained, traditional SIP, each offer or answer is self-contained,
including both the session descriptions and the transport including both the session descriptions and the transport
information. However, information. However,
<xref target="I-D.ietf-mmusic-trickle-ice-sip" /> allows SIP to <xref target="RFC8840" format="default"/> allows SIP to
be used with trickle ICE be used with Trickle ICE
<xref target="I-D.ietf-ice-trickle" />, in which the session <xref target="RFC8838" format="default"/>, in which the session
description can be sent immediately and the transport description can be sent immediately and the transport
information can be sent when available. Sending transport information can be sent when available. Sending transport
information separately can allow for faster ICE and DTLS information separately can allow for faster ICE and DTLS
startup, since ICE checks can start as soon as any transport startup, since ICE checks can start as soon as any transport
information is available rather than waiting for all of it. information is available rather than waiting for all of it.
JSEP's decoupling of the ICE and signaling state machines JSEP's decoupling of the ICE and signaling state machines
allows it to accommodate either model.</t> allows it to accommodate either model.</t>
<t>Through its abstraction of signaling, the JSEP approach does <t>Through its abstraction of signaling, the JSEP approach does
require the application to be aware of the signaling process. require the application to be aware of the signaling process.
While the application does not need to understand the contents While the application does not need to understand the contents
of session descriptions to set up a call, the application must of session descriptions to set up a call, the application must
call the right APIs at the right times, convert the session call the right APIs at the right times, convert the session
descriptions and ICE information into the defined messages of descriptions and ICE information into the defined messages of
its chosen signaling protocol, and perform the reverse its chosen signaling protocol, and perform the reverse
conversion on the messages it receives from the other side.</t> conversion on the messages it receives from the other side.</t>
<t>One way to make life easier for the application is to <t>One way to make life easier for the application is to
provide a JavaScript library that hides this complexity from provide a JavaScript library that hides this complexity from
the developer; said library would implement a given signaling the developer; said library would implement a given signaling
protocol along with its state machine and serialization code, protocol along with its state machine and serialization code,
presenting a higher level call-oriented interface to the presenting a higher-level call-oriented interface to the
application developer. For example, libraries exist to adapt application developer. For example, libraries exist to adapt
the JSEP API into an API suitable for a SIP or XMPP. Thus, JSEP the JSEP API into an API suitable for a SIP interface or an Extensible M
essaging
and Presence Protocol (XMPP) interface <xref target="RFC6120"/>.
<!-- [rfced] Section 1.1: For ease of the reader, and per other
documents in this cluster (cluster C238), we expanded "XMPP," cited
RFC 6120, and added RFC 6120 to the list of Informative References.
Please let us know any concerns.
Original:
For example,
libraries exist to adapt the JSEP API into an API suitable for a SIP
or XMPP.
Currently:
For example,
libraries exist to adapt the JSEP API into an API suitable for a SIP
interface or an Extensible Messaging and Presence Protocol (XMPP)
interface [RFC6120].
...
[RFC6120] Saint-Andre, P., "Extensible Messaging and Presence
Protocol (XMPP): Core", RFC 6120, DOI 10.17487/RFC6120,
March 2011, <https://www.rfc-editor.org/info/rfc6120>. -->
Thus, JSEP
provides greater control for the experienced developer without provides greater control for the experienced developer without
forcing any additional complexity on the novice developer.</t> forcing any additional complexity on the novice developer.</t>
</section> </section>
<section title="Other Approaches Considered" <section anchor="sec.other-approaches-consider" numbered="true" toc="defau
anchor="sec.other-approaches-consider"> lt">
<name>Other Approaches Considered</name>
<t>One approach that was considered instead of JSEP was to <t>One approach that was considered instead of JSEP was to
include a lightweight signaling protocol. Instead of providing include a lightweight signaling protocol. Instead of providing
session descriptions to the API, the API would produce and session descriptions to the API, the API would produce and
consume messages from this protocol. While providing a more consume messages from this protocol. While providing a more
high-level API, this put more control of signaling within the high-level API, this put more control of signaling within the
JSEP implementation, forcing it to have to understand and JSEP implementation, forcing it to have to understand and
handle concepts like signaling glare (see handle concepts like signaling glare (see
<xref target="RFC3264" />, Section 4).</t> <xref target="RFC3264" sectionFormat="comma" section="4"/>).</t>
<t>A second approach that was considered but not chosen was to <t>A second approach that was considered but not chosen was to
decouple the management of the media control objects from decouple the management of the media control objects from
session descriptions, instead offering APIs that would control session descriptions, instead offering APIs that would control
each component directly. This was rejected based on the each component directly. This was rejected based on the
argument that requiring exposure of this level of complexity to argument that requiring exposure of this level of complexity to
the application programmer would not be beneficial; it would the application programmer would not be beneficial; it would
result in an API where even a simple example would require a result in an API where even a simple example would require a
significant amount of code to orchestrate all the needed significant amount of code to orchestrate all the needed
interactions, as well as creating a large API surface that interactions, as well as creating a large API surface that
needed to be agreed upon and documented. In addition, these API needed to be agreed upon and documented.
<!-- [rfced] Section 1.2: We had trouble parsing this sentence.
If the suggested text is not correct, please clarify "as well as
creating ..."
Original (the previous sentence is included for context):
A second approach that was considered but not chosen was to decouple
the management of the media control objects from session
descriptions, instead offering APIs that would control each component
directly. This was rejected based on the argument that requiring
exposure of this level of complexity to the application programmer
would not be beneficial; it would result in an API where even a
simple example would require a significant amount of code to
orchestrate all the needed interactions, as well as creating a large
API surface that needed to be agreed upon and documented.
Suggested:
This was rejected based on the argument that requiring
exposure of this level of complexity to the application programmer
would not be beneficial; it would (1) result in an API where even a
simple example would require a significant amount of code to
orchestrate all the needed interactions and (2) create a large
API surface that needed to be agreed upon and documented. -->
In addition, these API
points could be called in any order, resulting in a more points could be called in any order, resulting in a more
complex set of interactions with the media subsystem than the complex set of interactions with the media subsystem than the
JSEP approach, which specifies how session descriptions are to JSEP approach, which specifies how session descriptions are to
be evaluated and applied.</t> be evaluated and applied.</t>
<t>One variation on JSEP that was considered was to keep the <t>One variation on JSEP that was considered was to keep the
basic session description-oriented API, but to move the basic session-description-oriented API but to move the
mechanism for generating offers and answers out of the JSEP mechanism for generating offers and answers out of the JSEP
implementation. Instead of providing createOffer/createAnswer implementation. Instead of providing createOffer/createAnswer
methods within the implementation, this approach would instead methods within the implementation, this approach would instead
expose a getCapabilities API which would provide the expose a getCapabilities API, which would provide the
application with the information it needed in order to generate application with the information it needed in order to generate
its own session descriptions. This increases the amount of work its own session descriptions. This increases the amount of work
that the application needs to do; it needs to know how to that the application needs to do; it needs to know how to
generate session descriptions from capabilities, and especially generate session descriptions from capabilities, and especially
how to generate the correct answer from an arbitrary offer and how to generate the correct answer from an arbitrary offer and
the supported capabilities. While this could certainly be the supported capabilities. While this could certainly be
addressed by using a library like the one mentioned above, it addressed by using a library like the one mentioned above, it
basically forces the use of said library even for a simple basically forces the use of said library even for a simple
example. Providing createOffer/createAnswer avoids this example. Providing createOffer/createAnswer avoids this
problem.</t> problem.</t>
</section> </section>
</section> </section>
<section title="Terminology" anchor="sec.terminology"> <section anchor="sec.terminology" numbered="true" toc="default">
<name>Terminology</name>
<t>The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL <t>The key words "<bcp14>MUST</bcp14>", "<bcp14>MUST NOT</bcp14>",
NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "<bcp14>REQUIRED</bcp14>", "<bcp14>SHALL</bcp14>",
"OPTIONAL" in this document are to be interpreted as described in "<bcp14>SHALL NOT</bcp14>", "<bcp14>SHOULD</bcp14>",
"<bcp14>SHOULD NOT</bcp14>",
<xref target="RFC2119"></xref>.</t> "<bcp14>RECOMMENDED</bcp14>", "<bcp14>NOT RECOMMENDED</bcp14>",
"<bcp14>MAY</bcp14>", and "<bcp14>OPTIONAL</bcp14>" in this document are
to be interpreted as described in BCP&nbsp;14 <xref target="RFC2119"/>
<xref target="RFC8174"/> when, and only when, they appear in all capitals,
as shown here.</t>
</section> </section>
<section title="Semantics and Syntax" <section anchor="sec.semantics-and-syntax" numbered="true" toc="default">
anchor="sec.semantics-and-syntax"> <name>Semantics and Syntax</name>
<section title="Signaling Model" anchor="sec.signaling-model"> <section anchor="sec.signaling-model" numbered="true" toc="default">
<name>Signaling Model</name>
<t>JSEP does not specify a particular signaling model or state <t>JSEP does not specify a particular signaling model or state
machine, other than the generic need to exchange session machine, other than the generic need to exchange session
descriptions in the fashion described by descriptions in the fashion described by
<xref target="RFC3264"></xref> (offer/answer) in order for both <xref target="RFC3264" format="default"/> (offer/answer) in order for bo th
sides of the session to know how to conduct the session. JSEP sides of the session to know how to conduct the session. JSEP
provides mechanisms to create offers and answers, as well as to provides mechanisms to create offers and answers, as well as to
apply them to a session. However, the JSEP implementation is apply them to a session. However, the JSEP implementation is
totally decoupled from the actual mechanism by which these totally decoupled from the actual mechanism by which these
offers and answers are communicated to the remote side, offers and answers are communicated to the remote side,
including addressing, retransmission, forking, and glare including addressing, retransmission, forking, and glare
handling. These issues are left entirely up to the application; handling. These issues are left entirely up to the application;
the application has complete control over which offers and the application has complete control over which offers and
answers get handed to the implementation, and when.</t> answers get handed to the implementation, and when.</t>
<figure anchor="fig-sigModel" title="JSEP Signaling Model"> <figure anchor="fig-sigModel">
<artwork> <name>JSEP Signaling Model</name>
<![CDATA[ <artwork name="" type="" align="left" alt=""><![CDATA[
+-----------+ +-----------+ +-----------+ +-----------+
| Web App |<--- App-Specific Signaling -->| Web App | | Web App |<--- App-Specific Signaling -->| Web App |
+-----------+ +-----------+ +-----------+ +-----------+
^ ^ ^ ^
| SDP | SDP | SDP | SDP
V V V V
+-----------+ +-----------+ +-----------+ +-----------+
| JSEP |<----------- Media ------------>| JSEP | | JSEP |<----------- Media ------------>| JSEP |
| Impl. | | Impl. | | Impl. | | Impl. |
+-----------+ +-----------+ +-----------+ +-----------+ ]]></artwork>
]]>
</artwork>
</figure> </figure>
</section> </section>
<section title="Session Descriptions and State Machine" <section anchor="sec.session-descriptions-and-state-machine" numbered="tru
anchor="sec.session-descriptions-and-state-machine"> e" toc="default">
<name>Session Descriptions and State Machine</name>
<t>In order to establish the media plane, the JSEP <t>In order to establish the media plane, the JSEP
implementation needs specific parameters to indicate what to implementation needs specific parameters to indicate what to
transmit to the remote side, as well as how to handle the media transmit to the remote side, as well as how to handle the media
that is received. These parameters are determined by the that is received. These parameters are determined by the
exchange of session descriptions in offers and answers, and exchange of session descriptions in offers and answers, and
there are certain details to this process that must be handled there are certain details to this process that must be handled
in the JSEP APIs.</t> in the JSEP APIs.</t>
<t>Whether a session description applies to the local side or <t>Whether a session description applies to the local side or
the remote side affects the meaning of that description. For the remote side affects the meaning of that description. For
example, the list of codecs sent to a remote party indicates example, the list of codecs sent to a remote party indicates
what the local side is willing to receive, which, when what the local side is willing to receive, which, when
intersected with the set of codecs the remote side supports, intersected with the set of codecs the remote side supports,
specifies what the remote side should send. However, not all specifies what the remote side should send. However, not all
parameters follow this rule; some parameters are declarative parameters follow this rule; some parameters are declarative,
and the remote side MUST either accept them or reject them and the remote side <bcp14>MUST</bcp14> either accept them or reject the
m
altogether. An example of such a parameter is the DTLS altogether. An example of such a parameter is the DTLS
fingerprints fingerprints
<xref target="RFC8122"></xref>, which are calculated based on <xref target="RFC8122" format="default"/>, which are calculated based on
the local certificate(s) offered, and are not subject to the local certificate(s) offered and are not subject to
negotiation.</t> negotiation.
<!-- [rfced] Section 3.2: We do not see any mention of DTLS in
RFC 8122; we only see "TLS." Will this citation be clear to
readers?
Original:
An example of such a parameter is the DTLS
fingerprints [RFC8122], which are calculated based on the local
certificate(s) offered, and are not subject to negotiation.
Possibly:
An example of such a parameter is the TLS
fingerprints [RFC8122] as used in the context of DTLS [RFC6347];
these fingerprints are calculated based on the local certificate(s)
offered and are not subject to negotiation. -->
</t>
<t>In addition, various RFCs put different conditions on the <t>In addition, various RFCs put different conditions on the
format of offers versus answers. For example, an offer may format of offers versus answers. For example, an offer may
propose an arbitrary number of m= sections (i.e., media propose an arbitrary number of "m=" sections (i.e., media
descriptions as described in descriptions as described in
<xref target="RFC4566" />, Section 5.14), but an answer must <xref target="RFC4566" sectionFormat="comma" section="5.14"/>), but an a nswer must
contain the exact same number as the offer.</t> contain the exact same number as the offer.</t>
<t>Lastly, while the exact media parameters are known only
<t>Lastly, while the exact media parameters are only known only
after an offer and an answer have been exchanged, the offerer after an offer and an answer have been exchanged, the offerer
may receive ICE checks, and possibly media (e.g., in the case may receive ICE checks, and possibly media (e.g., in the case
of a re-offer after a connection has been established) before of a re-offer after a connection has been established) before
it receives an answer. To properly process incoming media in it receives an answer. To properly process incoming media in
this case, the offerer's media handler must be aware of the this case, the offerer's media handler must be aware of the
details of the offer before the answer arrives.</t> details of the offer before the answer arrives.</t>
<t>Therefore, in order to handle session descriptions properly, <t>Therefore, in order to handle session descriptions properly,
the JSEP implementation needs: the JSEP implementation needs:
<list style="numbers"> </t>
<ol spacing="normal" type="1">
<t>To know if a session description pertains to the local or <li>To know if a session description pertains to the local or
remote side.</t> remote side.</li>
<li>To know if a session description is an offer or an
<t>To know if a session description is an offer or an answer.</li>
answer.</t> <li>To allow the offer to be specified independently of the
answer.</li>
<t>To allow the offer to be specified independently of the </ol>
answer.</t> <t>JSEP addresses this by adding both setLocalDescription
</list>JSEP addresses this by adding both setLocalDescription
and setRemoteDescription methods and having session description and setRemoteDescription methods and having session description
objects contain a type field indicating the type of session objects contain a type field indicating the type of session
description being supplied. This satisfies the requirements description being supplied. This satisfies the requirements
listed above for both the offerer, who first calls listed above for both the offerer, who first calls
setLocalDescription(sdp [offer]) and then later setLocalDescription(sdp [offer]) and then later
setRemoteDescription(sdp [answer]), as well as for the setRemoteDescription(sdp [answer]), and the
answerer, who first calls setRemoteDescription(sdp [offer]) and answerer, who first calls setRemoteDescription(sdp [offer]) and
then later setLocalDescription(sdp [answer]).</t> then later setLocalDescription(sdp [answer]).</t>
<t>During the offer/answer exchange, the outstanding offer is <t>During the offer/answer exchange, the outstanding offer is
considered to be "pending" at the offerer and the answerer, as considered to be "pending" at the offerer and the answerer, as
it may either be accepted or rejected. If this is a re-offer, it may be either accepted or rejected. If this is a re-offer,
each side will also have "current" local and remote each side will also have "current" local and remote
descriptions, which reflect the result of the last offer/answer descriptions, which reflect the result of the last offer/answer
exchange. Sections exchange. Sections
<xref target="sec.pendinglocaldescription" />, <xref target="sec.pendinglocaldescription" format="counter"/>,
<xref target="sec.pendingremotedescription" />, <xref target="sec.pendingremotedescription" format="counter"/>,
<xref target="sec.currentlocaldescription" />, and <xref target="sec.currentlocaldescription" format="counter"/>, and
<xref target="sec.currentremotedescription" />, provide more <xref target="sec.currentremotedescription" format="counter"/> provide m
ore
detail on pending and current descriptions.</t> detail on pending and current descriptions.</t>
<t>JSEP also allows for an answer to be treated as provisional <t>JSEP also allows for an answer to be treated as provisional
by the application. Provisional answers provide a way for an by the application. Provisional answers provide a way for an
answerer to communicate initial session parameters back to the answerer to communicate initial session parameters back to the
offerer, in order to allow the session to begin, while allowing offerer, in order to allow the session to begin, while allowing
a final answer to be specified later. This concept of a final a final answer to be specified later. This concept of a final
answer is important to the offer/answer model; when such an answer is important to the offer/answer model; when such an
answer is received, any extra resources allocated by the caller answer is received, any extra resources allocated by the caller
can be released, now that the exact session configuration is can be released, now that the exact session configuration is
known. These "resources" can include things like extra ICE known. These "resources" can include things like extra ICE
components, TURN candidates, or video decoders. Provisional components, Traversal Using Relays around NAT (TURN) candidates, or vide o decoders. Provisional
answers, on the other hand, do no such deallocation; as a answers, on the other hand, do no such deallocation; as a
result, multiple dissimilar provisional answers, with their own result, multiple dissimilar provisional answers, with their own
codec choices, transport parameters, etc., can be received and codec choices, transport parameters, etc., can be received and
applied during call setup. Note that the final answer itself applied during call setup. Note that the final answer itself
may be different than any received provisional answers.</t> may be different than any received provisional answers.</t>
<t>In <t>In
<xref target="RFC3264"></xref>, the constraint at the signaling <xref target="RFC3264" format="default"/>, the constraint at the signali ng
level is that only one offer can be outstanding for a given level is that only one offer can be outstanding for a given
session, but at the media stack level, a new offer can be session, but at the media stack level, a new offer can be
generated at any point. For example, when using SIP for generated at any point. For example, when using SIP for
signaling, if one offer is sent, then cancelled using a SIP signaling, if one offer is sent and is then canceled using a SIP
CANCEL, another offer can be generated even though no answer CANCEL, another offer can be generated even though no answer
was received for the first offer. To support this, the JSEP was received for the first offer. To support this, the JSEP
media layer can provide an offer via the createOffer() method media layer can provide an offer via the createOffer() method
whenever the JavaScript application needs one for the whenever the JavaScript application needs one for the
signaling. The answerer can send back zero or more provisional signaling. The answerer can send back zero or more provisional
answers, and finally end the offer-answer exchange by sending a answers and then finally end the offer/answer exchange by sending a
final answer. The state machine for this is as follows:</t> final answer. The state machine for this is as follows:</t>
<figure anchor="fig-state-machine">
<t> <name>JSEP State Machine</name>
<figure anchor="fig-state-machine" <artwork name="" type="" align="left" alt=""><![CDATA[
title="JSEP State Machine">
<artwork>
<![CDATA[
setRemote(OFFER) setLocal(PRANSWER) setRemote(OFFER) setLocal(PRANSWER)
/-----\ /-----\ /-----\ /-----\
| | | | | | | |
v | v | v | v |
+---------------+ | +---------------+ | +---------------+ | +---------------+ |
| |----/ | |----/ | |----/ | |----/
| have- | setLocal(PRANSWER) | have- | | have- | setLocal(PRANSWER) | have- |
| remote-offer |------------------- >| local-pranswer| | remote-offer |------------------- >| local-pranswer|
| | | | | | | |
| | | | | | | |
skipping to change at line 404 skipping to change at line 469
+---------------+ +---------------+ +---------------+ +---------------+
| | | | | | | |
| have- | setRemote(PRANSWER) |have- | | have- | setRemote(PRANSWER) |have- |
| local-offer |------------------- >|remote-pranswer| | local-offer |------------------- >|remote-pranswer|
| | | | | | | |
| |----\ | |----\ | |----\ | |----\
+---------------+ | +---------------+ | +---------------+ | +---------------+ |
^ | ^ | ^ | ^ |
| | | | | | | |
\-----/ \-----/ \-----/ \-----/
setLocal(OFFER) setRemote(PRANSWER) setLocal(OFFER) setRemote(PRANSWER) ]]></artwo
]]> rk>
</artwork> </figure>
</figure> <t>Aside from these state transitions, there is no other
</t>
<t>Aside from these state transitions there is no other
difference between the handling of provisional ("pranswer") and difference between the handling of provisional ("pranswer") and
final ("answer") answers.</t> final ("answer") answers.</t>
</section> </section>
<section title="Session Description Format" <section anchor="sec.session-description-forma" numbered="true" toc="defau
anchor="sec.session-description-forma"> lt">
<name>Session Description Format</name>
<t>JSEP's session descriptions use SDP syntax for their <t>JSEP's session descriptions use Session Description Protocol (SDP) sy
ntax for their
internal representation. While this format is not optimal for internal representation. While this format is not optimal for
manipulation from JavaScript, it is widely accepted, and manipulation from JavaScript, it is widely accepted and is
frequently updated with new features; any alternate encoding of frequently updated with new features; any alternate encoding of
session descriptions would have to keep pace with the changes session descriptions would have to keep pace with the changes
to SDP, at least until the time that this new encoding eclipsed to SDP, at least until the time that this new encoding eclipsed
SDP in popularity.</t> SDP in popularity.</t>
<t>However, to provide for future flexibility, the SDP syntax <t>However, to provide for future flexibility, the SDP syntax
is encapsulated within a SessionDescription object, which can is encapsulated within a SessionDescription object, which can
be constructed from SDP, and be serialized out to SDP. If be constructed from SDP and be serialized out to SDP. If
future specifications agree on a JSON format for session future specifications agree on a JSON format for session
descriptions, we could easily enable this object to generate descriptions, we could easily enable this object to generate
and consume that JSON.</t> and consume that JSON.</t>
<t>As detailed below, most applications should be able to treat <t>As detailed below, most applications should be able to treat
the SessionDescriptions produced and consumed by these various the SessionDescriptions produced and consumed by these various
API calls as opaque blobs; that is, the application will not API calls as opaque blobs; that is, the application will not
need to read or change them.</t> need to read or change them.</t>
</section> </section>
<section title="Session Description Control" <section anchor="sec.session-description-ctrl" numbered="true" toc="defaul
anchor="sec.session-description-ctrl"> t">
<name>Session Description Control</name>
<t>In order to give the application control over various common <t>In order to give the application control over various common
session parameters, JSEP provides control surfaces which tell session parameters, JSEP provides control surfaces that tell
the JSEP implementation how to generate session descriptions. the JSEP implementation how to generate session descriptions.
This avoids the need for JavaScript to modify session This avoids the need for JavaScript to modify session
descriptions in most cases.</t> descriptions in most cases.</t>
<t>Changes to these objects result in changes to the session <t>Changes to these objects result in changes to the session
descriptions generated by subsequent createOffer/Answer descriptions generated by subsequent createOffer/createAnswer
calls.</t> calls.</t>
<section title="RtpTransceivers" anchor="sec.rtptransceivers"> <section anchor="sec.rtptransceivers" numbered="true" toc="default">
<name>RtpTransceivers</name>
<t>RtpTransceivers allow the application to control the RTP <t>RtpTransceivers allow the application to control the RTP
media associated with one m= section. Each RtpTransceiver has media associated with one "m=" section. Each RtpTransceiver has
an RtpSender and an RtpReceiver, which an application can use an RtpSender and an RtpReceiver, which an application can use
to control the sending and receiving of RTP media. The to control the sending and receiving of RTP media. The
application may also modify the RtpTransceiver directly, for application may also modify the RtpTransceiver directly, for
instance, by stopping it.</t> instance, by stopping it.</t>
<t>RtpTransceivers generally have a 1:1 mapping with "m="
<t>RtpTransceivers generally have a 1:1 mapping with m= sections, although there may be more RtpTransceivers than "m="
sections, although there may be more RtpTransceivers than m=
sections when RtpTransceivers are created but not yet sections when RtpTransceivers are created but not yet
associated with a m= section, or if RtpTransceivers have been associated with an "m=" section, or if RtpTransceivers have been
stopped and disassociated from m= sections. An RtpTransceiver stopped and disassociated from "m=" sections. An RtpTransceiver
is said to be associated with an m= section if its mid is said to be associated with an "m=" section if its
property is non-null; otherwise it is said to be media identification (mid) property is non-null; otherwise, it is said
disassociated. The associated m= section is determined using to be
a mapping between transceivers and m= section indices, formed disassociated. The associated "m=" section is determined using
a mapping between transceivers and "m=" section indices, formed
when creating an offer or applying a remote offer.</t> when creating an offer or applying a remote offer.</t>
<t>An RtpTransceiver is never associated with more than one <t>An RtpTransceiver is never associated with more than one
m= section, and once a session description is applied, a m= "m=" section, and once a session description is applied, an "m="
section is always associated with exactly one RtpTransceiver. section is always associated with exactly one RtpTransceiver.
However, in certain cases where a m= section has been However, in certain cases where an "m=" section has been
rejected, as discussed in rejected, as discussed in
<xref target="sec.subsequent-offers" /> below, that m= section <xref target="sec.subsequent-offers" format="default"/> below, that "m =" section
will be "recycled" and associated with a new RtpTransceiver will be "recycled" and associated with a new RtpTransceiver
with a new mid value.</t> with a new mid value.</t>
<t>RtpTransceivers can be created explicitly by the <t>RtpTransceivers can be created explicitly by the
application or implicitly by calling setRemoteDescription application or implicitly by calling setRemoteDescription
with an offer that adds new m= sections.</t> with an offer that adds new "m=" sections.</t>
</section> </section>
<section title="RtpSenders" anchor="sec.rtpsenders"> <section anchor="sec.rtpsenders" numbered="true" toc="default">
<name>RtpSenders</name>
<t>RtpSenders allow the application to control how RTP media <t>RtpSenders allow the application to control how RTP media
is sent. An RtpSender is conceptually responsible for the is sent. An RtpSender is conceptually responsible for the
outgoing RTP stream(s) described by an m= section. This outgoing RTP stream(s) described by an "m=" section. This
includes encoding the attached MediaStreamTrack, sending RTP includes encoding the attached MediaStreamTrack, sending RTP
media packets, and generating/processing RTCP for the media packets, and generating/processing the RTP Control Protocol (RTC P) for the
outgoing RTP streams(s).</t> outgoing RTP streams(s).</t>
</section> </section>
<section title="RtpReceivers" anchor="sec.rtpreceivers"> <section anchor="sec.rtpreceivers" numbered="true" toc="default">
<name>RtpReceivers</name>
<t>RtpReceivers allow the application to inspect how RTP <t>RtpReceivers allow the application to inspect how RTP
media is received. An RtpReceiver is conceptually responsible media is received. An RtpReceiver is conceptually responsible
for the incoming RTP stream(s) described by an m= section. for the incoming RTP stream(s) described by an "m=" section.
This includes processing received RTP media packets, decoding This includes processing received RTP media packets, decoding
the incoming stream(s) to produce a remote MediaStreamTrack, the incoming stream(s) to produce a remote MediaStreamTrack,
and generating/processing RTCP for the incoming RTP and generating/processing RTCP for the incoming RTP
stream(s).</t> stream(s).</t>
</section> </section>
</section> </section>
<section title="ICE" anchor="sec.ice"> <section anchor="sec.ice" numbered="true" toc="default">
<section title="ICE Gathering Overview" <name>ICE</name>
anchor="sec.ice-gather-overview"> <section anchor="sec.ice-gather-overview" numbered="true" toc="default">
<name>ICE Gathering Overview</name>
<t>JSEP gathers ICE candidates as needed by the application. <t>JSEP gathers ICE candidates as needed by the application.
Collection of ICE candidates is referred to as a gathering Collection of ICE candidates is referred to as a gathering
phase, and this is triggered either by the addition of a new phase, and this is triggered either by the addition of a new
or recycled m= section to the local session description, or or recycled "m=" section to the local session description or by
new ICE credentials in the description, indicating an ICE new ICE credentials in the description, indicating an ICE
restart. Use of new ICE credentials can be triggered restart. Use of new ICE credentials can be triggered
explicitly by the application, or implicitly by the JSEP explicitly by the application or implicitly by the JSEP
implementation in response to changes in the ICE implementation in response to changes in the ICE
configuration.</t> configuration.</t>
<t>When the ICE configuration changes in a way that requires <t>When the ICE configuration changes in a way that requires
a new gathering phase, a 'needs-ice-restart' bit is set. When a new gathering phase, a 'needs-ice-restart' bit is set. When
this bit is set, calls to the createOffer API will generate this bit is set, calls to the createOffer API will generate
new ICE credentials. This bit is cleared by a call to the new ICE credentials. This bit is cleared by a call to the
setLocalDescription API with new ICE credentials from either setLocalDescription API with new ICE credentials from either
an offer or an answer, i.e., from either a local- or an offer or an answer, i.e., from either a locally or
remote-initiated ICE restart.</t> remotely initiated ICE restart.</t>
<t>When a new gathering phase starts, the ICE agent will <t>When a new gathering phase starts, the ICE agent will
notify the application that gathering is occurring through an notify the application that gathering is occurring through an
event. Then, when each new ICE candidate becomes available, event. Then, when each new ICE candidate becomes available,
the ICE agent will supply it to the application via an the ICE agent will supply it to the application via an
additional event; these candidates will also automatically be additional event; these candidates will also automatically be
added to the current and/or pending local session added to the current and/or pending local session
description. Finally, when all candidates have been gathered, description. Finally, when all candidates have been gathered,
an event will be dispatched to signal that the gathering an event will be dispatched to signal that the gathering
process is complete.</t> process is complete.</t>
<t>Note that gathering phases only gather the candidates <t>Note that gathering phases only gather the candidates
needed by new/recycled/restarting m= sections; other m= needed by new/recycled/restarting "m=" sections; other "m="
sections continue to use their existing candidates. Also, if sections continue to use their existing candidates. Also, if
an m= section is bundled (either by a successful bundle an "m=" section is bundled (either by a successful bundle
negotiation or by being marked as bundle-only), then negotiation or by being marked as bundle-only), then
candidates will be gathered and exchanged for that m= section candidates will be gathered and exchanged for that "m=" section
if and only if its MID is a BUNDLE-tag, as described in if and only if its MID item is a BUNDLE-tag, as described in
<xref target="I-D.ietf-mmusic-sdp-bundle-negotiation" />.</t> <xref target="RFC8843" format="default"/>.</t>
</section> </section>
<section title="ICE Candidate Trickling" <section anchor="sec.ice-candidate-trickling" numbered="true" toc="defau
anchor="sec.ice-candidate-trickling"> lt">
<name>ICE Candidate Trickling</name>
<t>Candidate trickling is a technique through which a caller <t>Candidate trickling is a technique through which a caller
may incrementally provide candidates to the callee after the may incrementally provide candidates to the callee after the
initial offer has been dispatched; the semantics of "Trickle initial offer has been dispatched; the semantics of "Trickle
ICE" are defined in ICE" are defined in
<xref target="I-D.ietf-ice-trickle"></xref>. This process <xref target="RFC8838" format="default"/>. This process
allows the callee to begin acting upon the call and setting allows the callee to begin acting upon the call and setting
up the ICE (and perhaps DTLS) connections immediately, up the ICE (and perhaps DTLS) connections immediately,
without having to wait for the caller to gather all possible without having to wait for the caller to gather all possible
candidates. This results in faster media setup in cases where candidates. This results in faster media setup in cases where
gathering is not performed prior to initiating the call.</t> gathering is not performed prior to initiating the call.</t>
<t>JSEP supports optional candidate trickling by providing <t>JSEP supports optional candidate trickling by providing
APIs, as described above, that provide control and feedback APIs, as described above, that provide control and feedback
on the ICE candidate gathering process. Applications that on the ICE candidate gathering process. Applications that
support candidate trickling can send the initial offer support candidate trickling can send the initial offer
immediately and send individual candidates when they get the immediately and send individual candidates when they get
notified of a new candidate; applications that do not support notified of a new candidate; applications that do not support
this feature can simply wait for the indication that this feature can simply wait for the indication that
gathering is complete, and then create and send their offer, gathering is complete, and then create and send their offer,
with all the candidates, at this time.</t> with all the candidates, at that time.</t>
<t>Upon receipt of trickled candidates, the receiving <t>Upon receipt of trickled candidates, the receiving
application will supply them to its ICE agent. This triggers application will supply them to its ICE agent. This triggers
the ICE agent to start using the new remote candidates for the ICE agent to start using the new remote candidates for
connectivity checks.</t> connectivity checks.</t>
<section title="ICE Candidate Format" <section anchor="sec.ice-candidate-format" numbered="true" toc="defaul
anchor="sec.ice-candidate-format"> t">
<name>ICE Candidate Format</name>
<t>In JSEP, ICE candidates are abstracted by an <t>In JSEP, ICE candidates are abstracted by an
IceCandidate object, and as with session descriptions, SDP IceCandidate object, and as with session descriptions, SDP
syntax is used for the internal representation.</t> syntax is used for the internal representation.</t>
<t>The candidate details are specified in an IceCandidate <t>The candidate details are specified in an IceCandidate
field, using the same SDP syntax as the field, using the same SDP syntax as the
"candidate-attribute" field defined in "candidate-attribute" field defined in
<xref target="I-D.ietf-mmusic-ice-sip-sdp" />, <xref target="RFC8839" sectionFormat="comma" section="5.1"/>. Note t
Section 4.1. Note that this hat this
field does not contain an "a=" prefix, as indicated in the field does not contain an "a=" prefix, as indicated in the
following example:</t> following example:</t>
<figure>
<artwork>
<![CDATA[
candidate:1 1 UDP 1694498815 192.0.2.33 10000 typ host
]]>
</artwork>
</figure>
<sourcecode name="" type="sdp"><![CDATA[
candidate:1 1 UDP 1694498815 192.0.2.33 10000 typ host ]]></sourcecode>
<t>The IceCandidate object contains a field to indicate <t>The IceCandidate object contains a field to indicate
which ICE ufrag it is associated with, as defined in which ICE ufrag it is associated with, as defined in
<xref target="I-D.ietf-mmusic-ice-sip-sdp" />, <xref target="RFC8839" sectionFormat="comma" section="5.4"/>. This v
Section 4.4. This value is used alue is used
to determine which session description (and thereby which to determine which session description (and thereby which
gathering phase) this IceCandidate belongs to, which helps gathering phase) this IceCandidate belongs to, which helps
resolve ambiguities during ICE restarts. If this field is resolve ambiguities during ICE restarts. If this field is
absent in a received IceCandidate (perhaps when absent in a received IceCandidate (perhaps when
communicating with a non-JSEP endpoint), the most recently communicating with a non-JSEP endpoint), the most recently
received session description is assumed.</t> received session description is assumed.</t>
<t>The IceCandidate object also contains fields to indicate <t>The IceCandidate object also contains fields to indicate
which m= section it is associated with, which can be which "m=" section it is associated with, which can be
identified in one of two ways, either by a m= section identified in one of two ways: either by an "m=" section
index, or a MID. The m= section index is a zero-based index or by a MID. The "m=" section index is a zero-based
index, with index N referring to the N+1th m= section in index, with index N referring to the N+1th "m=" section in
the session description referenced by this IceCandidate. the session description referenced by this IceCandidate.
The MID is a "media stream identification" value, as The MID is a "media stream identification" value, as
defined in defined in
<xref target="RFC5888"></xref>, Section 4, which provides a <xref target="RFC5888" sectionFormat="comma" section="4"/>, which pr
more robust way to identify the m= section in the session ovides a
more robust way to identify the "m=" section in the session
description, using the MID of the associated RtpTransceiver description, using the MID of the associated RtpTransceiver
object (which may have been locally generated by the object (which may have been locally generated by the
answerer when interacting with a non-JSEP endpoint that answerer when interacting with a non-JSEP endpoint that
does not support the MID attribute, as discussed in does not support the MID attribute, as discussed in
<xref target="sec.applying-a-remote-desc" /> below). If the <xref target="sec.applying-a-remote-desc" format="default"/> below).
MID field is present in a received IceCandidate, it MUST be If the
used for identification; otherwise, the m= section index is MID field is present in a received IceCandidate, it <bcp14>MUST</bcp
14> be
used for identification; otherwise, the "m=" section index is
used instead.</t> used instead.</t>
<t>When creating an IceCandidate object, JSEP <t>When creating an IceCandidate object, JSEP
implementations MUST populate each of the candidate, ufrag, implementations <bcp14>MUST</bcp14> populate each of the candidate,
m= section index, and MID fields. Implementations MUST also ufrag,
"m=" section index, and MID fields. Implementations <bcp14>MUST</bcp
14> also
be prepared to receive objects with some fields missing, as be prepared to receive objects with some fields missing, as
mentioned above.</t> mentioned above.</t>
</section> </section>
</section> </section>
<section title="ICE Candidate Policy" <section anchor="sec.ice-candidate-policy" numbered="true" toc="default"
anchor="sec.ice-candidate-policy"> >
<name>ICE Candidate Policy</name>
<t>Typically, when gathering ICE candidates, the JSEP <t>Typically, when gathering ICE candidates, the JSEP
implementation will gather all possible forms of initial implementation will gather all possible forms of initial
candidates - host, server reflexive, and relay. However, in candidates -- host, server-reflexive, and relay.
<!-- [rfced] Sections 3.5.3 and 8: Per author feedback for
RFC 8839 and per other documents in this cluster, we hyphenated the
term "server reflexive". Please let us know any objections.
Original:
Typically, when gathering ICE candidates, the JSEP implementation
will gather all possible forms of initial candidates - host, server
reflexive, and relay.
...
Thus,
for instance, it is not possible to prevent the remote peer from
learning your public IP address by removing server reflexive
candidates.
Currently:
Typically, when gathering ICE candidates, the JSEP implementation
will gather all possible forms of initial candidates - host, server-
reflexive, and relay.
...
Thus,
for instance, it is not possible to prevent the remote peer from
learning your public IP address by removing server-reflexive
candidates. -->
However, in
certain cases, applications may want to have more specific certain cases, applications may want to have more specific
control over the gathering process, due to privacy or related control over the gathering process, due to privacy or related
concerns. For example, one may want to only use relay concerns. For example, one may want to only use relay
candidates, to leak as little location information as candidates, to leak as little location information as
possible (keeping in mind that this choice comes with possible (keeping in mind that this choice comes with
corresponding operational costs). To accomplish this, JSEP corresponding operational costs). To accomplish this, JSEP
allows the application to restrict which ICE candidates are allows the application to restrict which ICE candidates are
used in a session. Note that this filtering is applied on top used in a session. Note that this filtering is applied on top
of any restrictions the implementation chooses to enforce of any restrictions the implementation chooses to enforce
regarding which IP addresses are permitted for the regarding which IP addresses are permitted for the
application, as discussed in application, as discussed in
<xref target="I-D.ietf-rtcweb-ip-handling" />.</t> <xref target="RFC8828" format="default"/>.</t>
<t>There may also be cases where the application wants to <t>There may also be cases where the application wants to
change which types of candidates are used while the session change which types of candidates are used while the session
is active. A prime example is where a callee may initially is active. A prime example is where a callee may initially
want to use only relay candidates, to avoid leaking location want to use only relay candidates, to avoid leaking location
information to an arbitrary caller, but then change to use information to an arbitrary caller, but then change to use
all candidates (for lower operational cost) once the user has all candidates (for lower operational cost) once the user has
indicated they want to take the call. For this scenario, the indicated that they want to take the call. For this scenario, the
JSEP implementation MUST allow the candidate policy to be JSEP implementation <bcp14>MUST</bcp14> allow the candidate policy to
be
changed in mid-session, subject to the aforementioned changed in mid-session, subject to the aforementioned
interactions with local policy.</t> interactions with local policy.</t>
<t>To administer the ICE candidate policy, the JSEP <t>To administer the ICE candidate policy, the JSEP
implementation will determine the current setting at the implementation will determine the current setting at the
start of each gathering phase. Then, during the gathering start of each gathering phase. Then, during the gathering
phase, the implementation MUST NOT expose candidates phase, the implementation <bcp14>MUST NOT</bcp14> expose candidates
disallowed by the current policy to the application, use them disallowed by the current policy to the application, use them
as the source of connectivity checks, or indirectly expose as the source of connectivity checks, or indirectly expose
them via other fields, such as the raddr/rport attributes for them via other fields, such as the raddr/rport attributes for
other ICE candidates. Later, if a different policy is other ICE candidates. Later, if a different policy is
specified by the application, the application can apply it by specified by the application, the application can apply it by
kicking off a new gathering phase via an ICE restart.</t> kicking off a new gathering phase via an ICE restart.</t>
</section> </section>
<section title="ICE Candidate Pool" <section anchor="sec.ice-candidate-pool" numbered="true" toc="default">
anchor="sec.ice-candidate-pool"> <name>ICE Candidate Pool</name>
<t>JSEP applications typically inform the JSEP implementation <t>JSEP applications typically inform the JSEP implementation
to begin ICE gathering via the information supplied to to begin ICE gathering via the information supplied to
setLocalDescription, as the local description indicates the setLocalDescription, as the local description indicates the
number of ICE components which will be needed and for which number of ICE components that will be needed and for which
candidates must be gathered. However, to accelerate cases candidates must be gathered. However, to accelerate cases
where the application knows the number of ICE components to where the application knows the number of ICE components to
use ahead of time, it may ask the implementation to gather a use ahead of time, it may ask the implementation to gather a
pool of potential ICE candidates to help ensure rapid media pool of potential ICE candidates to help ensure rapid media
setup.</t> setup.</t>
<!-- [rfced] We suggest clarifying "goes to gather" - perhaps "gathers",
"starts to gather", "prepares to gather", etc. "goes to gather" may be
confusing for some readers.
<t>When setLocalDescription is eventually called, and the Original:
When setLocalDescription is eventually called, and the JSEP
implementation goes to gather the needed ICE candidates, it SHOULD
start by checking if any candidates are available in the pool.
-->
<t>When setLocalDescription is eventually called and the
JSEP implementation goes to gather the needed ICE candidates, JSEP implementation goes to gather the needed ICE candidates,
it SHOULD start by checking if any candidates are available it <bcp14>SHOULD</bcp14> start by checking if any candidates are avail
in the pool. If there are candidates in the pool, they SHOULD able
in the pool. If there are candidates in the pool, they <bcp14>SHOULD</
bcp14>
be handed to the application immediately via the ICE be handed to the application immediately via the ICE
candidate event. If the pool becomes depleted, either because candidate event. If the pool becomes depleted, either because
a larger-than-expected number of ICE components is used, or a larger-than-expected number of ICE components are used or
because the pool has not had enough time to gather because the pool has not had enough time to gather
candidates, the remaining candidates are gathered as usual. candidates, the remaining candidates are gathered as usual.
This only occurs for the first offer/answer exchange, after This only occurs for the first offer/answer exchange, after
which the candidate pool is emptied and no longer used.</t> which the candidate pool is emptied and no longer used.</t>
<t>One example of where this concept is useful is an <t>One example of where this concept is useful is an
application that expects an incoming call at some point in application that expects an incoming call at some point in
the future, and wants to minimize the time it takes to the future, and wants to minimize the time it takes to
establish connectivity, to avoid clipping of initial media. establish connectivity, to avoid clipping of initial media.
By pre-gathering candidates into the pool, it can exchange By pre-gathering candidates into the pool, it can exchange
and start sending connectivity checks from these candidates and start sending connectivity checks from these candidates
almost immediately upon receipt of a call. Note though that almost immediately upon receipt of a call. Note, though, that
by holding on to these pre-gathered candidates, which will be by holding on to these pre-gathered candidates, which will be
kept alive as long as they may be needed, the application kept alive as long as they may be needed, the application
will consume resources on the STUN/TURN servers it is will consume resources on the STUN/TURN servers it is
using.</t> using. ("STUN" stands for "Session Traversal Utilities for NAT".)</t>
</section> </section>
<section title="ICE Versions"> <section numbered="true" toc="default">
<t>While this specification formally relies on <xref <name>ICE Versions</name>
target="RFC8445"></xref>, at the time of its publication, the <t>While this specification formally relies on <xref target="RFC8445"
format="default"/>, at the time of its publication, the
majority of WebRTC implementations support the version majority of WebRTC implementations support the version
of ICE described in <xref target="RFC5245"></xref>. The use of of ICE described in <xref target="RFC5245" format="default"/>. The "ic
the "ice2" attribute defined in <xref target="RFC8445"></xref> e2" attribute defined in <xref target="RFC8445" format="default"/>
can be used to detect the version in use by a remote endpoint can be used to detect the version in use by a remote endpoint
and to provide a smooth transition from the older specification and to provide a smooth transition from the older specification
to the newer one. Implementations MUST be able to accept remote to the newer one. Implementations <bcp14>MUST</bcp14> be able to acce pt remote
descriptions that do not have the "ice2" attribute.</t> descriptions that do not have the "ice2" attribute.</t>
</section> </section>
</section> </section>
<section anchor="sec.imageattr" title="Video Size Negotiation"> <section anchor="sec.imageattr" numbered="true" toc="default">
<name>Video Size Negotiation</name>
<t>Video size negotiation is the process through which a <t>Video size negotiation is the process through which a
receiver can use the "a=imageattr" SDP attribute receiver can use the "a=imageattr" SDP attribute
<xref target="RFC6236" /> to indicate what video frame sizes it <xref target="RFC6236" format="default"/> to indicate what video frame s izes it
is capable of receiving. A receiver may have hard limits on is capable of receiving. A receiver may have hard limits on
what its video decoder can process, or it may have some maximum what its video decoder can process, or it may have some maximum
set by policy. By specifying these limits in an "a=imageattr" set by policy. By specifying these limits in an "a=imageattr"
attribute, JSEP endpoints can attempt to ensure that the remote attribute, JSEP endpoints can attempt to ensure that the remote
sender transmits video at an acceptable resolution. However, sender transmits video at an acceptable resolution. However,
when communicating with a non-JSEP endpoint that does not when communicating with a non-JSEP endpoint that does not
understand this attribute, any signaled limits may be exceeded, understand this attribute, any signaled limits may be exceeded,
and the JSEP implementation MUST handle this gracefully, e.g., and the JSEP implementation <bcp14>MUST</bcp14> handle this gracefully, e.g.,
by discarding the video.</t> by discarding the video.</t>
<t>Note that certain codecs support transmission of samples <t>Note that certain codecs support transmission of samples
with aspect ratios other than 1.0 (i.e., non-square pixels). with aspect ratios other than 1.0 (i.e., non-square pixels).
JSEP implementations will not transmit non-square pixels, but JSEP implementations will not transmit non-square pixels but
SHOULD receive and render such video with the correct aspect <bcp14>SHOULD</bcp14> receive and render such video with the correct asp
ect
ratio. However, sample aspect ratio has no impact on the size ratio. However, sample aspect ratio has no impact on the size
negotiation described below; all dimensions are measured in negotiation described below; all dimensions are measured in
pixels, whether square or not.</t> pixels, whether square or not.</t>
<section anchor="sec.creating-imageattr" <section anchor="sec.creating-imageattr" numbered="true" toc="default">
title="Creating an imageattr Attribute"> <name>Creating an imageattr Attribute</name>
<t>The receiver will first intersect any known local limits <t>The receiver will first intersect any known local limits
(e.g., hardware decoder capababilities, local policy) to (e.g., hardware decoder capabilities, local policy) to
determine the absolute minimum and maximum sizes it can determine the absolute minimum and maximum sizes it can
receive. If there are no known local limits, the receive. If there are no known local limits, the
"a=imageattr" attribute SHOULD be omitted. If these local "a=imageattr" attribute <bcp14>SHOULD</bcp14> be omitted. If these loc al
limits preclude receiving any video, i.e., the degenerate limits preclude receiving any video, i.e., the degenerate
case of no permitted resolutions, the "a=imageattr" attribute case of no permitted resolutions, the "a=imageattr" attribute
MUST be omitted, and the m= section MUST be marked as <bcp14>MUST</bcp14> be omitted, and the "m=" section <bcp14>MUST</bcp1 4> be marked as
sendonly/inactive, as appropriate.</t> sendonly/inactive, as appropriate.</t>
<t>Otherwise, an "a=imageattr" attribute is created with a
<t>Otherwise, an "a=imageattr" attribute is created with
"recv" direction, and the resulting resolution space formed "recv" direction, and the resulting resolution space formed
from the aforementioned intersection is used to specify its from the aforementioned intersection is used to specify its
minimum and maximum x= and y= values.</t> minimum and maximum "x=" and "y=" values.</t>
<t>The rules here express a single set of preferences, and <t>The rules here express a single set of preferences, and
therefore, the "a=imageattr" q= value is not important. It therefore, the "a=imageattr" "q=" value is not important. It
SHOULD be set to 1.0.</t> <bcp14>SHOULD</bcp14> be set to "1.0".</t>
<t>The "a=imageattr" field is payload type specific. When all <t>The "a=imageattr" field is payload type specific. When all
video codecs supported have the same capabilities, use of a video codecs supported have the same capabilities, use of a
single attribute, with the wildcard payload type (*), is single attribute, with the wildcard payload type (*), is
RECOMMENDED. However, when the supported video codecs have <bcp14>RECOMMENDED</bcp14>. However, when the supported video codecs h
different limitations, specific "a=imageattr" attributes MUST ave
different limitations, specific "a=imageattr" attributes <bcp14>MUST</
bcp14>
be inserted for each payload type.</t> be inserted for each payload type.</t>
<t>As an example, consider a system with a multiformat video <t>As an example, consider a system with a multiformat video
decoder, which is capable of decoding any resolution from decoder, which is capable of decoding any resolution from
48x48 to 720p, In this case, the implementation would 48x48 to 720p. In this case, the implementation would
generate this attribute:</t> generate this attribute:</t>
<t>a=imageattr:* recv [x=[48:1280],y=[48:720],q=1.0]</t> <t>a=imageattr:* recv [x=[48:1280],y=[48:720],q=1.0]</t>
<t>This declaration indicates that the receiver is capable of <t>This declaration indicates that the receiver is capable of
decoding any image resolution from 48x48 up to 1280x720 decoding any image resolution from 48x48 up to 1280x720
pixels.</t> pixels.</t>
</section> </section>
<section anchor="sec.interpreting-imageattr" <section anchor="sec.interpreting-imageattr" numbered="true" toc="defaul
title="Interpreting imageattr Attributes"> t">
<name>Interpreting imageattr Attributes</name>
<t> <t>
<xref target="RFC6236" /> defines "a=imageattr" to be an <xref target="RFC6236" format="default"/> defines "a=imageattr" to be an
advisory field. This means that it does not absolutely advisory field. This means that it does not absolutely
constrain the video formats that the sender can use, but constrain the video formats that the sender can use but
gives an indication of the preferred values.</t> gives an indication of the preferred values.</t>
<t>This specification prescribes behavior that is more specific. When
<t>This specification prescribes more specific behavior. When
a MediaStreamTrack, which is producing video of a certain a MediaStreamTrack, which is producing video of a certain
resolution (the "track resolution"), is attached to a resolution (the "track resolution"), is attached to an
RtpSender, which is encoding the track video at the same or RtpSender, which is encoding the track video at the same or
lower resolution(s) (the "encoder resolutions"), and a remote lower resolution(s) (the "encoder resolutions"), and a remote
description is applied that references the sender and description is applied that references the sender and
contains valid "a=imageattr recv" attributes, it MUST follow contains valid "a=imageattr recv" attributes, it <bcp14>MUST</bcp14> f
the rules below to ensure the sender does not transmit a ollow
the rules below to ensure that the sender does not transmit a
resolution that would exceed the size criteria specified in resolution that would exceed the size criteria specified in
the attributes. These rules MUST be followed as long as the the attributes. These rules <bcp14>MUST</bcp14> be followed as long as the
attributes remain present in the remote description, attributes remain present in the remote description,
including cases in which the track changes its resolution, or including cases in which the track changes its resolution or
is replaced with a different track.</t> is replaced with a different track.</t>
<t>Depending on how the RtpSender is configured, it may be <t>Depending on how the RtpSender is configured, it may be
producing a single encoding at a certain resolution, or, if producing a single encoding at a certain resolution or, if
simulcast simulcast
<xref target="sec.simulcast" /> has been negotiated, multiple (<xref target="sec.simulcast" format="default"/>) has been negotiated, multiple
encodings, each at their own specific resolution. In encodings, each at their own specific resolution. In
addition, depending on the configuration, each encoding may addition, depending on the configuration, each encoding may
have the flexibility to reduce resolution when needed, or may have the flexibility to reduce resolution when needed or may
be locked to a specific output resolution.</t> be locked to a specific output resolution.</t>
<t>For each encoding being produced by the RtpSender, the set <t>For each encoding being produced by the RtpSender, the set
of "a=imageattr recv" attributes in the corresponding m= of "a=imageattr recv" attributes in the corresponding "m="
section of the remote description is processed to determine section of the remote description is processed to determine
what should be transmitted. Only attributes that reference what should be transmitted. Only attributes that reference
the media format selected for the encoding are considered; the media format selected for the encoding are considered;
each such attribute is evaluated individually, starting with each such attribute is evaluated individually, starting with
the attribute with the highest "q=" value. If multiple the attribute with the highest "q=" value. If multiple
attributes have the same "q=" value, they are evaluated in attributes have the same "q=" value, they are evaluated in
the order they appear in their containing m= section. Note the order they appear in their containing "m=" section. Note
that while JSEP endpoints will include at most one that while JSEP endpoints will include at most one
"a=imageattr recv" attribute per media format, JSEP endpoints "a=imageattr recv" attribute per media format, JSEP endpoints
may receive session descriptions from non-JSEP endpoints with may receive session descriptions from non-JSEP endpoints with
m= sections that contain multiple such attributes.</t> "m=" sections that contain multiple such attributes.</t>
<t>For each "a=imageattr recv" attribute, the following rules <t>For each "a=imageattr recv" attribute, the following rules
are applied. If this processing is successful, the encoding are applied. If this processing is successful, the encoding
is transmitted accordingly, and no further attributes are is transmitted accordingly, and no further attributes are
considered for that encoding. Otherwise, the next attribute considered for that encoding. Otherwise, the next attribute
is evaluated, in the aforementioned order. If none of the is evaluated, in the aforementioned order. If none of the
supplied attributes can be processed successfully, the supplied attributes can be processed successfully, the
encoding MUST NOT be transmitted, and an error SHOULD be encoding <bcp14>MUST NOT</bcp14> be transmitted, and an error <bcp14>S HOULD</bcp14> be
raised to the application. raised to the application.
<list style="symbols"> </t>
<ul spacing="normal">
<t>The limits from the attribute are compared to the <li>The limits from the attribute are compared to the
encoder resolution. Only the specific limits mentioned encoder resolution. Only the specific limits mentioned
below are considered; any other values, such as picture below are considered; any other values, such as picture
aspect ratio, MUST be ignored. When considering a aspect ratio, <bcp14>MUST</bcp14> be ignored. When considering a
MediaStreamTrack that is producing rotated video, the MediaStreamTrack that is producing rotated video, the
unrotated resolution MUST be used for the checks. This is unrotated resolution <bcp14>MUST</bcp14> be used for the checks. Thi s is
required regardless of whether the receiver supports required regardless of whether the receiver supports
performing receive-side rotation (e.g., through CVO performing receive-side rotation (e.g., through Coordination of
<xref target="TS26.114" />), as it significantly simplifies Video Orientation (CVO)
the matching logic.</t> <xref target="TS26.114" format="default"/>), as it significantly sim
plifies
<t>If the attribute includes a "sar=" (sample aspect ratio) the matching logic.</li>
value set to something other than "1.0", indicating the <li>If the attribute includes a "sar=" (sample aspect ratio)
value set to something other than "1.0", indicating that the
receiver wants to receive non-square pixels, this cannot be receiver wants to receive non-square pixels, this cannot be
satisfied and the attribute MUST NOT be used.</t> satisfied and the attribute <bcp14>MUST NOT</bcp14> be used.</li>
<li>If the encoder resolution exceeds the maximum size
<t>If the encoder resolution exceeds the maximum size permitted by the attribute and the encoder is allowed to
permitted by the attribute, and the encoder is allowed to adjust its resolution, the encoder <bcp14>SHOULD</bcp14> apply downs
adjust its resolution, the encoder SHOULD apply downscaling caling
in order to satisfy the limits. Downscaling MUST NOT change in order to satisfy the limits. Downscaling <bcp14>MUST NOT</bcp14>
change
the picture aspect ratio of the encoding, ignoring any the picture aspect ratio of the encoding, ignoring any
trivial differences due to rounding. For example, if the trivial differences due to rounding. For example, if the
encoder resolution is 1280x720, and the attribute specified encoder resolution is 1280x720 and the attribute specified
a maximum of 640x480, the expected output resolution would a maximum of 640x480, the expected output resolution would
be 640x360. If downscaling cannot be applied, the attribute be 640x360. If downscaling cannot be applied, the attribute
MUST NOT be used.</t> <bcp14>MUST NOT</bcp14> be used.</li>
<li>If the encoder resolution is less than the minimum size
<t>If the encoder resolution is less than the minimum size permitted by the attribute, the attribute <bcp14>MUST NOT</bcp14> be
permitted by the attribute, the attribute MUST NOT be used; used;
the encoder MUST NOT apply upscaling. JSEP implementations the encoder <bcp14>MUST NOT</bcp14> apply upscaling. JSEP implementa
SHOULD avoid this situation by allowing receipt of tions
<bcp14>SHOULD</bcp14> avoid this situation by allowing receipt of
arbitrarily small resolutions, perhaps via fallback to a arbitrarily small resolutions, perhaps via fallback to a
software decoder.</t> software decoder.</li>
<li>If the encoder resolution is within the maximum and
<t>If the encoder resolution is within the maximum and minimum sizes, no action is needed.</li>
minimum sizes, no action is needed.</t> </ul>
</list></t>
</section> </section>
</section> </section>
<section title="Simulcast" anchor="sec.simulcast"> <section anchor="sec.simulcast" numbered="true" toc="default">
<name>Simulcast</name>
<t>JSEP supports simulcast transmission of a MediaStreamTrack, <t>JSEP supports simulcast transmission of a MediaStreamTrack,
where multiple encodings of the source media can be transmitted where multiple encodings of the source media can be transmitted
within the context of a single m= section. The current JSEP API within the context of a single "m=" section. The current JSEP API
is designed to allow applications to send simulcasted media but is designed to allow applications to send simulcasted media but
only to receive a single encoding. This allows for multi-user only to receive a single encoding. This allows for multi-user
scenarios where each sending client sends multiple encodings to scenarios where each sending client sends multiple encodings to
a server, which then, for each receiving client, chooses the a server, which then, for each receiving client, chooses the
appropriate encoding to forward.</t> appropriate encoding to forward.</t>
<t>Applications request support for simulcast by configuring <t>Applications request support for simulcast by configuring
multiple encodings on an RtpSender. Upon generation of an offer multiple encodings on an RtpSender. Upon generation of an offer
or answer, these encodings are indicated via SDP markings on or answer, these encodings are indicated via SDP markings on
the corresponding m= section, as described below. Receivers the corresponding "m=" section, as described below. Receivers
that understand simulcast and are willing to receive it will that understand simulcast and are willing to receive it will
also include SDP markings to indicate their support, and JSEP also include SDP markings to indicate their support, and JSEP
endpoints will use these markings to determine whether endpoints will use these markings to determine whether
simulcast is permitted for a given RtpSender. If simulcast simulcast is permitted for a given RtpSender. If simulcast
support is not negotiated, the RtpSender will only use the support is not negotiated, the RtpSender will only use the
first configured encoding.</t> first configured encoding.</t>
<t>Note that the exact simulcast parameters are up to the <t>Note that the exact simulcast parameters are up to the
sending application. While the aforementioned SDP markings are sending application. While the aforementioned SDP markings are
provided to ensure the remote side can receive and demux provided to ensure that the remote side can receive and demux
multiple simulcast encodings, the specific resolutions and multiple simulcast encodings, the specific resolutions and
bitrates to be used for each encoding are purely a send-side bitrates to be used for each encoding are purely a send-side
decision in JSEP.</t> decision in JSEP.</t>
<t>JSEP currently does not provide a mechanism to configure <t>JSEP currently does not provide a mechanism to configure
receipt of simulcast. This means that if simulcast is offered receipt of simulcast. This means that if simulcast is offered
by the remote endpoint, the answer generated by a JSEP endpoint by the remote endpoint, the answer generated by a JSEP endpoint
will not indicate support for receipt of simulcast, and as such will not indicate support for receipt of simulcast, and as such
the remote endpoint will only send a single encoding per m= the remote endpoint will only send a single encoding per "m="
section.</t> section.</t>
<t>In addition, JSEP does not provide a mechanism to handle an <t>In addition, JSEP does not provide a mechanism to handle an
incoming offer requesting simulcast from the JSEP endpoint. incoming offer requesting simulcast from the JSEP endpoint.
This means that setting up simulcast in the case where the JSEP This means that setting up simulcast in the case where the JSEP
endpoint receives the initial offer requires out-of-band endpoint receives the initial offer requires out-of-band
signaling or SDP inspection. However, in the case where the signaling or SDP inspection. However, in the case where the
JSEP endpoint sets up simulcast in its in initial offer, any JSEP endpoint sets up simulcast in its initial offer, any
established simulcast streams will continue to work upon established simulcast streams will continue to work upon
receipt of an incoming re-offer. Future versions of this receipt of an incoming re-offer. Future versions of this
specification may add additional APIs to handle the incoming specification may add additional APIs to handle the incoming
initial offer scenario.</t> initial offer scenario.</t>
<t>When using JSEP to transmit multiple encodings from an
<t>When using JSEP to transmit multiple encodings from a
RtpSender, the techniques from RtpSender, the techniques from
<xref target="I-D.ietf-mmusic-sdp-simulcast" /> and <xref target="RFC8853" format="default"/> and
<xref target="I-D.ietf-mmusic-rid" /> are used. Specifically, <xref target="RFC8851" format="default"/> are used. Specifically,
when multiple encodings have been configured for a RtpSender, when multiple encodings have been configured for an RtpSender,
the m= section for the RtpSender will include an "a=simulcast" the "m=" section for the RtpSender will include an "a=simulcast"
attribute, as defined in attribute, as defined in
<xref target="I-D.ietf-mmusic-sdp-simulcast" />, Section 6.2, <xref target="RFC8853" sectionFormat="comma" section="6.2"/>,
with a "send" simulcast stream description that lists each with a "send" simulcast stream description that lists each
desired encoding, and no "recv" simulcast stream description. desired encoding, and no "recv" simulcast stream description.
The m= section will also include an "a=rid" attribute for each
<!-- [rfced] Sections 3.7 and 5.2.1: We do not see "a=simulcast" or
"send" mentioned anywhere in Section 6.2 of
RFC 8853 [I-D.ietf-mmusic-sdp-simulcast]. Please confirm that these citations
are correct and will be clear to readers.
Original:
Specifically, when multiple
encodings have been configured for a RtpSender, the m= section for
the RtpSender will include an "a=simulcast" attribute, as defined in
[I-D.ietf-mmusic-sdp-simulcast], Section 6.2, with a "send" simulcast
stream description that lists each desired encoding, and no "recv"
simulcast stream description.
...
o If the RtpTransceiver has a sendrecv or sendonly direction and
more than one "a=rid" line has been generated, an "a=simulcast"
line, with direction "send", as defined in
[I-D.ietf-mmusic-sdp-simulcast], Section 6.2. -->
The "m=" section will also include an "a=rid" attribute for each
encoding, as specified in encoding, as specified in
<xref target="I-D.ietf-mmusic-rid" />, Section 4; the use of <xref target="RFC8851" sectionFormat="comma" section="4"/>; the use of
RID identifiers allows the individual encodings to be Restriction Identifiers (RIDs) allows the individual encodings to be
disambiguated even though they are all part of the same m= disambiguated even though they are all part of the same "m="
section.</t> section.</t>
</section> </section>
<section title="Interactions With Forking" <section anchor="sec.interactions-with-forking" numbered="true" toc="defau
anchor="sec.interactions-with-forking"> lt">
<name>Interactions with Forking</name>
<t>Some call signaling systems allow various types of forking <t>Some call signaling systems allow various types of forking
where an SDP Offer may be provided to more than one device. For where an SDP Offer may be provided to more than one device. For
example, SIP example, SIP
<xref target="RFC3261"></xref> defines both a "Parallel Search" <xref target="RFC3261" format="default"/> defines both a "parallel searc
and "Sequential Search". Although these are primarily signaling h"
level issues that are outside the scope of JSEP, they do have and "sequential search". Although these are primarily signaling-level is
sues that are outside the scope of JSEP, they do have
some impact on the configuration of the media plane that is some impact on the configuration of the media plane that is
relevant. When forking happens at the signaling layer, the relevant. When forking happens at the signaling layer, the
JavaScript application responsible for the signaling needs to JavaScript application responsible for the signaling needs to
make the decisions about what media should be sent or received make the decisions about what media should be sent or received
at any point of time, as well as which remote endpoint it at any point in time, as well as which remote endpoint it
should communicate with; JSEP is used to make sure the media should communicate with; JSEP is used to make sure the media
engine can make the RTP and media perform as required by the engine can make the RTP and media perform as required by the
application. The basic operations that the applications can application. The basic operations that the applications can
have the media engine do are: have the media engine do are as follows:
<list style="symbols"> </t>
<ul spacing="normal">
<t>Start exchanging media with a given remote peer, but keep <li>Start exchanging media with a given remote peer, but keep
all the resources reserved in the offer.</t> all the resources reserved in the offer.</li>
<li>Start exchanging media with a given remote peer, and free
<t>Start exchanging media with a given remote peer, and free any resources in the offer that are not being used.</li>
any resources in the offer that are not being used.</t> </ul>
</list></t> <section anchor="sec.sequential-forking" numbered="true" toc="default">
<section title="Sequential Forking" <name>Sequential Forking</name>
anchor="sec.sequential-forking">
<t>Sequential forking involves a call being dispatched to <t>Sequential forking involves a call being dispatched to
multiple remote callees, where each callee can accept the multiple remote callees, where each callee can accept the
call, but only one active session ever exists at a time; no call, but only one active session ever exists at a time; no
mixing of received media is performed.</t> mixing of received media is performed.</t>
<t>JSEP handles sequential forking well, allowing the <t>JSEP handles sequential forking well, allowing the
application to easily control the policy for selecting the application to easily control the policy for selecting the
desired remote endpoint. When an answer arrives from one of desired remote endpoint. When an answer arrives from one of
the callees, the application can choose to apply it either as the callees, the application can choose to apply it as either
a provisional answer, leaving open the possibility of using a (1)&nbsp;a provisional answer, leaving open the possibility of using a
different answer in the future, or apply it as a final different answer in the future or (2)&nbsp;a final
answer, ending the setup flow.</t> answer, ending the setup flow.</t>
<t>In a "first-one-wins" situation, the first answer will be <t>In a "first-one-wins" situation, the first answer will be
applied as a final answer, and the application will reject applied as a final answer, and the application will reject
any subsequent answers. In SIP parlance, this would be ACK + any subsequent answers. In SIP parlance, this would be ACK +
BYE.</t> BYE.</t>
<t>In a "last-one-wins" situation, all answers would be <t>In a "last-one-wins" situation, all answers would be
applied as provisional answers, and any previous call leg applied as provisional answers, and any previous call leg
will be terminated. At some point, the application will end will be terminated. At some point, the application will end
the setup process, perhaps with a timer; at this point, the the setup process, perhaps with a timer; at this point, the
application could reapply the pending remote description as a application could reapply the pending remote description as a
final answer.</t> final answer.</t>
</section> </section>
<section title="Parallel Forking" <section anchor="sec.parallel-forking" numbered="true" toc="default">
anchor="sec.parallel-forking"> <name>Parallel Forking</name>
<t>Parallel forking involves a call being dispatched to <t>Parallel forking involves a call being dispatched to
multiple remote callees, where each callee can accept the multiple remote callees, where each callee can accept the
call, and multiple simultaneous active signaling sessions can call and multiple simultaneous active signaling sessions can
be established as a result. If multiple callees send media at be established as a result. If multiple callees send media at
the same time, the possibilities for handling this are the same time, the possibilities for handling this are
described in described in
<xref target="RFC3960"></xref>, Section 3.1. Most SIP devices <xref target="RFC3960" sectionFormat="comma" section="3.1"/>. Most SIP devices
today only support exchanging media with a single device at a today only support exchanging media with a single device at a
time, and do not try to mix multiple early media audio time and do not try to mix multiple early media audio
sources, as that could result in a confusing situation. For sources, as that could result in a confusing situation. For
example, consider having a European ringback tone mixed example, consider having a European ringback tone mixed
together with the North American ringback tone - the together with the North American ringback tone -- the
resulting sound would not be like either tone, and would resulting sound would not be like either tone and would
confuse the user. If the signaling application wishes to only confuse the user. If the signaling application wishes to only
exchange media with one of the remote endpoints at a time, exchange media with one of the remote endpoints at a time,
then from a media engine point of view, this is exactly like then from a media engine point of view, this is exactly like
the sequential forking case.</t> the sequential forking case.</t>
<t>In the parallel forking case where the JavaScript <t>In the parallel forking case where the JavaScript
application wishes to simultaneously exchange media with application wishes to simultaneously exchange media with
multiple peers, the flow is slightly more complex, but the multiple peers, the flow is slightly more complex, but the
JavaScript application can follow the strategy that JavaScript application can follow the strategy that
<xref target="RFC3960"></xref> describes using UPDATE. The <xref target="RFC3960" format="default"/> describes, using UPDATE. The
UPDATE approach allows the signaling to set up a separate UPDATE approach allows the signaling to set up a separate
media flow for each peer that it wishes to exchange media media flow for each peer that it wishes to exchange media
with. In JSEP, this offer used in the UPDATE would be formed with. In JSEP, this offer used in the UPDATE would be formed
by simply creating a new PeerConnection (see by simply creating a new PeerConnection (see
<xref target="sec.peerconnection" />) and making sure that <xref target="sec.peerconnection" format="default"/>) and making sure that
the same local media streams have been added into this new the same local media streams have been added into this new
PeerConnection. Then the new PeerConnection object would PeerConnection. Then the new PeerConnection object would
produce a SDP offer that could be used by the signaling to produce an SDP offer that could be used by the signaling to
perform the UPDATE strategy discussed in perform the UPDATE strategy discussed in
<xref target="RFC3960"></xref>.</t> <xref target="RFC3960" format="default"/>.</t>
<t>As a result of sharing the media streams, the application <t>As a result of sharing the media streams, the application
will end up with N parallel PeerConnection sessions, each will end up with N parallel PeerConnection sessions, each
with a local and remote description and their own local and with a local and remote description and their own local and
remote addresses. The media flow from these sessions can be remote addresses. The media flow from these sessions can be
managed using setDirection (see managed using setDirection (see
<xref target="sec.transceiver-set-direction" />), or the <xref target="sec.transceiver-set-direction" format="default"/>), or t he
application can choose to play out the media from all application can choose to play out the media from all
sessions mixed together. Of course, if the application wants sessions mixed together. Of course, if the application wants
to only keep a single session, it can simply terminate the to only keep a single session, it can simply terminate the
sessions that it no longer needs.</t> sessions that it no longer needs.</t>
</section> </section>
</section> </section>
</section> </section>
<section title="Interface" anchor="sec.interface"> <section anchor="sec.interface" numbered="true" toc="default">
<name>Interface</name>
<t>This section details the basic operations that must be present <t>This section details the basic operations that must be present
to implement JSEP functionality. The actual API exposed in the to implement JSEP functionality. The actual API exposed in the
W3C API may have somewhat different syntax, but should map easily W3C API may have somewhat different syntax but should map easily
to these concepts.</t> to these concepts.
<section title="PeerConnection" anchor="sec.peerconnection">
<section title="Constructor" anchor="sec.pc-constructor"> <!-- [rfced] Sections 4 and 7: For ease of the reader, please let
us know if we may cite [W3C.webrtc] as a reminder regarding
"addTrack," "removeTrack," etc. (Section 4) as well as
"onicecandidate," "addIceCandidate," and "ontrack" (Section 7),
as follows.
Original:
The actual API exposed in the W3C API
may have somewhat different syntax, but should map easily to these
concepts.
...
More examples of SDP for WebRTC call flows, including examples with
IPv6 addresses, can be found in [I-D.ietf-rtcweb-sdp].
Suggested:
The actual API exposed in the W3C API [W3C.webrtc]
may have somewhat different syntax but should map easily to these
concepts.
...
More examples of SDP for WebRTC call flows, including examples with
IPv6 addresses, can be found in [SDP4WebRTC]. See [W3C.webrtc] for
information regarding "onicecandidate," "addIceCandidate," and
"ontrack". -->
</t>
<section anchor="sec.peerconnection" numbered="true" toc="default">
<name>PeerConnection</name>
<section anchor="sec.pc-constructor" numbered="true" toc="default">
<name>Constructor</name>
<t>The PeerConnection constructor allows the application to <t>The PeerConnection constructor allows the application to
specify global parameters for the media session, such as the specify global parameters for the media session, such as the
STUN/TURN servers and credentials to use when gathering STUN/TURN servers and credentials to use when gathering
candidates, as well as the initial ICE candidate policy and candidates, as well as the initial ICE candidate policy and
pool size, and also the bundle policy to use.</t> pool size, and also the bundle policy to use.</t>
<t>If an ICE candidate policy is specified, it functions as <t>If an ICE candidate policy is specified, it functions as
described in described in
<xref target="sec.ice-candidate-policy" />, causing the JSEP <xref target="sec.ice-candidate-policy" format="default"/>, causing th e JSEP
implementation to only surface the permitted candidates implementation to only surface the permitted candidates
(including any implementation-internal filtering) to the (including any implementation-internal filtering) to the
application, and only use those candidates for connectivity application and only use those candidates for connectivity
checks. The set of available policies is as follows: checks. The set of available policies is as follows:
<list style="hanging"> </t>
<t hangText="all:">All candidates permitted by <dl newline="false" spacing="normal">
implementation policy will be gathered and used.</t> <dt>all:</dt>
<dd>All candidates permitted by
<t></t> implementation policy will be gathered and used.</dd>
<t hangText="relay:">All candidates except relay candidates <dt>relay:</dt>
<dd>All candidates except relay candidates
will be filtered out. This obfuscates the location will be filtered out. This obfuscates the location
information that might be ascertained by the remote peer information that might be ascertained by the remote peer
from the received candidates. Depending on how the from the received candidates. Depending on how the
application deploys and chooses relay servers, this could application deploys and chooses relay servers, this could
obfuscate location to a metro or possibly even global obfuscate location to a metro or possibly even global
level.</t> level.</dd>
</list></t> </dl>
<t>The default ICE candidate policy <bcp14>MUST</bcp14> be set to "all
<t>The default ICE candidate policy MUST be set to "all" as ", as
this is generally the desired policy, and also typically this is generally the desired policy and also typically
reduces use of application TURN server resources reduces the use of application TURN server resources
significantly.</t> significantly.</t>
<t>If a size is specified for the ICE candidate pool, this <t>If a size is specified for the ICE candidate pool, this
indicates the number of ICE components to pre-gather indicates the number of ICE components to pre-gather
candidates for. Because pre-gathering results in utilizing candidates for. Because pre&nbhy;gathering results in utilizing
STUN/TURN server resources for potentially long periods of STUN/TURN server resources for potentially long periods of
time, this must only occur upon application request, and time, this must only occur upon application request, and
therefore the default candidate pool size MUST be zero.</t> therefore the default candidate pool size <bcp14>MUST</bcp14> be zero.
</t>
<t>The application can specify its preferred policy regarding <t>The application can specify its preferred policy regarding
use of bundle, the multiplexing mechanism defined in use of bundle, the multiplexing mechanism defined in
<xref target="I-D.ietf-mmusic-sdp-bundle-negotiation"> <xref target="RFC8843" format="default">
</xref>. Regardless of policy, the application will always </xref>. Regardless of policy, the application will always
try to negotiate bundle onto a single transport, and will try to negotiate bundle onto a single transport and will
offer a single bundle group across all m= sections; use of offer a single bundle group across all "m=" sections; use of
this single transport is contingent upon the answerer this single transport is contingent upon the answerer
accepting bundle. However, by specifying a policy from the accepting bundle. However, by specifying a policy from the
list below, the application can control exactly how list below, the application can control exactly how
aggressively it will try to bundle media streams together, aggressively it will try to bundle media streams together,
which affects how it will interoperate with a which affects how it will interoperate with a
non-bundle-aware endpoint. When negotiating with a non-bundle-aware endpoint. When negotiating with a
non-bundle-aware endpoint, only the streams not marked as non-bundle-aware endpoint, only the streams not marked as
bundle-only streams will be established.</t> bundle-only streams will be established.</t>
<t>The set of available policies is as follows: <t>The set of available policies is as follows:
<list style="hanging"> </t>
<t hangText="balanced:">The first m= section of each type <dl newline="false" spacing="normal">
<dt>balanced:</dt>
<dd>The first "m=" section of each type
(audio, video, or application) will contain transport (audio, video, or application) will contain transport
parameters, which will allow an answerer to unbundle that parameters, which will allow an answerer to unbundle that
section. The second and any subsequent m= section of each section. The second and any subsequent "m=" sections of each
type will be marked bundle-only. The result is that if type will be marked bundle-only. The result is that if
there are N distinct media types, then candidates will be there are N distinct media types, then candidates will be
gathered for for N media streams. This policy balances gathered for N media streams. This policy balances
desire to multiplex with the need to ensure basic audio and desire to multiplex with the need to ensure that basic audio and
video can still be negotiated in legacy cases. When acting video can still be negotiated in legacy cases. When acting
as answerer, if there is no bundle group in the offer, the as answerer, if there is no bundle group in the offer, the
implementation will reject all but the first m= section of implementation will reject all but the first "m=" section of
each type.</t> each type.</dd>
<dt>max-compat:</dt>
<t></t> <dd>All "m=" sections will contain
<t hangText="max-compat:">All m= sections will contain
transport parameters; none will be marked as bundle-only. transport parameters; none will be marked as bundle-only.
This policy will allow all streams to be received by This policy will allow all streams to be received by
non-bundle-aware endpoints, but require separate candidates non-bundle-aware endpoints but will require separate candidates
to be gathered for each media stream.</t> to be gathered for each media stream.</dd>
<dt>max-bundle:</dt>
<t></t> <dd>Only the first "m=" section will
<t hangText="max-bundle:">Only the first m= section will
contain transport parameters; all streams other than the contain transport parameters; all streams other than the
first will be marked as bundle-only. This policy aims to first will be marked as bundle-only. This policy aims to
minimize candidate gathering and maximize multiplexing, at minimize candidate gathering and maximize multiplexing, at
the cost of less compatibility with legacy endpoints. When the cost of less compatibility with legacy endpoints. When
acting as answerer, the implementation will reject any m= acting as answerer, the implementation will reject any "m="
sections other than the first m= section, unless they are sections other than the first "m=" section, unless they are
in the same bundle group as that m= section.</t> in the same bundle group as that "m=" section.</dd>
</list></t> </dl>
<t>As it provides the best trade-off between performance and
<t>As it provides the best tradeoff between performance and
compatibility with legacy endpoints, the default bundle compatibility with legacy endpoints, the default bundle
policy MUST be set to "balanced".</t> policy <bcp14>MUST</bcp14> be set to "balanced".</t>
<t>The application can specify its preferred policy regarding <t>The application can specify its preferred policy regarding
use of RTP/RTCP multiplexing use of RTP/RTCP multiplexing
<xref target="RFC5761"></xref> using one of the following <xref target="RFC5761" format="default"/> using one of the following
policies: policies:
<list style="hanging"> </t>
<t hangText="negotiate:">The JSEP implementation will <dl newline="false" spacing="normal">
<dt>negotiate:</dt>
<dd>The JSEP implementation will
gather both RTP and RTCP candidates but also will offer gather both RTP and RTCP candidates but also will offer
"a=rtcp-mux", thus allowing for compatibility with either "a=rtcp-mux", thus allowing for compatibility with either
multiplexing or non-multiplexing endpoints.</t> multiplexing or non-multiplexing endpoints.</dd>
<t hangText="require:">The JSEP implementation will only <dt>require:</dt>
<dd>The JSEP implementation will only
gather RTP candidates and will insert an "a=rtcp-mux-only" gather RTP candidates and will insert an "a=rtcp-mux-only"
indication into any new m= sections in offers it generates. indication into any new "m=" sections in offers it generates.
This halves the number of candidates that the offerer needs This halves the number of candidates that the offerer needs
to gather. Applying a description with an m= section that to gather. Applying a description with an "m=" section that
does not contain an "a=rtcp-mux" attribute will cause an does not contain an "a=rtcp-mux" attribute will cause an
error to be returned.</t> error to be returned.</dd>
</list></t> </dl>
<t>The default multiplexing policy <bcp14>MUST</bcp14> be set to "requ
<t>The default multiplexing policy MUST be set to "require". ire".
Implementations MAY choose to reject attempts by the Implementations <bcp14>MAY</bcp14> choose to reject attempts by the
application to set the multiplexing policy to application to set the multiplexing policy to
"negotiate".</t> "negotiate".</t>
</section> </section>
<section title="addTrack" anchor="sec.addTrack"> <section anchor="sec.addTrack" numbered="true" toc="default">
<name>addTrack</name>
<t>The addTrack method adds a MediaStreamTrack to the <t>The addTrack method adds a MediaStreamTrack to the
PeerConnection, using the MediaStream argument to associate PeerConnection, using the MediaStream argument to associate
the track with other tracks in the same MediaStream, so that the track with other tracks in the same MediaStream, so that
they can be added to the same "LS" group when creating an they can be added to the same "LS" (Lip Synchronization) group when cr eating an
offer or answer. Adding tracks to the same "LS" group offer or answer. Adding tracks to the same "LS" group
indicates that the playback of these tracks should be indicates that the playback of these tracks should be
synchronized for proper lip sync, as described in synchronized for proper lip sync, as described in
<xref target="RFC5888"></xref>, Section 7. addTrack attempts <xref target="RFC5888" sectionFormat="comma" section="7"/>. &nbsp;addT
to minimize the number of transceivers as follows: If the rack attempts
PeerConnection is in the "have-remote-offer" state, the track to minimize the number of transceivers as follows: if the
PeerConnection is in the "have&nbhy;remote-offer" state, the track
will be attached to the first compatible transceiver that was will be attached to the first compatible transceiver that was
created by the most recent call to setRemoteDescription() and created by the most recent call to setRemoteDescription() and
does not have a local track. Otherwise, a new transceiver does not have a local track. Otherwise, a new transceiver
will be created, as described in will be created, as described in
<xref target="sec.addTransceiver" />.</t> <xref target="sec.addTransceiver" format="default"/>.</t>
</section> </section>
<section title="removeTrack" anchor="sec.removeTrack"> <section anchor="sec.removeTrack" numbered="true" toc="default">
<name>removeTrack</name>
<t>The removeTrack method removes a MediaStreamTrack from the <t>The removeTrack method removes a MediaStreamTrack from the
PeerConnection, using the RtpSender argument to indicate PeerConnection, using the RtpSender argument to indicate
which sender should have its track removed. The sender's which sender should have its track removed. The sender's
track is cleared, and the sender stops sending. Future calls track is cleared, and the sender stops sending. Future calls
to createOffer will mark the m= section associated with the to createOffer will mark the "m=" section associated with the
sender as recvonly (if transceiver.direction is sendrecv) or sender as recvonly (if transceiver.direction is sendrecv) or
as inactive (if transceiver.direction is sendonly).</t> as inactive (if transceiver.direction is sendonly).</t>
</section> </section>
<section title="addTransceiver" anchor="sec.addTransceiver"> <section anchor="sec.addTransceiver" numbered="true" toc="default">
<name>addTransceiver</name>
<t>The addTransceiver method adds a new RtpTransceiver to the <t>The addTransceiver method adds a new RtpTransceiver to the
PeerConnection. If a MediaStreamTrack argument is provided, PeerConnection. If a MediaStreamTrack argument is provided,
then the transceiver will be configured with that media type then the transceiver will be configured with that media type
and the track will be attached to the transceiver. Otherwise, and the track will be attached to the transceiver. Otherwise,
the application MUST explicitly specify the type; this mode the application <bcp14>MUST</bcp14> explicitly specify the type; this mode
is useful for creating recvonly transceivers as well as for is useful for creating recvonly transceivers as well as for
creating transceivers to which a track can be attached at creating transceivers to which a track can be attached at
some later point.</t> some later point.</t>
<t>At the time of creation, the application can also specify <t>At the time of creation, the application can also specify
a transceiver direction attribute, a set of MediaStreams a transceiver direction attribute, a set of MediaStreams
which the transceiver is associated with (allowing LS group that the transceiver is associated with (allowing "LS" group
assignments), and a set of encodings for the media (used for assignments), and a set of encodings for the media (used for
simulcast as described in simulcast as described in
<xref target="sec.simulcast" />).</t> <xref target="sec.simulcast" format="default"/>).</t>
</section> </section>
<section title="createDataChannel" <section anchor="sec.createDataChannel" numbered="true" toc="default">
anchor="sec.createDataChannel"> <name>createDataChannel</name>
<t>The createDataChannel method creates a new data channel <t>The createDataChannel method creates a new data channel
and attaches it to the PeerConnection. If no data channel and attaches it to the PeerConnection. If no data channel
currently exists for this PeerConnection, then a new currently exists for this PeerConnection, then a new
offer/answer exchange is required. All data channels on a offer/answer exchange is required. All data channels on a
given PeerConnection share the same SCTP/DTLS association and given PeerConnection share the same SCTP/DTLS association ("SCTP" stan
therefore the same m= section, so subsequent creation of data ds
for "Stream Control Transmission Protocol") and
therefore the same "m=" section, so subsequent creation of data
channels does not have any impact on the JSEP state.</t> channels does not have any impact on the JSEP state.</t>
<t>The createDataChannel method also includes a number of <t>The createDataChannel method also includes a number of
arguments which are used by the PeerConnection (e.g., arguments that are used by the PeerConnection (e.g.,
maxPacketLifetime) but are not reflected in the SDP and do maxPacketLifetime) but are not reflected in the SDP and do
not affect the JSEP state.</t> not affect the JSEP state.</t>
</section> </section>
<section title="createOffer" anchor="sec.createoffer"> <section anchor="sec.createoffer" numbered="true" toc="default">
<name>createOffer</name>
<t>The createOffer method generates a blob of SDP that <t>The createOffer method generates a blob of SDP that
contains a contains an offer per <xref target="RFC3264" format="default"/> with t
<xref target="RFC3264"></xref> offer with the supported he supported
configurations for the session, including descriptions of the configurations for the session, including descriptions of the
media added to this PeerConnection, the codec/RTP/RTCP media added to this PeerConnection, the codec/RTP/RTCP
options supported by this implementation, and any candidates options supported by this implementation, and any candidates
that have been gathered by the ICE agent. An options that have been gathered by the ICE agent. An options
parameter may be supplied to provide additional control over parameter may be supplied to provide additional control over
the generated offer. This options parameter allows an the generated offer. This options parameter allows an
application to trigger an ICE restart, for the purpose of application to trigger an ICE restart, for the purpose of
reestablishing connectivity.</t> reestablishing connectivity.</t>
<t>In the initial offer, the generated SDP will contain all <t>In the initial offer, the generated SDP will contain all
desired functionality for the session (functionality that is desired functionality for the session (functionality that is
supported but not desired by default may be omitted); for supported but not desired by default may be omitted); for
each SDP line, the generation of the SDP will follow the each SDP line, the generation of the SDP will follow the
process defined for generating an initial offer from the process defined for generating an initial offer from the
document that specifies the given SDP line. The exact document that specifies the given SDP line. The exact
handling of initial offer generation is detailed in handling of initial offer generation is detailed in
<xref target="sec.initial-offers" /> below.</t> <xref target="sec.initial-offers" format="default"/> below.</t>
<t>In the event createOffer is called after the session is <t>In the event createOffer is called after the session is
established, createOffer will generate an offer to modify the established, createOffer will generate an offer to modify the
current session based on any changes that have been made to current session based on any changes that have been made to
the session, e.g., adding or stopping RtpTransceivers, or the session, e.g., adding or stopping RtpTransceivers, or
requesting an ICE restart. For each existing stream, the requesting an ICE restart. For each existing stream, the
generation of each SDP line must follow the process defined generation of each SDP line must follow the process defined
for generating an updated offer from the RFC that specifies for generating an updated offer from the RFC that specifies
the given SDP line. For each new stream, the generation of the given SDP line. For each new stream, the generation of
the SDP must follow the process of generating an initial the SDP must follow the process of generating an initial
offer, as mentioned above. If no changes have been made, or offer, as mentioned above. If no changes have been made, or
for SDP lines that are unaffected by the requested changes, for SDP lines that are unaffected by the requested changes,
the offer will only contain the parameters negotiated by the the offer will only contain the parameters negotiated by the
last offer-answer exchange. The exact handling of subsequent last offer/answer exchange. The exact handling of subsequent
offer generation is detailed in offer generation is detailed in
<xref target="sec.subsequent-offers" />. below.</t> <xref target="sec.subsequent-offers" format="default"/> below.</t>
<t>Session descriptions generated by createOffer must be <t>Session descriptions generated by createOffer must be
immediately usable by setLocalDescription; if a system has immediately usable by setLocalDescription; if a system has
limited resources (e.g. a finite number of decoders), limited resources (e.g., a finite number of decoders),
createOffer should return an offer that reflects the current createOffer should return an offer that reflects the current
state of the system, so that setLocalDescription will succeed state of the system, so that setLocalDescription will succeed
when it attempts to acquire those resources.</t> when it attempts to acquire those resources.</t>
<t>Calling this method may do things such as generating new <t>Calling this method may do things such as generating new
ICE credentials, but does not change the PeerConnection ICE credentials, but it does not change the PeerConnection
state, trigger candidate gathering, or cause media to start state, trigger candidate gathering, or cause media to start
or stop flowing. Specifically, the offer is not applied, and or stop flowing. Specifically, the offer is not applied, and
does not become the pending local description, until does not become the pending local description, until
setLocalDescription is called.</t> setLocalDescription is called.</t>
</section> </section>
<section title="createAnswer" anchor="sec.createanswer"> <section anchor="sec.createanswer" numbered="true" toc="default">
<name>createAnswer</name>
<t>The createAnswer method generates a blob of SDP that <t>The createAnswer method generates a blob of SDP that
contains a contains an SDP answer per <xref target="RFC3264" format="default"/> w
<xref target="RFC3264"></xref> SDP answer with the supported ith the supported
configuration for the session that is compatible with the configuration for the session that is compatible with the
parameters supplied in the most recent call to parameters supplied in the most recent call to
setRemoteDescription, which MUST have been called prior to setRemoteDescription, which <bcp14>MUST</bcp14> have been called prior to
calling createAnswer. Like createOffer, the returned blob calling createAnswer. Like createOffer, the returned blob
contains descriptions of the media added to this contains descriptions of the media added to this
PeerConnection, the codec/RTP/RTCP options negotiated for PeerConnection, the codec/RTP/RTCP options negotiated for
this session, and any candidates that have been gathered by this session, and any candidates that have been gathered by
the ICE agent. An options parameter may be supplied to the ICE agent. An options parameter may be supplied to
provide additional control over the generated answer.</t> provide additional control over the generated answer.</t>
<t>As an answer, the generated SDP will contain a specific <t>As an answer, the generated SDP will contain a specific
configuration that specifies how the media plane should be configuration that specifies how the media plane should be
established; for each SDP line, the generation of the SDP established; for each SDP line, the generation of the SDP
must follow the process defined for generating an answer from must follow the process defined for generating an answer from
the document that specifies the given SDP line. The exact the document that specifies the given SDP line. The exact
handling of answer generation is detailed in handling of answer generation is detailed in
<xref target="sec.generating-an-answer" />. below.</t> <xref target="sec.generating-an-answer" format="default"/> below.</t>
<t>Session descriptions generated by createAnswer must be <t>Session descriptions generated by createAnswer must be
immediately usable by setLocalDescription; like createOffer, immediately usable by setLocalDescription; like createOffer,
the returned description should reflect the current state of the returned description should reflect the current state of
the system.</t> the system.</t>
<t>Calling this method may do things such as generating new <t>Calling this method may do things such as generating new
ICE credentials, but does not change the PeerConnection ICE credentials, but it does not change the PeerConnection
state, trigger candidate gathering, or or cause a media state state, trigger candidate gathering, or cause a media state
change. Specifically, the answer is not applied, and does not change. Specifically, the answer is not applied, and does not
become the current local description, until become the current local description, until
setLocalDescription is called.</t> setLocalDescription is called.</t>
</section> </section>
<section title="SessionDescriptionType" <section anchor="sec.sessiondescriptiontype" numbered="true" toc="defaul
anchor="sec.sessiondescriptiontype"> t">
<name>SessionDescriptionType</name>
<t>Session description objects (RTCSessionDescription) may be <t>Session description objects (RTCSessionDescription) may be
of type "offer", "pranswer", "answer" or "rollback". These of type "offer", "pranswer", "answer", or "rollback". These
types provide information as to how the description parameter types provide information as to how the description parameter
should be parsed, and how the media state should be should be parsed and how the media state should be
changed.</t> changed.</t>
<t>"offer" indicates that a description should be parsed as <t>"offer" indicates that a description should be parsed as
an offer; said description may include many possible media an offer; said description may include many possible media
configurations. A description used as an "offer" may be configurations. A description used as an "offer" may be
applied anytime the PeerConnection is in a stable state, or applied any time the PeerConnection is in a stable state or
as an update to a previously supplied but unanswered applied as an update to a previously supplied but unanswered
"offer".</t> "offer".</t>
<t>"pranswer" indicates that a description should be parsed <t>"pranswer" indicates that a description should be parsed
as an answer, but not a final answer, and so should not as an answer, but not a final answer, and so should not
result in the freeing of allocated resources. It may result result in the freeing of allocated resources. It may result
in the start of media transmission, if the answer does not in the start of media transmission, if the answer does not
specify an inactive media direction. A description used as a specify an inactive media direction. A description used as a
"pranswer" may be applied as a response to an "offer", or an "pranswer" may be applied as a response to an "offer" or as an
update to a previously sent "pranswer".</t> update to a previously sent "pranswer".</t>
<t>"answer" indicates that a description should be parsed as <t>"answer" indicates that a description should be parsed as
an answer, the offer-answer exchange should be considered an answer, the offer/answer exchange should be considered
complete, and any resources (decoders, candidates) that are complete, and any resources (decoders, candidates) that are
no longer needed can be released. A description used as an no longer needed can be released. A description used as an
"answer" may be applied as a response to an "offer", or an "answer" may be applied as a response to an "offer" or as an
update to a previously sent "pranswer".</t> update to a previously sent "pranswer".</t>
<t>The only difference between a provisional and final answer <t>The only difference between a provisional and final answer
is that the final answer results in the freeing of any unused is that the final answer results in the freeing of any unused
resources that were allocated as a result of the offer. As resources that were allocated as a result of the offer. As
such, the application can use some discretion on whether an such, the application can use some discretion on whether an
answer should be applied as provisional or final, and can answer should be applied as provisional or final and can
change the type of the session description as needed. For change the type of the session description as needed. For
example, in a serial forking scenario, an application may example, in a serial forking scenario, an application may
receive multiple "final" answers, one from each remote receive multiple "final" answers, one from each remote
endpoint. The application could choose to accept the initial endpoint. The application could choose to accept the initial
answers as provisional answers, and only apply an answer as answers as provisional answers and only apply an answer as
final when it receives one that meets its criteria (e.g. a final when it receives one that meets its criteria (e.g., a
live user instead of voicemail).</t> live user instead of voicemail).</t>
<t>"rollback" is a special session description type implying <t>"rollback" is a special session description type implying
that the state machine should be rolled back to the previous that the state machine should be rolled back to the previous
stable state, as described in stable state, as described in
<xref target="sec.rollback" />. The contents MUST be <xref target="sec.rollback" format="default"/>. The contents <bcp14>MU ST</bcp14> be
empty.</t> empty.</t>
<section title="Use of Provisional Answers" <section anchor="sec.use-of-provisional-answer" numbered="true" toc="d
anchor="sec.use-of-provisional-answer"> efault">
<name>Use of Provisional Answers</name>
<t>Most applications will not need to create answers using <t>Most applications will not need to create answers using
the "pranswer" type. While it is good practice to send an the "pranswer" type. While it is good practice to send an
immediate response to an offer, in order to warm up the immediate response to an offer, in order to warm up the
session transport and prevent media clipping, the preferred session transport and prevent media clipping, the preferred
handling for a JSEP application is to create and send a handling for a JSEP application is to create and send a
"sendonly" final answer with a null MediaStreamTrack "sendonly" final answer with a null MediaStreamTrack
immediately after receiving the offer, which will prevent immediately after receiving the offer, which will prevent
media from being sent by the caller, and allow media to be media from being sent by the caller and allow media to be
sent immediately upon answer by the callee. Later, when the sent immediately upon answer by the callee. Later, when the
callee actually accepts the call, the application can plug callee actually accepts the call, the application can plug
in the real MediaStreamTrack and create a new "sendrecv" in the real MediaStreamTrack and create a new "sendrecv"
offer to update the previous offer/answer pair and start offer to update the previous offer/answer pair and start
bidirectional media flow. While this could also be done bidirectional media flow. While this could also be done
with a "sendonly" pranswer, followed by a "sendrecv" with a "sendonly" pranswer, followed by a "sendrecv"
answer, the initial pranswer leaves the offer-answer answer, the initial pranswer leaves the offer/answer
exchange open, which means that the caller cannot send an exchange open, which means that the caller cannot send an
updated offer during this time.</t> updated offer during this time.
<!-- [rfced] Section 4.1.8.1: We had trouble with this sentence.
If neither suggestion below is correct, please clarify.
Original:
While
this could also be done with a "sendonly" pranswer, followed by a
"sendrecv" answer, the initial pranswer leaves the offer-answer
exchange open, which means that the caller cannot send an updated
offer during this time.
Suggestion #1:
While
this could also be done with a "sendonly" pranswer, if followed by a
"sendrecv" answer the initial pranswer leaves the offer/answer
exchange open, which means that the caller cannot send an updated
offer during this time.
Suggestion #2:
While
this could also be done with a "sendonly" pranswer followed by a
"sendrecv" answer, the initial pranswer leaves the offer/answer
exchange open, which means that the caller cannot send an updated
offer during this time. -->
</t>
<t>As an example, consider a typical JSEP application that <t>As an example, consider a typical JSEP application that
wants to set up audio and video as quickly as possible. wants to set up audio and video as quickly as possible.
When the callee receives an offer with audio and video When the callee receives an offer with audio and video
MediaStreamTracks, it will send an immediate answer MediaStreamTracks, it will send an immediate answer
accepting these tracks as sendonly (meaning that the caller accepting these tracks as sendonly (meaning that the caller
will not send the callee any media yet, and because the will not send the callee any media yet, and because the
callee has not yet added its own MediaStreamTracks, the callee has not yet added its own MediaStreamTracks, the
callee will not send any media either). It will then ask callee will not send any media either). It will then ask
the user to accept the call and acquire the needed local the user to accept the call and acquire the needed local
tracks. Upon acceptance by the user, the application will tracks. Upon acceptance by the user, the application will
plug in the tracks it has acquired, which, because ICE and plug in the tracks it has acquired, which, because ICE handshaking
DTLS handshaking have likely completed by this point, can and DTLS handshaking have likely completed by this point, can
start transmitting immediately. The application will also start transmitting immediately.
<!-- [rfced] Section 4.1.8.1: As it appears that "ICE and DTLS
handshaking have" means "ICE handshaking and DTLS handshaking have,"
we updated this sentence accordingly. Please let us know if this is
incorrect (i.e., if the text refers to one handshaking process, in
which case "have" should be "has").
Original:
Upon acceptance by the user, the
application will plug in the tracks it has acquired, which, because
ICE and DTLS handshaking have likely completed by this point, can
start transmitting immediately.
Currently:
Upon acceptance by the user, the
application will plug in the tracks it has acquired, which, because
ICE handshaking and DTLS handshaking have likely completed by this
point, can start transmitting immediately. -->
The application will also
send a new offer to the remote side indicating call send a new offer to the remote side indicating call
acceptance and moving the audio and video to be two-way acceptance and moving the audio and video to be two-way
media. A detailed example flow along these lines is shown media. A detailed example flow along these lines is shown
in in
<xref target="sec.warmup-example"></xref>.</t> <xref target="sec.warmup-example" format="default"/>.</t>
<t>Of course, some applications may not be able to perform <t>Of course, some applications may not be able to perform
this double offer-answer exchange, particularly ones that this double offer/answer exchange, particularly ones that
are attempting to gateway to legacy signaling protocols. In are attempting to gateway to legacy signaling protocols. In
these cases, pranswer can still provide the application these cases, pranswer can still provide the application
with a mechanism to warm up the transport.</t> with a mechanism to warm up the transport.</t>
</section> </section>
<section title="Rollback" anchor="sec.rollback"> <section anchor="sec.rollback" numbered="true" toc="default">
<name>Rollback</name>
<t>In certain situations it may be desirable to "undo" a <t>In certain situations, it may be desirable to "undo" a
change made to setLocalDescription or setRemoteDescription. change made to setLocalDescription or setRemoteDescription.
Consider a case where a call is ongoing, and one side wants Consider a case where a call is ongoing and one side wants
to change some of the session parameters; that side to change some of the session parameters; that side
generates an updated offer and then calls generates an updated offer and then calls
setLocalDescription. However, the remote side, either setLocalDescription. However, the remote side, either
before or after setRemoteDescription, decides it does not before or after setRemoteDescription, decides it does not
want to accept the new parameters, and sends a reject want to accept the new parameters and sends a reject
message back to the offerer. Now, the offerer, and possibly message back to the offerer. Now, the offerer, and possibly
the answerer as well, need to return to a stable state and the answerer as well, needs to return to a stable state and
the previous local/remote description. To support this, we the previous local/remote description. To support this, we
introduce the concept of "rollback", which discards any introduce the concept of "rollback", which discards any
proposed changes to the session, returning the state proposed changes to the session, returning the state
machine to the stable state. A rollback is performed by machine to the stable state. A rollback is performed by
supplying a session description of type "rollback" with supplying a session description of type "rollback" with
empty contents to either setLocalDescription or empty contents to either setLocalDescription or
setRemoteDescription.</t> setRemoteDescription.</t>
</section> </section>
</section> </section>
<section title="setLocalDescription" <section anchor="sec.setlocaldescription" numbered="true" toc="default">
anchor="sec.setlocaldescription"> <name>setLocalDescription</name>
<t>The setLocalDescription method instructs the <t>The setLocalDescription method instructs the
PeerConnection to apply the supplied session description as PeerConnection to apply the supplied session description as
its local configuration. The type field indicates whether the its local configuration. The type field indicates whether the
description should be processed as an offer, provisional description should be processed as an offer, provisional
answer, final answer, or rollback; offers and answers are answer, final answer, or rollback; offers and answers are
checked differently, using the various rules that exist for checked differently, using the various rules that exist for
each SDP line.</t> each SDP line.</t>
<t>This API changes the local media state; among other <t>This API changes the local media state; among other
things, it sets up local resources for receiving and decoding things, it sets up local resources for receiving and decoding
media. In order to successfully handle scenarios where the media. In order to successfully handle scenarios where the
application wants to offer to change from one media format to application wants to offer to change from one media format to
a different, incompatible format, the PeerConnection must be a different, incompatible format, the PeerConnection must be
able to simultaneously support use of both the current and able to simultaneously support use of both the current and
pending local descriptions (e.g., support the codecs that pending local descriptions (e.g., support the codecs that
exist in either description). This dual processing begins exist in either description). This dual processing begins
when the PeerConnection enters the "have-local-offer" state, when the PeerConnection enters the "have-local-offer" state,
and continues until setRemoteDescription is called with and it continues until setRemoteDescription is called with
either a final answer, at which point the PeerConnection can either (1)&nbsp;a final answer, at which point the PeerConnection can
fully adopt the pending local description, or a rollback, fully adopt the pending local description or (2)&nbsp;a rollback,
which results in a revert to the current local which results in a revert to the current local
description.</t> description.</t>
<t>This API indirectly controls the candidate gathering <t>This API indirectly controls the candidate gathering
process. When a local description is supplied, and the number process. When a local description is supplied and the number
of transports currently in use does not match the number of of transports currently in use does not match the number of
transports needed by the local description, the transports needed by the local description, the
PeerConnection will create transports as needed and begin PeerConnection will create transports as needed and begin
gathering candidates for each transport, using ones from the gathering candidates for each transport, using ones from the
candidate pool if available.</t> candidate pool if available.</t>
<t>If setRemoteDescription was previously called with an <t>If setRemoteDescription was previously called with an
offer, and setLocalDescription is called with an answer offer, and setLocalDescription is called with an answer
(provisional or final), and the media directions are (provisional or final), and the media directions are
compatible, and media is available to send, this will result compatible, and media is available to send, this will result
in the starting of media transmission.</t> in the starting of media transmission.
</section>
<section title="setRemoteDescription" <!-- [rfced] Sections 4.1.9 and 4.1.10: We had trouble following the
anchor="sec.setremotedescription"> purpose of all of the "and"s in these sentences. Are four conditions
set in these sentences, or fewer?
Original:
If setRemoteDescription was previously called with an offer, and
setLocalDescription is called with an answer (provisional or final),
and the media directions are compatible, and media is available to
send, this will result in the starting of media transmission.
...
If setLocalDescription was previously called with an offer, and
setRemoteDescription is called with an answer (provisional or final),
and the media directions are compatible, and media is available to
send, this will result in the starting of media transmission.
Possibly:
If (1) setRemoteDescription was previously called with an offer,
(2) setLocalDescription is called with an answer (provisional or
final), (3) the media directions are compatible, and (4) media is
available to send, media transmission can start.
...
If (1) setLocalDescription was previously called with an offer,
(2) setRemoteDescription is called with an answer (provisional or
final), (3) the media directions are compatible, and (4) media is
available to send, media transmission can start. -->
</t>
</section>
<section anchor="sec.setremotedescription" numbered="true" toc="default"
>
<name>setRemoteDescription</name>
<t>The setRemoteDescription method instructs the <t>The setRemoteDescription method instructs the
PeerConnection to apply the supplied session description as PeerConnection to apply the supplied session description as
the desired remote configuration. As in setLocalDescription, the desired remote configuration. As in setLocalDescription,
the type field of the description indicates how it should be the type field of the description indicates how it should be
processed.</t> processed.</t>
<t>This API changes the local media state; among other <t>This API changes the local media state; among other
things, it sets up local resources for sending and encoding things, it sets up local resources for sending and encoding
media.</t> media.
<!-- [rfced] Section 4.1.10: Please confirm that "local media state"
and "local resources" (as opposed to remote) are correct in the
context of setRemoteDescription. (We ask because we see identical
wording in Section 4.1.9 ("setLocalDescription").)
Original:
This API changes the local media state; among other things, it sets
up local resources for sending and encoding media. -->
</t>
<t>If setLocalDescription was previously called with an <t>If setLocalDescription was previously called with an
offer, and setRemoteDescription is called with an answer offer, and setRemoteDescription is called with an answer
(provisional or final), and the media directions are (provisional or final), and the media directions are
compatible, and media is available to send, this will result compatible, and media is available to send, this will result
in the starting of media transmission.</t> in the starting of media transmission.</t>
</section> </section>
<section title="currentLocalDescription" <section anchor="sec.currentlocaldescription" numbered="true" toc="defau
anchor="sec.currentlocaldescription"> lt">
<name>currentLocalDescription</name>
<t>The currentLocalDescription method returns the current <t>The currentLocalDescription method returns the current
negotiated local description - i.e., the local description negotiated local description -- i.e., the local description
from the last successful offer/answer exchange - in addition from the last successful offer/answer exchange -- in addition
to any local candidates that have been generated by the ICE to any local candidates that have been generated by the ICE
agent since the local description was set.</t> agent since the local description was set.</t>
<t>A null object will be returned if an offer/answer exchange <t>A null object will be returned if an offer/answer exchange
has not yet been completed.</t> has not yet been completed.</t>
</section> </section>
<section title="pendingLocalDescription" <section anchor="sec.pendinglocaldescription" numbered="true" toc="defau
anchor="sec.pendinglocaldescription"> lt">
<name>pendingLocalDescription</name>
<t>The pendingLocalDescription method returns a copy of the <t>The pendingLocalDescription method returns a copy of the
local description currently in negotiation - i.e., a local local description currently in negotiation -- i.e., a local
offer set without any corresponding remote answer - in offer set without any corresponding remote answer -- in
addition to any local candidates that have been generated by addition to any local candidates that have been generated by
the ICE agent since the local description was set.</t> the ICE agent since the local description was set.</t>
<t>A null object will be returned if the state of the <t>A null object will be returned if the state of the
PeerConnection is "stable" or "have-remote-offer".</t> PeerConnection is "stable" or "have-remote-offer".</t>
</section> </section>
<section title="currentRemoteDescription" <section anchor="sec.currentremotedescription" numbered="true" toc="defa
anchor="sec.currentremotedescription"> ult">
<name>currentRemoteDescription</name>
<t>The currentRemoteDescription method returns a copy of the <t>The currentRemoteDescription method returns a copy of the
current negotiated remote description - i.e., the remote current negotiated remote description -- i.e., the remote
description from the last successful offer/answer exchange - description from the last successful offer/answer exchange --
in addition to any remote candidates that have been supplied in addition to any remote candidates that have been supplied
via processIceMessage since the remote description was via processIceMessage since the remote description was
set.</t> set.</t>
<t>A null object will be returned if an offer/answer exchange <t>A null object will be returned if an offer/answer exchange
has not yet been completed.</t> has not yet been completed.</t>
</section> </section>
<section title="pendingRemoteDescription" <section anchor="sec.pendingremotedescription" numbered="true" toc="defa
anchor="sec.pendingremotedescription"> ult">
<name>pendingRemoteDescription</name>
<t>The pendingRemoteDescription method returns a copy of the <t>The pendingRemoteDescription method returns a copy of the
remote description currently in negotiation - i.e., a remote remote description currently in negotiation -- i.e., a remote
offer set without any corresponding local answer - in offer set without any corresponding local answer -- in
addition to any remote candidates that have been supplied via addition to any remote candidates that have been supplied via
processIceMessage since the remote description was set.</t> processIceMessage since the remote description was set.</t>
<t>A null object will be returned if the state of the <t>A null object will be returned if the state of the
PeerConnection is "stable" or "have-local-offer".</t> PeerConnection is "stable" or "have-local-offer".</t>
</section> </section>
<section title="canTrickleIceCandidates" <section anchor="sec.cantrickle" numbered="true" toc="default">
anchor="sec.cantrickle"> <name>canTrickleIceCandidates</name>
<t>The canTrickleIceCandidates property indicates whether the <t>The canTrickleIceCandidates property indicates whether the
remote side supports receiving trickled candidates. There are remote side supports receiving trickled candidates. There are
three potential values: three potential values:
<list style="hanging"> </t>
<t hangText="null:">No SDP has been received from the other <dl newline="false" spacing="normal">
<dt>null:</dt>
<dd>No SDP has been received from the other
side, so it is not known if it can handle trickle. This is side, so it is not known if it can handle trickle. This is
the initial value before setRemoteDescription() is the initial value before setRemoteDescription() is
called.</t> called.</dd>
<t hangText="true:">SDP has been received from the other <dt>true:</dt>
side indicating that it can support trickle.</t> <dd>SDP has been received from the other
<t hangText="false:">SDP has been received from the other side indicating that it can support trickle.</dd>
side indicating that it cannot support trickle.</t> <dt>false:</dt>
</list></t> <dd>SDP has been received from the other
side indicating that it cannot support trickle.</dd>
</dl>
<t>As described in <t>As described in
<xref target="sec.ice-candidate-trickling" />, JSEP <xref target="sec.ice-candidate-trickling" format="default"/>, JSEP
implementations always provide candidates to the application implementations always provide candidates to the application
individually, consistent with what is needed for Trickle ICE. individually, consistent with what is needed for Trickle ICE.
However, applications can use the canTrickleIceCandidates However, applications can use the canTrickleIceCandidates
property to determine whether their peer can actually do property to determine whether their peer can actually do
Trickle ICE, i.e., whether it is safe to send an initial Trickle ICE, i.e., whether it is safe to send an initial
offer or answer followed later by candidates as they are offer or answer followed later by candidates as they are
gathered. As "true" is the only value that definitively gathered. As "true" is the only value that definitively
indicates remote Trickle ICE support, an application which indicates remote Trickle ICE support, an application that
compares canTrickleIceCandidates against "true" will by compares canTrickleIceCandidates against "true" will by
default attempt Half Trickle on initial offers and Full default attempt Half Trickle on initial offers and Full
Trickle on subsequent interactions with a Trickle Trickle on subsequent interactions with a Trickle
ICE-compatible agent.</t> ICE-compatible agent.</t>
</section> </section>
<section title="setConfiguration" <section anchor="sec.setconfiguration" numbered="true" toc="default">
anchor="sec.setconfiguration"> <name>setConfiguration</name>
<t>The setConfiguration method allows the global <t>The setConfiguration method allows the global
configuration of the PeerConnection, which was initially set configuration of the PeerConnection, which was initially set
by constructor parameters, to be changed during the session. by constructor parameters, to be changed during the session.
The effects of this method call depend on when it is invoked, The effects of this method call depend on when it is invoked,
and differ depending on which specific parameters are and they will differ, depending on which specific parameters are
changed:</t> changed:
<t> <!-- [rfced] Section 4.1.16: Should "The effects of this method call"
<list style="symbols"> be "The effects of calling this method," and should "This call" be
"Calling this method"?
<t>Any changes to the STUN/TURN servers to use affect the Original:
The effects of this method call
depend on when it is invoked, and differ depending on which specific
parameters are changed:
...
This call may result in a change to the state of the ICE Agent. -->
</t>
<ul spacing="normal">
<li>Any changes to the STUN/TURN servers to use affect the
next gathering phase. If an ICE gathering phase has next gathering phase. If an ICE gathering phase has
already started or completed, the 'needs-ice-restart' bit already started or completed, the 'needs-ice-restart' bit
mentioned in mentioned in
<xref target="sec.ice-gather-overview" /> will be set. <xref target="sec.ice-gather-overview" format="default"/> will be set.
This will cause the next call to createOffer to generate This will cause the next call to createOffer to generate
new ICE credentials, for the purpose of forcing an ICE new ICE credentials, for the purpose of forcing an ICE
restart and kicking off a new gathering phase, in which restart and kicking off a new gathering phase, in which
the new servers will be used. If the ICE candidate pool the new servers will be used. If the ICE candidate pool
has a nonzero size, and a local description has not yet has a nonzero size and a local description has not yet
been applied, any existing candidates will be discarded, been applied, any existing candidates will be discarded,
and new candidates will be gathered from the new and new candidates will be gathered from the new
servers.</t> servers.</li>
<li>Any change to the ICE candidate policy affects the
<t>Any change to the ICE candidate policy affects the
next gathering phase. If an ICE gathering phase has next gathering phase. If an ICE gathering phase has
already started or completed, the 'needs-ice-restart' bit already started or completed, the 'needs-ice-restart' bit
will be set. Either way, changes to the policy have no will be set. Either way, changes to the policy have no
effect on the candidate pool, because pooled candidates effect on the candidate pool, because pooled candidates
are not made available to the application until a are not made available to the application until a
gathering phase occurs, and so any necessary filtering gathering phase occurs, and so any necessary filtering
can still be done on any pooled candidates.</t> can still be done on any pooled candidates.</li>
<li>The ICE candidate pool size <bcp14>MUST NOT</bcp14> be changed a
<t>The ICE candidate pool size MUST NOT be changed after fter
applying a local description. If a local description has applying a local description. If a local description has
not yet been applied, any changes to the ICE candidate not yet been applied, any changes to the ICE candidate
pool size take effect immediately; if increased, pool size take effect immediately; if increased,
additional candidates are pre-gathered; if decreased, the additional candidates are pre-gathered; if decreased, the
now-superfluous candidates are discarded.</t> now-superfluous candidates are discarded.</li>
<li>The bundle and RTCP-multiplexing policies <bcp14>MUST NOT</bcp14
<t>The bundle and RTCP-multiplexing policies MUST NOT be > be
changed after the construction of the PeerConnection.</t> changed after the construction of the PeerConnection.</li>
</list> </ul>
</t>
<t>This call may result in a change to the state of the ICE <t>This call may result in a change to the state of the ICE
Agent.</t> agent.</t>
</section> </section>
<section title="addIceCandidate" anchor="sec.addicecandidate"> <section anchor="sec.addicecandidate" numbered="true" toc="default">
<name>addIceCandidate</name>
<t>The addIceCandidate method provides an update to the ICE <t>The addIceCandidate method provides an update to the ICE
agent via an IceCandidate object agent via an IceCandidate object
<xref target="sec.ice-candidate-format" />. If the (<xref target="sec.ice-candidate-format" format="default"/>). If the
IceCandidate's candidate field is filled in, the IceCandidate IceCandidate's candidate field is filled in, the IceCandidate
is treated as a new remote ICE candidate, which will be added is treated as a new remote ICE candidate, which will be added
to the current and/or pending remote description according to to the current and/or pending remote description according to
the rules defined for Trickle ICE. Otherwise, the the rules defined for Trickle ICE. Otherwise, the
IceCandidate is treated as an end-of-candidates indication, IceCandidate is treated as an end-of-candidates indication,
as defined in as defined in
<xref target="I-D.ietf-ice-trickle" />.</t> <xref target="RFC8838" format="default"/>.</t>
<t>In either case, the "m=" section index, MID, and ufrag
<t>In either case, the m= section index, MID, and ufrag
fields from the supplied IceCandidate are used to determine fields from the supplied IceCandidate are used to determine
which m= section and ICE candidate generation the which "m=" section and ICE candidate generation the
IceCandidate belongs to, as described in IceCandidate belongs to, as described in
<xref target="sec.ice-candidate-format" /> above. In the case <xref target="sec.ice-candidate-format" format="default"/> above. In t he case
of an end-of-candidates indication, the absence of both the of an end-of-candidates indication, the absence of both the
m= section index and MID fields is interpreted to mean that "m=" section index and MID fields is interpreted to mean that
the indication applies to all m= sections in the specified the indication applies to all "m=" sections in the specified
ICE candidate generation. However, if both fields are absent ICE candidate generation. However, if both fields are absent
for a new remote candidate, this MUST be treated as an for a new remote candidate, this <bcp14>MUST</bcp14> be treated as an
invalid condition, as specified below.</t> invalid condition, as specified below.</t>
<t>If any IceCandidate fields contain invalid values or an
<t>If any IceCandidate fields contain invalid values, or an
error occurs during the processing of the IceCandidate error occurs during the processing of the IceCandidate
object, the supplied IceCandidate MUST be ignored and an object, the supplied IceCandidate <bcp14>MUST</bcp14> be ignored and a
error MUST be returned.</t> n
error <bcp14>MUST</bcp14> be returned.</t>
<t>Otherwise, the new remote candidate or end-of-candidates <t>Otherwise, the new remote candidate or end-of-candidates
indication is supplied to the ICE agent. In the case of a new indication is supplied to the ICE agent. In the case of a new
remote candidate, connectivity checks will be sent to the new remote candidate, connectivity checks will be sent to the new
candidate.</t> candidate.</t>
</section> </section>
</section> </section>
<section title="RtpTransceiver" anchor="sec.transceiver"> <section anchor="sec.transceiver" numbered="true" toc="default">
<section title="stop" anchor="sec.transceiver-stop"> <name>RtpTransceiver</name>
<section anchor="sec.transceiver-stop" numbered="true" toc="default">
<name>stop</name>
<t>The stop method stops an RtpTransceiver. This will cause <t>The stop method stops an RtpTransceiver. This will cause
future calls to createOffer to generate a zero port for the future calls to createOffer to generate a zero port for the
associated m= section. See below for more details.</t> associated "m=" section. See below for more details.</t>
</section> </section>
<section title="stopped" anchor="sec.transceiver-stopped"> <section anchor="sec.transceiver-stopped" numbered="true" toc="default">
<name>stopped</name>
<t>The stopped property indicates whether the transceiver has <t>The stopped property indicates whether the transceiver has
been stopped, either by a call to stopTransceiver or by been stopped, either by a call to stopTransceiver or by
applying an answer that rejects the associated m= section. In applying an answer that rejects the associated "m=" section.
either of these cases, it is set to "true", and otherwise
will be set to "false".</t> <!-- [rfced] Section 4.2.2: Will "stopTransceiver" be clear to
readers? We could not find this string elsewhere in this document,
anywhere else in this cluster of documents, in any published RFC, or
in google searches.
Original:
The stopped property indicates whether the transceiver has been
stopped, either by a call to stopTransceiver or by applying an answer
that rejects the associated m= section. -->
In
either of these cases, it is set to "true" and otherwise
will be set to "false".</t>
<t>A stopped RtpTransceiver does not send any outgoing RTP or <t>A stopped RtpTransceiver does not send any outgoing RTP or
RTCP or process any incoming RTP or RTCP. It cannot be RTCP or process any incoming RTP or RTCP. It cannot be
restarted.</t> restarted.</t>
</section> </section>
<section title="setDirection" <section anchor="sec.transceiver-set-direction" numbered="true" toc="def
anchor="sec.transceiver-set-direction"> ault">
<name>setDirection</name>
<t>The setDirection method sets the direction of a <t>The setDirection method sets the direction of a
transceiver, which affects the direction property of the transceiver, which affects the direction property of the
associated m= section on future calls to createOffer and associated "m=" section on future calls to createOffer and
createAnswer. The permitted values for direction are createAnswer. The permitted values for direction are
"recvonly", "sendrecv", "sendonly", and "inactive", mirroring "recvonly", "sendrecv", "sendonly", and "inactive", mirroring
the identically-named directional attributes defined in the identically named directional attributes defined in
<xref target="RFC4566"></xref>, Section 6.</t> <xref target="RFC4566" sectionFormat="comma" section="6"/>.</t>
<t>When creating offers, the transceiver direction is <t>When creating offers, the transceiver direction is
directly reflected in the output, even for re-offers. When directly reflected in the output, even for re-offers. When
creating answers, the transceiver direction is intersected creating answers, the transceiver direction is intersected
with the offered direction, as explained in with the offered direction, as explained in
<xref target="sec.generating-an-answer" /> below.</t> <xref target="sec.generating-an-answer" format="default"/> below.</t>
<t>Note that while setDirection sets the direction property <t>Note that while setDirection sets the direction property
of the transceiver immediately ( of the transceiver immediately (<xref target="sec.transceiver-directio
<xref target="sec.transceiver-direction" />), this property n" format="default"/>), this property
does not immediately affect whether the transceiver's does not immediately affect whether the transceiver's
RtpSender will send or its RtpReceiver will receive. The RtpSender will send or its RtpReceiver will receive. The
direction in effect is represented by the currentDirection direction in effect is represented by the currentDirection
property, which is only updated when an answer is property, which is only updated when an answer is
applied.</t> applied.</t>
</section> </section>
<section title="direction" anchor="sec.transceiver-direction"> <section anchor="sec.transceiver-direction" numbered="true" toc="default
">
<name>direction</name>
<t>The direction property indicates the last value passed <t>The direction property indicates the last value passed
into setDirection. If setDirection has never been called, it into setDirection. If setDirection has never been called, it
is set to the direction the transceiver was initialized is set to the direction the transceiver was initialized
with.</t> with.</t>
</section> </section>
<section title="currentDirection" <section anchor="sec.transceiver-current-direction" numbered="true" toc=
anchor="sec.transceiver-current-direction"> "default">
<name>currentDirection</name>
<t>The currentDirection property indicates the last <t>The currentDirection property indicates the last
negotiated direction for the transceiver's associated m= negotiated direction for the transceiver's associated "m="
section. More specifically, it indicates the section. More specifically, it indicates the
<xref target="RFC3264"></xref> directional attribute of the directional attribute <xref target="RFC3264" format="default"/> of the
associated m= section in the last applied answer (including associated "m=" section in the last applied answer (including
provisional answers), with "send" and "recv" directions provisional answers), with "send" and "recv" directions
reversed if it was a remote answer. For example, if the reversed if it was a remote answer.
directional attribute for the associated m= section in a
<!-- [rfced] Section 4.2.5: We see "direction attribute" but not
"directional attribute" in RFC 3264. Will this text be clear to
readers? (We ask because we also see "A direction attribute,
determined by applying the rules regarding the offered direction
specified in [RFC3264], Section 6.1" in Section 5.3.1 of this
document.)
Original:
More specifically, it
indicates the [RFC3264] directional attribute of the associated m=
section in the last applied answer (including provisional answers),
with "send" and "recv" directions reversed if it was a remote answer. -->
For example, if the
directional attribute for the associated "m=" section in a
remote answer is "recvonly", currentDirection is set to remote answer is "recvonly", currentDirection is set to
"sendonly".</t> "sendonly".</t>
<t>If an answer that references this transceiver has not yet <t>If an answer that references this transceiver has not yet
been applied, or if the transceiver is stopped, been applied or if the transceiver is stopped,
currentDirection is set to null.</t> currentDirection is set to "null".</t>
</section> </section>
<section title="setCodecPreferences" <section anchor="sec.transceiver-set-codec-preferences" numbered="true"
anchor="sec.transceiver-set-codec-preferences"> toc="default">
<name>setCodecPreferences</name>
<t>The setCodecPreferences method sets the codec preferences <t>The setCodecPreferences method sets the codec preferences
of a transceiver, which in turn affect the presence and order of a transceiver, which in turn affect the presence and order
of codecs of the associated m= section on future calls to of codecs of the associated "m=" section on future calls to
createOffer and createAnswer. Note that setCodecPreferences createOffer and createAnswer. Note that setCodecPreferences
does not directly affect which codec the implementation does not directly affect which codec the implementation
decides to send. It only affects which codecs the decides to send. It only affects which codecs the
implementation indicates that it prefers to receive, via the implementation indicates that it prefers to receive, via the
offer or answer. Even when a codec is excluded by offer or answer. Even when a codec is excluded by
setCodecPreferences, it still may be used to send until the setCodecPreferences, it still may be used to send until the
next offer/answer exchange discards it.</t> next offer/answer exchange discards it.</t>
<t>The codec preferences of an RtpTransceiver can cause <t>The codec preferences of an RtpTransceiver can cause
codecs to be excluded by subsequent calls to createOffer and codecs to be excluded by subsequent calls to createOffer and
createAnswer, in which case the corresponding media formats createAnswer, in which case the corresponding media formats
in the associated m= section will be excluded. The codec in the associated "m=" section will be excluded. The codec
preferences cannot add media formats that would otherwise not preferences cannot add media formats that would otherwise not
be present.</t> be present.</t>
<t>The codec preferences of an RtpTransceiver can also <t>The codec preferences of an RtpTransceiver can also
determine the order of codecs in subsequent calls to determine the order of codecs in subsequent calls to
createOffer and createAnswer, in which case the order of the createOffer and createAnswer, in which case the order of the
media formats in the associated m= section will follow the media formats in the associated "m=" section will follow the
specified preferences.</t> specified preferences.</t>
</section> </section>
</section> </section>
</section> </section>
<section title="SDP Interaction Procedures" <section anchor="sec.sdp-interaction-procedure" numbered="true" toc="default
anchor="sec.sdp-interaction-procedure"> ">
<name>SDP Interaction Procedures</name>
<t>This section describes the specific procedures to be followed <t>This section describes the specific procedures to be followed
when creating and parsing SDP objects.</t> when creating and parsing SDP objects.</t>
<section title="Requirements Overview" <section anchor="sec.requirements-overview" numbered="true" toc="default">
anchor="sec.requirements-overview"> <name>Requirements Overview</name>
<t>JSEP implementations must comply with the specifications <t>JSEP implementations must comply with the specifications
listed below that govern the creation and processing of offers listed below that govern the creation and processing of offers
and answers.</t> and answers.</t>
<section title="Usage Requirements" <section anchor="sec.usage-requirements" numbered="true" toc="default">
anchor="sec.usage-requirements"> <name>Usage Requirements</name>
<t>All session descriptions handled by JSEP implementations, <t>All session descriptions handled by JSEP implementations,
both local and remote, MUST indicate support for the both local and remote, <bcp14>MUST</bcp14> indicate support for the
following specifications. If any of these are absent, this following specifications. If any of these are absent, this
omission MUST be treated as an error. omission <bcp14>MUST</bcp14> be treated as an error.
<list style="symbols"> </t>
<ul spacing="normal">
<li>ICE, as specified in
<xref target="RFC8445" format="default"/>, <bcp14>MUST</bcp14> be us
ed. Note that the
remote endpoint may use a lite implementation;
implementations <bcp14>MUST</bcp14> properly handle remote endpoints
that
do ICE-lite.</li>
<li>DTLS
<xref target="RFC6347" format="default"/> or DTLS-SRTP
<xref target="RFC5763" format="default"/> <bcp14>MUST</bcp14> be use
d, as
appropriate for the media type, as specified in
<xref target="RFC8827" format="default"/>.</li>
</ul>
<t>The SDES SRTP keying mechanism from
<xref target="RFC4568" format="default"/> <bcp14>MUST NOT</bcp14> be u
sed, as discussed in
<xref target="RFC8827" format="default"/>.
<t>ICE, as specified in <!-- [rfced] Section 5.1.1: Does SDES refer to "source description" or
<xref target="RFC8445"></xref>, MUST be used. Note that the "security description"? Neither [RFC4568] nor RFC 8827
remote endpoint may use a Lite implementation; [I-D.ietf-rtcweb-security-arch] mention "SDES" or "source description" (as
implementations MUST properly handle remote endpoints which used in RFC 8852 and other documents in this cluster).
do ICE-Lite.</t>
<t>DTLS Original:
<xref target="RFC6347" /> or DTLS-SRTP The SDES SRTP keying mechanism from [RFC4568] MUST NOT be used, as
<xref target="RFC5763"></xref>, MUST be used, as discussed in [I-D.ietf-rtcweb-security-arch]. -->
appropriate for the media type, as specified in
<xref target="I-D.ietf-rtcweb-security-arch" /></t>
</list></t>
<t>The SDES SRTP keying mechanism from </t>
<xref target="RFC4568" /> MUST NOT be used, as discussed in
<xref target="I-D.ietf-rtcweb-security-arch" />.</t>
</section> </section>
<section title="Profile Names and Interoperability" <section anchor="sec.profile-names" numbered="true" toc="default">
anchor="sec.profile-names"> <name>Profile Names and Interoperability</name>
<t>For media "m=" sections, JSEP implementations <bcp14>MUST</bcp14> s
<t>For media m= sections, JSEP implementations MUST support upport
the "UDP/TLS/RTP/SAVPF" profile specified in the "UDP/TLS/RTP/SAVPF" profile specified in
<xref target="RFC5764"></xref> as well as the "TCP/DTLS/RTP/SAVPF" <xref target="RFC5764" format="default"/> as well as the "TCP/DTLS/RTP
profile specified in <xref target="RFC7850"></xref>, and MUST indicate /SAVPF"
one of these profiles for each media m= line they produce in an offer. profile specified in <xref target="RFC7850" format="default"/> and <bc
For data m= sections, implementations MUST support the p14>MUST</bcp14> indicate
"UDP/DTLS/SCTP" profile as well as the "TCP/DTLS/SCTP" profile, and one of these profiles for each media "m=" line they produce in an offe
MUST indicate one of these profiles for each data m= line they produce r.
For data "m=" sections, implementations <bcp14>MUST</bcp14> support th
e
"UDP/DTLS/SCTP" profile as well as the "TCP/DTLS/SCTP" profile and
<bcp14>MUST</bcp14> indicate one of these profiles for each data "m="
line they produce
in an offer. The exact profile to use is determined by the protocol in an offer. The exact profile to use is determined by the protocol
associated with the current default or selected ICE candidate, as associated with the current default or selected ICE candidate, as
described in described in
<xref target="I-D.ietf-mmusic-ice-sip-sdp"></xref>, Section 3.2.1.2. <xref target="RFC8839" sectionFormat="comma" section="4.2.1.2"/>.</t>
</t>
<t>Unfortunately, in an attempt at compatibility, some <t>Unfortunately, in an attempt at compatibility, some
endpoints generate other profile strings even when they mean endpoints generate other profile strings even when they mean
to support one of these profiles. For instance, an endpoint to support one of these profiles. For instance, an endpoint
might generate "RTP/AVP" but supply "a=fingerprint" and might generate "RTP/AVP" but supply "a=fingerprint" and
"a=rtcp-fb" attributes, indicating its willingness to support "a=rtcp-fb" attributes, indicating its willingness to support
"UDP/TLS/RTP/SAVPF" or "TCP/DTLS/RTP/SAVPF". In order to "UDP/TLS/RTP/SAVPF" or "TCP/DTLS/RTP/SAVPF". In order to
simplify compatibility with such endpoints, JSEP simplify compatibility with such endpoints, JSEP
implementations MUST follow the following rules when implementations <bcp14>MUST</bcp14> follow the following rules when
processing the media m= sections in a received offer:</t> processing the media "m=" sections in a received offer:</t>
<ul spacing="normal">
<t> <li>
<list style="symbols">
<t>Any profile in the offer matching one of the following <t>Any profile in the offer matching one of the following
MUST be accepted: <bcp14>MUST</bcp14> be accepted:
<list style="symbols"> </t>
<ul spacing="normal">
<t>"RTP/AVP" (Defined in <li>"RTP/AVP" (defined in
<xref target="RFC4566"></xref>, Section 8.2.2)</t> <xref target="RFC4566" sectionFormat="comma" section="8.2.2"/>)<
/li>
<t>"RTP/AVPF" (Defined in <li>"RTP/AVPF" (defined in
<xref target="RFC4585"></xref>, Section 9)</t> <xref target="RFC4585" sectionFormat="comma" section="9"/>)</li>
<li>"RTP/SAVP" (defined in
<t>"RTP/SAVP" (Defined in <xref target="RFC3711" sectionFormat="comma" section="12"/>)</li
<xref target="RFC3711"></xref>, Section 12)</t> >
<li>"RTP/SAVPF" (defined in
<t>"RTP/SAVPF" (Defined in <xref target="RFC5124" sectionFormat="comma" section="6"/>)</li>
<xref target="RFC5124"></xref>, Section 6)</t> <li>"TCP/DTLS/RTP/SAVP" (defined in
<xref target="RFC7850" sectionFormat="comma" section="3.4"/>)</l
<t>"TCP/DTLS/RTP/SAVP" (Defined in i>
<xref target="RFC7850"></xref>, Section 3.4)</t> <li>"TCP/DTLS/RTP/SAVPF" (defined in
<xref target="RFC7850" sectionFormat="comma" section="3.5"/>)</l
<t>"TCP/DTLS/RTP/SAVPF" (Defined in i>
<xref target="RFC7850"></xref>, Section 3.5)</t> <li>"UDP/TLS/RTP/SAVP" (defined in
<xref target="RFC5764" sectionFormat="comma" section="9"/>)</li>
<t>"UDP/TLS/RTP/SAVP" (Defined in <li>"UDP/TLS/RTP/SAVPF" (defined in
<xref target="RFC5764"></xref>, Section 9)</t> <xref target="RFC5764" sectionFormat="comma" section="9"/>)</li>
</ul>
<t>"UDP/TLS/RTP/SAVPF" (Defined in </li>
<xref target="RFC5764"></xref>, Section 9)</t> <li>The profile in any "m=" line in any generated answer
</list></t> <bcp14>MUST</bcp14> exactly match the profile provided in the offe
r.</li>
<t>The profile in any "m=" line in any generated answer <li>Because DTLS-SRTP is <bcp14>REQUIRED</bcp14>, the choice of SAVP
MUST exactly match the profile provided in the offer.</t> or
<t>Because DTLS-SRTP is REQUIRED, the choice of SAVP or
AVP has no effect; support for DTLS-SRTP is determined by AVP has no effect; support for DTLS-SRTP is determined by
the presence of one or more "a=fingerprint" attribute. the presence of one or more "a=fingerprint" attributes.
Note that lack of an "a=fingerprint" attribute will lead Note that lack of an "a=fingerprint" attribute will lead
to negotiation failure.</t> to negotiation failure.</li>
<li>The use of AVPF or AVP simply controls the timing
<t>The use of AVPF or AVP simply controls the timing rules used for RTCP feedback. If AVPF is provided or an
rules used for RTCP feedback. If AVPF is provided, or an
"a=rtcp-fb" attribute is present, assume AVPF timing, "a=rtcp-fb" attribute is present, assume AVPF timing,
i.e., a default value of "trr-int=0". Otherwise, assume i.e., a default value of "trr-int=0". Otherwise, assume
that AVPF is being used in an AVP compatible mode and use that AVPF is being used in an AVP-compatible mode and use
a value of "trr-int=4000".</t> a value of "trr-int=4000".</li>
<li>For data "m=" sections, implementations <bcp14>MUST</bcp14> supp
<t>For data m= sections, implementations MUST support ort
receiving the "UDP/DTLS/SCTP", "TCP/DTLS/SCTP", or receiving the "UDP/DTLS/SCTP", "TCP/DTLS/SCTP", or
"DTLS/SCTP" (for backwards compatibility) profiles.</t> "DTLS/SCTP" (for backwards compatibility) profiles.</li>
</list> </ul>
</t> <t>Note that re-offers by JSEP implementations <bcp14>MUST</bcp14> use
the
<t>Note that re-offers by JSEP implementations MUST use the
correct profile strings even if the initial offer/answer correct profile strings even if the initial offer/answer
exchange used an (incorrect) older profile string. This exchange used an (incorrect) older profile string. This
simplifies JSEP behavior, with minimal downside, as any simplifies JSEP behavior, with minimal downside, as any
remote endpoint that fails to handle such a re-offer will remote endpoint that fails to handle such a re-offer will
also fail to handle a JSEP endpoint's initial offer.</t> also fail to handle a JSEP endpoint's initial offer.</t>
</section> </section>
</section> </section>
<section anchor="sec-create-offer" title="Constructing an Offer"> <section anchor="sec-create-offer" numbered="true" toc="default">
<name>Constructing an Offer</name>
<t>When createOffer is called, a new SDP description must be <t>When createOffer is called, a new SDP description must be
created that includes the functionality specified in created that includes the functionality specified in
<xref target="I-D.ietf-rtcweb-rtp-usage"></xref>. The exact <xref target="RFC8834" format="default"/>. The exact
details of this process are explained below.</t> details of this process are explained below.</t>
<section title="Initial Offers" anchor="sec.initial-offers"> <section anchor="sec.initial-offers" numbered="true" toc="default">
<name>Initial Offers</name>
<t>When createOffer is called for the first time, the result <t>When createOffer is called for the first time, the result
is known as the initial offer.</t> is known as the initial offer.</t>
<t>The first step in generating an initial offer is to <t>The first step in generating an initial offer is to
generate session-level attributes, as specified in generate session-level attributes, as specified in
<xref target="RFC4566"></xref>, Section 5. Specifically: <xref target="RFC4566" sectionFormat="comma" section="5"/>. Specifical
<list style="symbols"> ly:
</t>
<ul spacing="normal">
<li>The first SDP line <bcp14>MUST</bcp14> be "v=0", as specified in
<xref target="RFC4566" sectionFormat="comma" section="5.1"/>.</li>
<li>The second SDP line <bcp14>MUST</bcp14> be an "o=" line, as spec
ified
in
<xref target="RFC4566" sectionFormat="comma" section="5.2"/>.
<t>The first SDP line MUST be "v=0", as specified in <!-- [rfced] Sections 5.2.1 and subsequent: Several instances of
<xref target="RFC4566"></xref>, Section 5.1</t> ", as specified in [RFC..." are confusing as written. Please see the
items below, and let us know if we may either remove the leading
commas or rephrase in these instances (e.g., use "defined" instead of
"specified"). (For example, in the current text it looks like
[RFC4566], Section 5.1 sets the requirement, but we don't see a
requirement listed there - only a definition.)
<t>The second SDP line MUST be an "o=" line, as specified Examples from original:
in o The first SDP line MUST be "v=0", as specified in [RFC4566],
<xref target="RFC4566"></xref>, Section 5.2. The value of Section 5.1
the &lt;username&gt; field SHOULD be "-". The sess-id MUST
o The second SDP line MUST be an "o=" line, as specified in
[RFC4566], Section 5.2.
...
o The third SDP line MUST be a "s=" line, as specified in [RFC4566],
Section 5.3;
...
o A "t=" line MUST be added, as specified in [RFC4566], Section 5.9;
...
The m= line MUST be followed immediately by a "c=" line, as specified
in [RFC4566], Section 5.7.
...
* If RTCP mux is indicated, prepare to demux RTP and RTCP from
the RTP ICE component, as specified in [RFC5761],
Section 5.1.3.
...
If media is already being transmitted,
the same SSRC SHOULD be used unless the clockrate of the new
codec is different, in which case a new SSRC MUST be chosen, as
specified in [RFC7160], Section 3.1. -->
The value of
the &lt;username&gt; field <bcp14>SHOULD</bcp14> be "-". The sess-id
<bcp14>MUST</bcp14>
be representable by a 64-bit signed integer, and the be representable by a 64-bit signed integer, and the
value MUST be less than (2**63)-1. It is RECOMMENDED that the value <bcp14>MUST</bcp14> be less than (2**63)-1
((2<sup>63</sup>)-1).
<!-- [rfced] Section 5.2.1: Equations and other math-related items:
Per <https://www.rfc-editor.org/materials/FAQ-xml2rfcv3.html>,
xml2rfc v3 provides the ability to use superscript (among other
features previously not available). Please note that superscript
will display in the .html and .pdf files but not in the .txt file
(where it appears as "(2^(63))-1"). We added the "superscript"
version of "(2**63)-1" in parentheses here, to show you how it would
appear. Please let us know if you would like to use the superscript
feature.
Also, if there are other items in this document that could use
similar treatment, please let us know.
Original:
The sess-id MUST be representable by a 64-bit signed
integer, and the value MUST be less than (2**63)-1.
Currently:
.xml (to be updated per your preference):
... less than (2**63)-1 ((2<sup>63</sup>)-1). -->
It is <bcp14>RECOMMENDED</bcp14> that the
sess-id be constructed by generating a 64-bit quantity with sess-id be constructed by generating a 64-bit quantity with
the highest bit set to zero and the remaining 63 the highest bit set to zero and the remaining 63
bits being cryptographically random. The value of the bits being cryptographically random. The value of the
&lt;nettype&gt; &lt;addrtype&gt; &lt;unicast-address&gt; &lt;nettype&gt; &lt;addrtype&gt; &lt;unicast-address&gt;
tuple SHOULD be set to a non-meaningful address, such as IN tuple <bcp14>SHOULD</bcp14> be set to a non-meaningful address, such as IN
IP4 0.0.0.0, to prevent leaking a local IP address in this IP4 0.0.0.0, to prevent leaking a local IP address in this
field; this problem is discussed in field; this problem is discussed in
<xref target="I-D.ietf-rtcweb-ip-handling" />. As mentioned in <xref target="RFC8828" format="default"/>. As mentioned in
<xref target="RFC4566"></xref>, the entire o= line needs to <xref target="RFC4566" format="default"/>, the entire "o=" line need
s to
be unique, but selecting a random number for be unique, but selecting a random number for
&lt;sess-id&gt; is sufficient to accomplish this.</t> &lt;sess-id&gt; is sufficient to accomplish this.</li>
<li>The third SDP line <bcp14>MUST</bcp14> be a "s=" line, as specif
<t>The third SDP line MUST be a "s=" line, as specified in ied in
<xref target="RFC4566"></xref>, Section 5.3; to match the <xref target="RFC4566" sectionFormat="comma" section="5.3"/>; to mat
"o=" line, a single dash SHOULD be used as the session ch the
name, e.g. "s=-". Note that this differs from the advice in "o=" line, a single dash <bcp14>SHOULD</bcp14> be used as the sessio
n
name, e.g., "s=-". Note that this differs from the advice in
<xref target="RFC4566" /> which proposes a single space, but <xref target="RFC4566" format="default"/>, which proposes a single s pace, but
as both "o=" and "s=" are meaningless in JSEP, having the as both "o=" and "s=" are meaningless in JSEP, having the
same meaningless value seems clearer.</t> same meaningless value seems clearer.</li>
<li>Session Information ("i="), URI ("u="), Email Address
<t>Session Information ("i="), URI ("u="), Email Address
("e="), Phone Number ("p="), Repeat Times ("r="), and Time ("e="), Phone Number ("p="), Repeat Times ("r="), and Time
Zones ("z=") lines are not useful in this context and Zones ("z=") lines are not useful in this context and
SHOULD NOT be included.</t> <bcp14>SHOULD NOT</bcp14> be included.</li>
<li>Encryption Keys ("k=") lines do not provide sufficient
security and <bcp14>MUST NOT</bcp14> be included.</li>
<li>A "t=" line <bcp14>MUST</bcp14> be added, as specified in
<xref target="RFC4566" sectionFormat="comma" section="5.9"/>; both
&lt;start-time&gt; and &lt;stop-time&gt; <bcp14>SHOULD</bcp14> be se
t to
zero, e.g., "t=0 0".</li>
<li>An "a=ice-options" line with the "trickle" and "ice2"
options <bcp14>MUST</bcp14> be added, as specified in <xref
target="RFC8840" sectionFormat="comma" section="4.1.1"/> and
<xref target="RFC8445" sectionFormat="comma" section="10"/>.
<t>Encryption Keys ("k=") lines do not provide sufficient <!-- [rfced] Section 5.2.1: We found this RFC Editor Note on
security and MUST NOT be included.</t> <https://datatracker.ietf.org/doc/draft-ietf-rtcweb-jsep/writeup/>:
<t>A "t=" line MUST be added, as specified in "OLD:
<xref target="RFC4566"></xref>, Section 5.9; both o An "a=ice-options" line with the "trickle" option MUST be added,
&lt;start-time&gt; and &lt;stop-time&gt; SHOULD be set to as specified in [I-D.ietf-ice-trickle], Section 4.
zero, e.g. "t=0 0".</t>
<t>An "a=ice-options" line with the "trickle" and "ice2" NEW:
options MUST be added, as specified in <xref o An "a=ice-options" line with the "trickle" option MUST be added,
target="I-D.ietf-ice-trickle"></xref>, Section 3 and as specified in [I-D.ietf-mmusic-trickle-ice-sip], Section 4.1.1."
<xref target="RFC8445"></xref>, Section 10.</t>
<t>If WebRTC identity is being used, an "a=identity" line Please note that the "OLD" text does not match what we found in the
as described in provided draft:
<xref target="I-D.ietf-rtcweb-security-arch" />, Section
5.</t>
</list></t>
<t>The next step is to generate m= sections, as specified in o An "a=ice-options" line with the "trickle" and "ice2" options MUST
<xref target="RFC4566" />, Section 5.14. An m= section is be added, as specified in [I-D.ietf-ice-trickle], Section 3 and
[RFC8445], Section 10.
We updated as follows. Is the "and [RFC8445], Section 10" still
applicable?
Currently:
* An "a=ice-options" line with the "trickle" and "ice2" options MUST
be added, as specified in [RFC8840], Section 4.1.1 and [RFC8445],
Section 10. -->
</li>
<li>If WebRTC identity is being used, an "a=identity" line, as descr
ibed in
<xref target="RFC8827" sectionFormat="comma" section="5"/>, needs
to be included.
<!-- [rfced] Section 5.2.1: This is the only bullet item in this
list that (1) is a sentence fragment, unlike the rest and (2) does
not contain "RFC 2119" language. We have updated as follows, but
please let us know if further changes are needed.
Original:
o If WebRTC identity is being used, an "a=identity" line as
described in [I-D.ietf-rtcweb-security-arch], Section 5.
Currently:
* If WebRTC identity is being used, an "a=identity" line, as
described in [RFC8827], Section 5, needs to be included. -->
</li>
</ul>
<t>The next step is to generate "m=" sections, as specified in
<xref target="RFC4566" sectionFormat="comma" section="5.14"/>. An "m="
section is
generated for each RtpTransceiver that has been added to the generated for each RtpTransceiver that has been added to the
PeerConnection, excluding any stopped RtpTransceivers; this PeerConnection, excluding any stopped RtpTransceivers; this
is done in the order the RtpTransceivers were added to the is done in the order the RtpTransceivers were added to the
PeerConnection. If there are no such RtpTransceivers, no m= PeerConnection. If there are no such RtpTransceivers, no "m="
sections are generated; more can be added later, as discussed sections are generated; more can be added later, as discussed
in in
<xref target="RFC3264" />, Section 5.</t> <xref target="RFC3264" sectionFormat="comma" section="5"/>.</t>
<t>For each "m=" section generated for an RtpTransceiver,
<t>For each m= section generated for an RtpTransceiver,
establish a mapping between the transceiver and the index of establish a mapping between the transceiver and the index of
the generated m= section.</t> the generated "m=" section.</t>
<t>Each "m=" section, provided it is not marked as bundle-only,
<t>Each m= section, provided it is not marked as bundle-only, <bcp14>MUST</bcp14> generate a unique set of ICE credentials and gathe
MUST generate a unique set of ICE credentials and gather its r its
own unique set of ICE candidates. Bundle-only m= sections own unique set of ICE candidates. Bundle-only "m=" sections
MUST NOT contain any ICE credentials and MUST NOT gather any <bcp14>MUST NOT</bcp14> contain any ICE credentials and <bcp14>MUST NO
T</bcp14> gather any
candidates.</t> candidates.</t>
<t>For DTLS, all "m=" sections <bcp14>MUST</bcp14> use any and all cer
<t>For DTLS, all m= sections MUST use all the certificate(s) tificates
that have been specified for the PeerConnection; as a result, that have been specified for the PeerConnection; as a result,
they MUST all have the same they <bcp14>MUST</bcp14> all have the same fingerprint value or values
<xref target="RFC8122"></xref> fingerprint value(s), or these <xref target="RFC8122" format="default"/>, or these
value(s) MUST be session-level attributes.</t> values <bcp14>MUST</bcp14> be session-level attributes.</t>
<t>Each "m=" section should be generated as specified in
<t>Each m= section should be generated as specified in <xref target="RFC4566" sectionFormat="comma" section="5.14"/>. For the
<xref target="RFC4566"></xref>, Section 5.14. For the m= line "m=" line
itself, the following rules MUST be followed: itself, the following rules <bcp14>MUST</bcp14> be followed:
<list style="symbols"> </t>
<ul spacing="normal">
<t>If the m= section is marked as bundle-only, then the <li>If the "m=" section is marked as bundle-only, then the
port value MUST be set to 0. Otherwise, the port value is port value <bcp14>MUST</bcp14> be set to 0. Otherwise, the port valu
set to the port of the default ICE candidate for this m= e is
set to the port of the default ICE candidate for this "m="
section, but given that no candidates are available yet, section, but given that no candidates are available yet,
the "dummy" port value of 9 (Discard) MUST be used, as the "dummy" port value of 9 (Discard) <bcp14>MUST</bcp14> be used, a s
indicated in indicated in
<xref target="I-D.ietf-ice-trickle"></xref>, Section <xref target="RFC8840" sectionFormat="comma" section="4.1.1"/>.</li>
5.1.</t> <!-- This instance of ", as specified in" is OK. -->
<li>To properly indicate use of DTLS, the &lt;proto&gt;
field <bcp14>MUST</bcp14> be set to "UDP/TLS/RTP/SAVPF", as specifie
d in
<xref target="RFC5764" sectionFormat="comma" section="8"/>.</li>
<li>If codec preferences have been set for the associated
transceiver, media formats <bcp14>MUST</bcp14> be generated in the
corresponding order and <bcp14>MUST</bcp14> exclude any codecs not
present in the codec preferences.</li>
<li>Unless excluded by the above restrictions, the media
formats <bcp14>MUST</bcp14> include the mandatory audio/video codecs
as
specified in
<xref target="RFC7874" sectionFormat="comma" section="3"/> and
<xref target="RFC7742" sectionFormat="comma" section="5"/>.
<t>To properly indicate use of DTLS, the &lt;proto&gt; <!-- [rfced] Sections 5.2.1, 5.3.1, and 5.11: As it appears that
field MUST be set to "UDP/TLS/RTP/SAVPF", as specified in each instance of "Section" in these sentences refers to the section
<xref target="RFC5764" />, Section 8.</t> number of the cited RFC, as opposed to a section in this document,
we removed the commas after each first-listed section number, as
follows (particularly for any readers of the text file). Please let
us know if anything is incorrect.
<t>If codec preferences have been set for the associated Original:
transceiver, media formats MUST be generated in the o Unless excluded by the above restrictions, the media formats MUST
corresponding order, and MUST exclude any codecs not include the mandatory audio/video codecs as specified in
present in the codec preferences.</t> [RFC7874], Section 3, and [RFC7742], Section 5.
...
o For each media format on the m= line, "a=rtpmap" and "a=fmtp"
lines, as specified in [RFC4566], Section 6, and [RFC3264],
Section 5.1.
...
o For each media format on the m= line, "a=rtpmap" and "a=fmtp"
lines, as specified in [RFC4566], Section 6, and [RFC3264],
Section 6.1.
...
However, new media formats and new RTP header extension values
are permitted in the answer, as described in [RFC3264],
Section 7, and [RFC5285], Section 6.
<t>Unless excluded by the above restrictions, the media Currently:
formats MUST include the mandatory audio/video codecs as * Unless excluded by the above restrictions, the media formats MUST
specified in include the mandatory audio/video codecs as specified in
<xref target="RFC7874"></xref>, Section 3, and [RFC7874], Section 3 and [RFC7742], Section 5.
<xref target="RFC7742"></xref>, Section 5.</t> ...
</list></t> * For each media format on the "m=" line, "a=rtpmap" and "a=fmtp"
lines, as specified in [RFC4566], Section 6 and [RFC3264],
Section 5.1.
...
* For each media format on the "m=" line, "a=rtpmap" and "a=fmtp"
lines, as specified in [RFC4566], Section 6 and [RFC3264],
Section 6.1.
...
However, new media formats and new RTP header extension values
are permitted in the answer, as described in [RFC3264],
Section 7 and [RFC5285], Section 6. -->
</li>
</ul>
<t>The m= line MUST be followed immediately by a "c=" line, <t>The "m=" line <bcp14>MUST</bcp14> be followed immediately by a "c=" line,
as specified in as specified in
<xref target="RFC4566"></xref>, Section 5.7. Again, as no <xref target="RFC4566" sectionFormat="comma" section="5.7"/>. Again, a s no
candidates are available yet, the "c=" line must contain the candidates are available yet, the "c=" line must contain the
"dummy" value "IN IP4 0.0.0.0", as defined in "dummy" value "IN IP4 0.0.0.0", as defined in
<xref target="I-D.ietf-ice-trickle"></xref>, Section 5.1.</t> <xref target="RFC8840" sectionFormat="comma" section="4.1.1"/>.</t>
<t> <t>
<xref target="I-D.ietf-mmusic-sdp-mux-attributes" /> groups <xref target="RFC8859" format="default"/> groups
SDP attributes into different categories. To avoid SDP attributes into different categories. To avoid
unnecessary duplication when bundling, attributes of category unnecessary duplication when bundling, attributes of category
IDENTICAL or TRANSPORT MUST NOT be repeated in bundled m= IDENTICAL or TRANSPORT <bcp14>MUST NOT</bcp14> be repeated in bundled "m="
sections, repeating the guidance from sections, repeating the guidance from
<xref target="I-D.ietf-mmusic-sdp-bundle-negotiation" />, <xref target="RFC8843"
Section 8.1. This includes m= sections for which bundling has sectionFormat="comma" section="8.1"/>.
been negotiated and is still desired, as well as m= sections
marked as bundle-only.</t>
<t>The following attributes, which are of a category other <!-- [rfced] Sections 5.2.1 and 5.2.2: Please confirm that these
than IDENTICAL or TRANSPORT, MUST be included in each m= citations for RFC 8843 [I-D.ietf-mmusic-sdp-bundle-negotiation], Section 8.1
section:</t> are correct; we could not see a relationship.
<t> Original:
<list style="symbols"> To avoid unnecessary duplication when
bundling, attributes of category IDENTICAL or TRANSPORT MUST NOT be
repeated in bundled m= sections, repeating the guidance from
[I-D.ietf-mmusic-sdp-bundle-negotiation], Section 8.1.
...
Instead,
JSEP implementations MUST simply omit parameters in the IDENTICAL
and TRANSPORT categories for bundled m= sections, as described in
[I-D.ietf-mmusic-sdp-bundle-negotiation], Section 8.1. -->
<t>An "a=mid" line, as specified in This includes "m=" sections for which bundling has
<xref target="RFC5888"></xref>, Section 4. All MID values been negotiated and is still desired, as well as "m=" sections
MUST be generated in a fashion that does not leak user marked as bundle-only.</t>
<t>The following attributes, which are of a category other
than IDENTICAL or TRANSPORT, <bcp14>MUST</bcp14> be included in each "
m="
section:</t>
<ul spacing="normal">
<li>An "a=mid" line, as specified in
<xref target="RFC5888" sectionFormat="comma" section="4"/>. All MI
D values
<bcp14>MUST</bcp14> be generated in a fashion that does not leak u
ser
information, e.g., randomly or using a per-PeerConnection information, e.g., randomly or using a per-PeerConnection
counter, and SHOULD be 3 bytes or less, to allow them to counter, and <bcp14>SHOULD</bcp14> be 3 bytes or less, to allow th em to
efficiently fit into the RTP header extension defined in efficiently fit into the RTP header extension defined in
<xref target="I-D.ietf-mmusic-sdp-bundle-negotiation"> <xref target="RFC8843" sectionFormat="comma" section="14"/>.
</xref>, Section 14. Note that this does not set the
RtpTransceiver mid property, as that only occurs when the
description is applied. The generated MID value can be
considered a "proposed" MID at this point.</t>
<t>A direction attribute which is the same as that of the <!-- [rfced] Section 5.2.1: Please confirm that this citation is
associated transceiver.</t> correct; we could not see a relationship between Section 14 of
RFC 8843 [I-D.ietf-mmusic-sdp-bundle-negotiation] and the RTP header extension.
<t>For each media format on the m= line, "a=rtpmap" and Original:
"a=fmtp" lines, as specified in All MID
<xref target="RFC4566"></xref>, Section 6, and values MUST be generated in a fashion that does not leak user
<xref target="RFC3264"></xref>, Section 5.1.</t> information, e.g., randomly or using a per-PeerConnection counter,
and SHOULD be 3 bytes or less, to allow them to efficiently fit
into the RTP header extension defined in
[I-D.ietf-mmusic-sdp-bundle-negotiation], Section 14. -->
<t>For each primary codec where RTP retransmission should Note that this does not set the
RtpTransceiver mid property, as that only occurs when the
description is applied. The generated MID value can be
considered a "proposed" MID at this point.</li>
<li>A direction attribute that is the same as that of the
associated transceiver.</li>
<li>For each media format on the "m=" line, "a=rtpmap" and "a=fmtp"
lines, as specified in
<xref target="RFC4566" sectionFormat="comma" section="6"/> and
<xref target="RFC3264" sectionFormat="comma" section="5.1"/>.</li>
<li>For each primary codec where RTP retransmission should
be used, a corresponding "a=rtpmap" line indicating "rtx" be used, a corresponding "a=rtpmap" line indicating "rtx"
with the clock rate of the primary codec and an "a=fmtp" with the clock rate of the primary codec and an "a=fmtp"
line that references the payload type of the primary line that references the payload type of the primary
codec, as specified in codec, as specified in
<xref target="RFC4588"></xref>, Section 8.1.</t> <xref target="RFC4588" sectionFormat="comma" section="8.1"/>.</li>
<li>For each supported Forward Error Correction (FEC) mechanism, "a=
<t>For each supported FEC mechanism, "a=rtpmap" and rtpmap" and
"a=fmtp" lines, as specified in "a=fmtp" lines, as specified in
<xref target="RFC4566"></xref>, Section 6. The FEC <xref target="RFC4566" sectionFormat="comma" section="6"/>. The FE
mechanisms that MUST be supported are specified in C
<xref target="I-D.ietf-rtcweb-fec"></xref>, Section 6, mechanisms that <bcp14>MUST</bcp14> be supported are specified in
<xref target="RFC8854" sectionFormat="comma" section="6"/>,
and specific usage for each media type is outlined in and specific usage for each media type is outlined in
Sections 4 and 5.</t> Sections <xref target="sec.interface" format="counter"/> and <xref
target="sec.sdp-interaction-procedure"
format="counter"/>.
<t>If this m= section is for media with configurable <!-- [rfced] Sections 5.2.1 and 5.3.1: Please confirm that Section 6
of RFC 8854 [I-D.ietf-rtcweb-fec] is the correct section to cite here; we
could not see a relationship.
Also, should "Sections 4 and 5" be "Sections 4 and 5 of
RFC 8854 [I-D.ietf-rtcweb-fec]"? The hyperlinks in the .html file steer to
Sections 4 and 5 of this document, and we could not find relevant
text in those sections.
Original:
The FEC mechanisms that
MUST be supported are specified in [I-D.ietf-rtcweb-fec],
Section 6, and specific usage for each media type is outlined in
Sections 4 and 5.
...
The FEC mechanisms that
MUST be supported are specified in [I-D.ietf-rtcweb-fec],
Section 6, and specific usage for each media type is outlined in
Sections 4 and 5. -->
</li>
<li>If this "m=" section is for media with configurable
durations of media per packet, e.g., audio, an durations of media per packet, e.g., audio, an
"a=maxptime" line, indicating the maximum amount of "a=maxptime" line, indicating the maximum amount of
media, specified in milliseconds, that can be media, specified in milliseconds, that can be
encapsulated in each packet, as specified in encapsulated in each packet, as specified in
<xref target="RFC4566"></xref>, Section 6. This value is <xref target="RFC4566" sectionFormat="comma" section="6"/>. This v alue is
set to the smallest of the maximum duration values across set to the smallest of the maximum duration values across
all the codecs included in the m= section.</t> all the codecs included in the "m=" section.</li>
<li>If this "m=" section is for video media and there are
<t>If this m= section is for video media, and there are known limitations on the size of images that can be
known limitations on the size of images which can be
decoded, an "a=imageattr" line, as specified in decoded, an "a=imageattr" line, as specified in
<xref target="sec.imageattr"></xref>.</t> <xref target="sec.imageattr" format="default"/>.</li>
<li>For each supported RTP header extension, an "a=extmap"
<t>For each supported RTP header extension, an "a=extmap"
line, as specified in line, as specified in
<xref target="RFC5285"></xref>, Section 5. The list of <xref target="RFC5285" sectionFormat="comma" section="5"/>.
header extensions that SHOULD/MUST be supported is
<!-- [rfced] Sections 5.2.1 and subsequent: RFC 5285 has been
obsoleted by RFC 8285. Should RFC 8285 be cited throughout the
document and listed as a Normative Reference instead of RFC 5285?
If yes, please review all textual citations (eight, by our count),
and let us know if any of the section numbers (Sections 5, 6, and
7 of RFC 5285 are cited) also need to be updated.
Original:
o For each supported RTP header extension, an "a=extmap" line, as
specified in [RFC5285], Section 5.
...
o For each supported RTP header extension that is present in the
offer, an "a=extmap" line, as specified in [RFC5285], Section 5.
...
... etc. -->
The list of
header extensions that <bcp14>SHOULD</bcp14>/<bcp14>MUST</bcp14> b
e supported is
specified in specified in
<xref target="I-D.ietf-rtcweb-rtp-usage"></xref>, Section <xref target="RFC8834" sectionFormat="comma" section="5.2"/>. Any
5.2. Any header extensions that require encryption MUST header extensions that require encryption <bcp14>MUST</bcp14>
be specified as indicated in be specified as indicated in
<xref target="RFC6904"></xref>, Section 4.</t> <xref target="RFC6904" sectionFormat="comma" section="4"/>.</li>
<li>For each supported RTCP feedback mechanism, an
<t>For each supported RTCP feedback mechanism, an
"a=rtcp-fb" line, as specified in "a=rtcp-fb" line, as specified in
<xref target="RFC4585"></xref>, Section 4.2. The list of <xref target="RFC4585" sectionFormat="comma" section="4.2"/>. The
RTCP feedback mechanisms that SHOULD/MUST be supported is list of
RTCP feedback mechanisms that <bcp14>SHOULD</bcp14>/<bcp14>MUST</b
cp14> be supported is
specified in specified in
<xref target="I-D.ietf-rtcweb-rtp-usage"></xref>, Section <xref target="RFC8834" sectionFormat="comma" section="5.1"/>.</li>
5.1.</t> <li>
<t>If the RtpTransceiver has a sendrecv or sendonly <t>If the RtpTransceiver has a sendrecv or sendonly
direction: direction:
<list style="symbols"> </t>
<ul spacing="normal">
<t>For each MediaStream that was associated with the <li>For each MediaStream that was associated with the
transceiver when it was created via addTrack or transceiver when it was created via addTrack or
addTransceiver, an "a=msid" line, as specified in addTransceiver, an "a=msid" line, as specified in
<xref target="I-D.ietf-mmusic-msid"></xref>, Section 2, <xref target="RFC8830" sectionFormat="comma" section="2"/>,
but omitting the "appdata" field.</t> but omitting the "appdata" field.</li>
</list></t> </ul>
</li>
<t>If the RtpTransceiver has a sendrecv or sendonly <li>If the RtpTransceiver has a sendrecv or sendonly
direction, and the application has specified RID values direction, and the application has specified RID values
or has specified more than one encoding in the or has specified more than one encoding in the
RtpSenders's parameters, an "a=rid" line for each RtpSenders's parameters, an "a=rid" line for each
encoding specified. The "a=rid" line is specified in encoding specified. The "a=rid" line is specified in
<xref target="I-D.ietf-mmusic-rid"></xref>, and its <xref target="RFC8851" format="default"/>, and its
direction MUST be "send". If the application has chosen a direction <bcp14>MUST</bcp14> be "send". If the application has ch
RID value, it MUST be used as the rid-identifier; osen a
otherwise a RID value MUST be generated by the RID value, it <bcp14>MUST</bcp14> be used as the rid-identifier;
implementation. RID values MUST be generated in a fashion otherwise, a RID value <bcp14>MUST</bcp14> be generated by the
implementation. RID values <bcp14>MUST</bcp14> be generated in a f
ashion
that does not leak user information, e.g., randomly or that does not leak user information, e.g., randomly or
using a per-PeerConnection counter, and SHOULD be 3 bytes using a per-PeerConnection counter, and <bcp14>SHOULD</bcp14> be 3 bytes
or less, to allow them to efficiently fit into the RTP or less, to allow them to efficiently fit into the RTP
header extension defined in header extension defined in
<xref target="I-D.ietf-avtext-rid"></xref>, Section 3. If <xref target="RFC8852" sectionFormat="comma" section="3"/>.
<!-- [rfced] Section 5.2.1: Section 3 of RFC 8852 [I-D.ietf-avtext-rid] lists
several header extensions. Should "extension" in this sentence be
"extensions," or should one of the extension types be specified here?
Original:
RID values MUST be generated in a fashion that
does not leak user information, e.g., randomly or using a per-
PeerConnection counter, and SHOULD be 3 bytes or less, to allow
them to efficiently fit into the RTP header extension defined in
[I-D.ietf-avtext-rid], Section 3. -->
If
no encodings have been specified, or only one encoding is no encodings have been specified, or only one encoding is
specified but without a RID value, then no "a=rid" lines specified but without a RID value, then no "a=rid" lines
are generated.</t> are generated.</li>
<li>If the RtpTransceiver has a sendrecv or sendonly
<t>If the RtpTransceiver has a sendrecv or sendonly
direction and more than one "a=rid" line has been direction and more than one "a=rid" line has been
generated, an "a=simulcast" line, with direction "send", generated, an "a=simulcast" line, with direction "send",
as defined in as defined in
<xref target="I-D.ietf-mmusic-sdp-simulcast"></xref>, <xref target="RFC8853" sectionFormat="comma"
Section 6.2. The list of RIDs MUST include all of the RID section="6.2"/>. The list of RIDs <bcp14>MUST</bcp14>
identifiers used in the "a=rid" lines for this m= include all of the RID identifiers
section.</t> used in the "a=rid" lines for this "m="
section.</li>
<t>If the bundle policy for this PeerConnection is set to <li>If the bundle policy for this PeerConnection is set to
"max-bundle", and this is not the first m= section, or "max-bundle", and this is not the first "m=" section, or
the bundle policy is set to "balanced", and this is not the bundle policy is set to "balanced", and this is not
the first m= section for this media type, an the first "m=" section for this media type, an
"a=bundle-only" line.</t> "a=bundle-only" line.
</list>
</t>
<t>The following attributes, which are of category IDENTICAL <!-- [rfced] Section 5.2.1: We had trouble sorting out the "and"
or TRANSPORT, MUST appear only in "m=" sections which either ... "or" relationships here. If the suggested text is not correct,
have a unique address or which are associated with the please clarify.
bundle-tag. (In initial offers, this means those "m="
sections which do not contain an "a=bundle-only"
attribute.)</t>
<t> Original:
<list style="symbols"> o If the bundle policy for this PeerConnection is set to "max-
bundle", and this is not the first m= section, or the bundle
policy is set to "balanced", and this is not the first m= section
for this media type, an "a=bundle-only" line.
<t>"a=ice-ufrag" and "a=ice-pwd" lines, as specified in Suggested:
<xref target="I-D.ietf-mmusic-ice-sip-sdp"></xref>, o If (1) the bundle policy for this PeerConnection is set to "max-
Section 4.4.</t> bundle" and this is not the first "m=" section or (2) the bundle
policy is set to "balanced" and this is not the first "m=" section
for this media type, an "a=bundle-only" line. -->
<t>For each desired digest algorithm, one or more </li>
</ul>
<t>The following attributes, which are of category IDENTICAL
or TRANSPORT, <bcp14>MUST</bcp14> appear only in "m=" sections that ei
ther
have a unique address or are associated with the
BUNDLE-tag. (In initial offers, this means those "m="
sections that do not contain an "a=bundle-only"
attribute.)</t>
<ul spacing="normal">
<li>"a=ice-ufrag" and "a=ice-pwd" lines, as specified in
<xref target="RFC8839" sectionFormat="comma" section="5.4"/>.</li>
<li>For each desired digest algorithm, one or more
"a=fingerprint" lines for each of the endpoint's "a=fingerprint" lines for each of the endpoint's
certificates, as specified in certificates, as specified in
<xref target="RFC8122"></xref>, Section 5.</t> <xref target="RFC8122" sectionFormat="comma" section="5"/>.</li>
<li>An "a=setup" line, as specified in
<t>An "a=setup" line, as specified in <xref target="RFC4145" sectionFormat="comma" section="4"/> and cla
<xref target="RFC4145"></xref>, Section 4, and clarified rified
for use in DTLS-SRTP scenarios in for use in DTLS-SRTP scenarios in
<xref target="RFC5763"></xref>, Section 5. The role value <xref target="RFC5763" sectionFormat="comma" section="5"/>. The ro
in the offer MUST be "actpass".</t> le value
in the offer <bcp14>MUST</bcp14> be "actpass".</li>
<t>An "a=tls-id" line, as specified in <li>An "a=tls-id" line, as specified in
<xref target="I-D.ietf-mmusic-dtls-sdp" />, Section <xref target="RFC8842" sectionFormat="comma" section="5.2"/>.</li>
5.2.</t> <li>An "a=rtcp" line, as specified in
<xref target="RFC3605" sectionFormat="comma" section="2.1"/>, cont
<t>An "a=rtcp" line, as specified in aining
<xref target="RFC3605"></xref>, Section 2.1, containing
the dummy value "9 IN IP4 0.0.0.0", because no candidates the dummy value "9 IN IP4 0.0.0.0", because no candidates
have yet been gathered.</t> have yet been gathered.
<t>An "a=rtcp-mux" line, as specified in <!-- [rfced] Sections 5.2.1 and 5.3.1: Please confirm that
<xref target="RFC5761"></xref>, Section 5.1.3.</t> "9 IN IP4 0.0.0.0" (and not "IN IP4 0.0.0.0") is correct in these
two items.
<t>If the RTP/RTCP multiplexing policy is "require", an Original:
o An "a=rtcp" line, as specified in [RFC3605], Section 2.1,
containing the dummy value "9 IN IP4 0.0.0.0", because no
candidates have yet been gathered.
...
Otherwise, an "a=rtcp" line, as
specified in [RFC3605], Section 2.1, containing the dummy value "9
IN IP4 0.0.0.0" (because no candidates have yet been gathered). -->
</li>
<li>An "a=rtcp-mux" line, as specified in
<xref target="RFC5761" sectionFormat="comma" section="5.1.3"/>.</l
i>
<li>If the RTP/RTCP multiplexing policy is "require", an
"a=rtcp-mux-only" line, as specified in "a=rtcp-mux-only" line, as specified in
<xref target="I-D.ietf-mmusic-mux-exclusive" />, Section <xref target="RFC8858" sectionFormat="comma" section="4"/>.</li>
4.</t> <li>An "a=rtcp-rsize" line, as specified in
<xref target="RFC5506" sectionFormat="comma" section="5"/>.</li>
</ul>
<t>Lastly, if a data channel has been created, an "m=" section
<bcp14>MUST</bcp14> be generated for data. The &lt;media&gt; field <bc
p14>MUST</bcp14> be
set to "application", and the &lt;proto&gt; field <bcp14>MUST</bcp14>
be set
to "UDP/DTLS/SCTP"
<xref target="RFC8841" format="default"/>. The "fmt"
value <bcp14>MUST</bcp14> be set to "webrtc-datachannel" as specified
in
<xref target="RFC8841" sectionFormat="comma" section="4.1"/>.
<t>An "a=rtcp-rsize" line, as specified in <!-- [rfced] Section 5.2.1: We could not find any mention of
<xref target="RFC5506"></xref>, Section 5.</t> "webrtc-datachannel" in Section 4.1 of RFC 8841 [I-D.ietf-mmusic-sctp-sdp].
</list> Please confirm that this citation is correct and will be clear to
</t> readers.
<t>Lastly, if a data channel has been created, a m= section Original:
MUST be generated for data. The &lt;media&gt; field MUST be The "fmt" value MUST be set to "webrtc-
set to "application" and the &lt;proto&gt; field MUST be set datachannel" as specified in [I-D.ietf-mmusic-sctp-sdp], Section 4.1. -->
to "UDP/DTLS/SCTP"
<xref target="I-D.ietf-mmusic-sctp-sdp"></xref>. The "fmt"
value MUST be set to "webrtc-datachannel" as specified in
<xref target="I-D.ietf-mmusic-sctp-sdp"></xref>, Section
4.1.</t>
<t>Within the data m= section, an "a=mid" line MUST be </t>
<t>Within the data "m=" section, an "a=mid" line <bcp14>MUST</bcp14> b
e
generated and included as described above, along with an generated and included as described above, along with an
"a=sctp-port" line referencing the SCTP port number, as "a=sctp-port" line referencing the SCTP port number, as
defined in defined in
<xref target="I-D.ietf-mmusic-sctp-sdp"></xref>, Section 5.1, <xref target="RFC8841" sectionFormat="comma" section="5.1"/>;
and, if appropriate, an "a=max-message-size" line, as defined and, if appropriate, an "a=max-message-size" line, as defined
in in
<xref target="I-D.ietf-mmusic-sctp-sdp"></xref>, Section <xref target="RFC8841" sectionFormat="comma" section="6.1"/>.</t>
6.1.</t>
<t>As discussed above, the following attributes of category <t>As discussed above, the following attributes of category
IDENTICAL or TRANSPORT are included only if the data m= IDENTICAL or TRANSPORT are included only if the data "m="
section either has a unique address or is associated with the section either has a unique address or is associated with the
bundle-tag (e.g., if it is the only m= section): BUNDLE-tag (e.g., if it is the only "m=" section):
<list style="symbols"> </t>
<ul spacing="normal">
<t>"a=ice-ufrag"</t> <li>"a=ice-ufrag"</li>
<li>"a=ice-pwd"</li>
<t>"a=ice-pwd"</t> <li>"a=fingerprint"</li>
<li>"a=setup"</li>
<t>"a=fingerprint"</t> <li>"a=tls-id"</li>
</ul>
<t>"a=setup"</t> <t>Once all "m=" sections have been generated, a session-level
"a=group" attribute <bcp14>MUST</bcp14> be added as specified in
<t>"a=tls-id"</t> <xref target="RFC5888" format="default"/>. This attribute <bcp14>MUST<
</list></t> /bcp14> have
semantics "BUNDLE" and <bcp14>MUST</bcp14> include the mid identifiers
<t>Once all m= sections have been generated, a session-level of
"a=group" attribute MUST be added as specified in each "m=" section. The effect of this is that the JSEP
<xref target="RFC5888"></xref>. This attribute MUST have implementation offers all "m=" sections as one bundle group.
semantics "BUNDLE", and MUST include the mid identifiers of However, whether the "m=" sections are bundle-only or not
each m= section. The effect of this is that the JSEP
implementation offers all m= sections as one bundle group.
However, whether the m= sections are bundle-only or not
depends on the bundle policy.</t> depends on the bundle policy.</t>
<t>The next step is to generate session-level lip sync groups <t>The next step is to generate session-level lip sync groups
as defined in as defined in
<xref target="RFC5888" />, Section 7. For each MediaStream <xref target="RFC5888" sectionFormat="comma" section="7"/>. For each M ediaStream
referenced by more than one RtpTransceiver (by passing those referenced by more than one RtpTransceiver (by passing those
MediaStreams as arguments to the addTrack and addTransceiver MediaStreams as arguments to the addTrack and addTransceiver
methods), a group of type "LS" MUST be added that contains methods), a group of type "LS" <bcp14>MUST</bcp14> be added that conta ins
the mid values for each RtpTransceiver.</t> the mid values for each RtpTransceiver.</t>
<t>Attributes that SDP permits to be at either the session
<t>Attributes which SDP permits to either be at the session level or the media level <bcp14>SHOULD</bcp14> generally be at the med
level or the media level SHOULD generally be at the media ia
level even if they are identical. This assists development level even if they are identical. This assists development
and debugging by making it easier to understand individual and debugging by making it easier to understand individual
media sections, especially if one of a set of initially media sections, especially if one of a set of initially
identical attributes is subsequently changed. However, identical attributes is subsequently changed. However,
implementations MAY choose to aggregate attributes at the implementations <bcp14>MAY</bcp14> choose to aggregate attributes at t
session level and JSEP implementations MUST be prepared to he
session level, and JSEP implementations <bcp14>MUST</bcp14> be prepare
d to
receive attributes in either location.</t> receive attributes in either location.</t>
<t>Attributes other than the ones specified above <bcp14>MAY</bcp14> b
<t>Attributes other than the ones specified above MAY be e
included, except for the following attributes which are included, except for the following attributes, which are
specifically incompatible with the requirements of specifically incompatible with the requirements of
<xref target="I-D.ietf-rtcweb-rtp-usage"></xref>, and MUST <xref target="RFC8834" format="default"/> and <bcp14>MUST
NOT be included: NOT</bcp14> be included:
<list style="symbols"> </t>
<ul spacing="normal">
<t>"a=crypto"</t> <li>"a=crypto"</li>
<li>"a=key-mgmt"</li>
<t>"a=key-mgmt"</t> <li>"a=ice-lite"</li>
</ul>
<t>"a=ice-lite"</t>
</list></t>
<t>Note that when bundle is used, any additional attributes <t>Note that when bundle is used, any additional attributes
that are added MUST follow the advice in that are added <bcp14>MUST</bcp14> follow the advice in
<xref target="I-D.ietf-mmusic-sdp-mux-attributes"></xref> on <xref target="RFC8859" format="default"/> on
how those attributes interact with bundle.</t> how those attributes interact with bundle.</t>
<t>Note that these requirements are in some cases stricter <t>Note that these requirements are in some cases stricter
than those of SDP. Implementations MUST be prepared to accept than those of SDP. Implementations <bcp14>MUST</bcp14> be prepared to accept
compliant SDP even if it would not conform to the compliant SDP even if it would not conform to the
requirements for generating SDP in this specification.</t> requirements for generating SDP in this specification.</t>
</section> </section>
<section title="Subsequent Offers" <section anchor="sec.subsequent-offers" numbered="true" toc="default">
anchor="sec.subsequent-offers"> <name>Subsequent Offers</name>
<t>When createOffer is called a second (or later) time or is
<t>When createOffer is called a second (or later) time, or is
called after a local description has already been installed, called after a local description has already been installed,
the processing is somewhat different than for an initial the processing is somewhat different than for an initial
offer.</t> offer.</t>
<t>If the previous offer was not applied using <t>If the previous offer was not applied using
setLocalDescription, meaning the PeerConnection is still in setLocalDescription, meaning the PeerConnection is still in
the "stable" state, the steps for generating an initial offer the "stable" state, the steps for generating an initial offer
should be followed, subject to the following restriction: should be followed, subject to the following restriction:
<list style="symbols"> </t>
<ul spacing="normal">
<t>The fields of the "o=" line MUST stay the same except <li>The fields of the "o=" line <bcp14>MUST</bcp14> stay the same ex
for the &lt;session-version&gt; field, which MUST increment cept
for the &lt;session-version&gt; field, which <bcp14>MUST</bcp14> inc
rement
by one on each call to createOffer if the offer might by one on each call to createOffer if the offer might
differ from the output of the previous call to createOffer; differ from the output of the previous call to createOffer;
implementations MAY opt to increment implementations <bcp14>MAY</bcp14> opt to increment
&lt;session-version&gt; on every call. The value of the &lt;session-version&gt; on every call. The value of the
generated &lt;session-version&gt; is independent of the generated &lt;session-version&gt; is independent of the
&lt;session-version&gt; of the current local description; &lt;session-version&gt; of the current local description;
in particular, in the case where the current version is N, in particular, in the case where the current version is N,
an offer is created and applied with version N+1, and then an offer is created and applied with version N+1, and then
that offer is rolled back so that the current version is that offer is rolled back so that the current version is
again N, the next generated offer will still have version again N, the next generated offer will still have version
N+2.</t> N+2.</li>
</list></t> </ul>
<t>Note that if the application creates an offer by reading <t>Note that if the application creates an offer by reading
currentLocalDescription instead of calling createOffer, the currentLocalDescription instead of calling createOffer, the
returned SDP may be different than when setLocalDescription returned SDP may be different than when setLocalDescription
was originally called, due to the addition of gathered ICE was originally called, due to the addition of gathered ICE
candidates, but the &lt;session-version&gt; will not have candidates, but the &lt;session-version&gt; will not have
changed. There are no known scenarios in which this causes changed. There are no known scenarios in which this causes
problems, but if this is a concern, the solution is simply to problems, but if this is a concern, the solution is simply to
use createOffer to ensure a unique use createOffer to ensure a unique
&lt;session-version&gt;.</t> &lt;session-version&gt;.</t>
<t>If the previous offer was applied using <t>If the previous offer was applied using
setLocalDescription, but a corresponding answer from the setLocalDescription, but a corresponding answer from the
remote side has not yet been applied, meaning the remote side has not yet been applied, meaning the
PeerConnection is still in the "have-local-offer" state, an PeerConnection is still in the "have-local-offer" state, an
offer is generated by following the steps in the "stable" offer is generated by following the steps in the "stable"
state above, along with these exceptions: state above, along with these exceptions:
<list style="symbols"> </t>
<ul spacing="normal">
<t>The "s=" and "t=" lines MUST stay the same.</t> <li>The "s=" and "t=" lines <bcp14>MUST</bcp14> stay the same.</li>
<li>If any RtpTransceiver has been added and there exists
<t>If any RtpTransceiver has been added, and there exists an "m=" section with a zero port in the current local
an m= section with a zero port in the current local description or the current remote description, that "m="
description or the current remote description, that m= section <bcp14>MUST</bcp14> be recycled by generating an "m=" sectio
section MUST be recycled by generating an m= section for n for
the added RtpTransceiver as if the m= section were being the added RtpTransceiver as if the "m=" section were being
added to the session description (including a new MID added to the session description (including a new MID
value), and placing it at the same index as the m= section value) and placing it at the same index as the "m=" section
with a zero port.</t> with a zero port.</li>
<li>If an RtpTransceiver is stopped and is not associated
<t>If an RtpTransceiver is stopped and is not associated with an "m=" section, an "m=" section <bcp14>MUST NOT</bcp14> be gen
with an m= section, an m= section MUST NOT be generated for erated for
it. This prevents adding back RtpTransceivers whose m= it. This prevents adding back RtpTransceivers whose "m="
sections were recycled and used for a new RtpTransceiver in sections were recycled and used for a new RtpTransceiver in
a previous offer/ answer exchange, as described above.</t> a previous offer/ answer exchange, as described above.</li>
<li>If an RtpTransceiver has been stopped and is associated
<t>If an RtpTransceiver has been stopped and is associated with an "m=" section, and the "m=" section is not being
with an m= section, and the m= section is not being recycled as described above, an "m=" section <bcp14>MUST</bcp14> be
recycled as described above, an m= section MUST be
generated for it with the port set to zero and all "a=msid" generated for it with the port set to zero and all "a=msid"
lines removed.</t> lines removed.</li>
<li>For RtpTransceivers that are not stopped, the "a=msid"
<t>For RtpTransceivers that are not stopped, the "a=msid" line or lines <bcp14>MUST</bcp14> stay the same if they are present
line(s) MUST stay the same if they are present in the in the
current description, regardless of changes to the current description, regardless of changes to the
transceiver's direction or track. If no "a=msid" line is transceiver's direction or track. If no "a=msid" line is
present in the current description, "a=msid" line(s) MUST present in the current description, "a=msid" line(s) <bcp14>MUST</bc p14>
be generated according to the same rules as for an initial be generated according to the same rules as for an initial
offer.</t> offer.</li>
<li>Each "m=" and "c=" line <bcp14>MUST</bcp14> be filled in with th
<t>Each "m=" and c=" line MUST be filled in with the port, e port,
relevant RTP profile, and address of the default candidate for the relevant RTP profile, and address of the default candidate for the
m= section, as described in "m=" section, as described in
<xref target="I-D.ietf-mmusic-ice-sip-sdp"></xref>, <xref target="RFC8839" sectionFormat="comma" section="4.2.1.2"/> and
Section 3.2.1.2, and clarified in clarified in
<xref target="sec.profile-names"/>. <xref target="sec.profile-names" format="default"/>.
If no RTP candidates have yet been gathered, dummy If no RTP candidates have yet been gathered, dummy
values MUST still be used, as described above.</t> values <bcp14>MUST</bcp14> still be used, as described above.</li>
<li>Each "a=mid" line <bcp14>MUST</bcp14> stay the same.</li>
<li>Each "a=ice-ufrag" and "a=ice-pwd" line <bcp14>MUST</bcp14> stay
the
same, unless the ICE configuration has changed (e.g., changes to
either the supported STUN/TURN servers or the ICE
candidate policy) or the "IceRestart" option
(<xref target="sec.icerestart" format="default"/>) was specified.
<t>Each "a=mid" line MUST stay the same.</t> <!-- [rfced] Section 5.2.2: As it appears that "either changes to"
means "changes to either," we updated this sentence accordingly.
Please let us know if this is incorrect.
<t>Each "a=ice-ufrag" and "a=ice-pwd" line MUST stay the Original (the parentheses around "Section 5.2.3.1" have been fixed):
same, unless the ICE configuration has changed (either o Each "a=ice-ufrag" and "a=ice-pwd" line MUST stay the same, unless
changes to the supported STUN/TURN servers, or the ICE the ICE configuration has changed (either changes to the supported
candidate policy), or the "IceRestart" option ( STUN/TURN servers, or the ICE candidate policy), or the
<xref target="sec.icerestart" /> was specified. If the m= "IceRestart" option ( Section 5.2.3.1 was specified.
section is bundled into another m= section, it still MUST
NOT contain any ICE credentials.</t>
<t>If the m= section is not bundled into another m= Currently:
section, its "a=rtcp" attribute line MUST be filled in with * Each "a=ice-ufrag" and "a=ice-pwd" line MUST stay the same, unless
the ICE configuration has changed (e.g., changes to either the
supported STUN/TURN servers or the ICE candidate policy) or the
"IceRestart" option (Section 5.2.3.1) was specified. -->
If the "m="
section is bundled into another "m=" section, it still <bcp14>MUST
NOT</bcp14> contain any ICE credentials.</li>
<li>If the "m=" section is not bundled into another "m="
section, its "a=rtcp" attribute line <bcp14>MUST</bcp14> be filled i
n with
the port and address of the default RTCP candidate, as the port and address of the default RTCP candidate, as
indicated in indicated in
<xref target="RFC5761"></xref>, Section 5.1.3. If no RTCP <xref target="RFC5761" sectionFormat="comma" section="5.1.3"/>. If n
candidates have yet been gathered, dummy values MUST be o RTCP
used, as described in the initial offer section above.</t> candidates have yet been gathered, dummy values <bcp14>MUST</bcp14>
be
used, as described in <xref target="sec.initial-offers"/> above.
<t>If the m= section is not bundled into another m= <!-- [rfced] Sections 5.2.2, 5.3.1, and 5.3.2: We changed "the
initial offer section" and "the initial offers section" to
"Section 5.2.1," and we changed "the initial answer section" to
"Section 5.3.1." Please let us know if these changes are incorrect.
Original:
If no RTCP candidates have yet been gathered,
dummy values MUST be used, as described in the initial offer
section above.
...
The process here is identical to that
indicated in the initial offers section above, except that the
"a=ice-options" line, with the "trickle" option as specified in
[I-D.ietf-ice-trickle], Section 3, and the "ice2" option as specified
in [RFC8445], Section 10, is only included if such an option was
present in the offer.
...
If no RTCP candidates have yet been gathered, dummy values MUST be
used, as described in the initial answer section above.
Currently:
If no RTCP candidates have yet been gathered,
dummy values MUST be used, as described in Section 5.2.1 above.
...
The process here is identical to that
indicated in Section 5.2.1 above, except that the "a=ice-options"
line, with the "trickle" option as specified in [RFC8840],
Section 4.1.3 and the "ice2" option as specified in [RFC8445],
Section 10, is only included if such an option was present in the
offer.
...
If no RTCP candidates have yet been gathered, dummy values MUST be
used, as described in Section 5.3.1 above. -->
</li>
<li>If the "m=" section is not bundled into another "m="
section, for each candidate that has been gathered during section, for each candidate that has been gathered during
the most recent gathering phase (see the most recent gathering phase (see
<xref target="sec.ice-gather-overview"></xref>), an <xref target="sec.ice-gather-overview" format="default"/>), an
"a=candidate" line MUST be added, as defined in "a=candidate" line <bcp14>MUST</bcp14> be added, as defined in
<xref target="I-D.ietf-mmusic-ice-sip-sdp"></xref>, Section 4.1. <xref target="RFC8839" sectionFormat="comma" section="5.1"/>.
If candidate gathering for the section has completed, an If candidate gathering for the section has completed, an
"a=end-of-candidates" attribute MUST be added, as described "a=end-of-candidates" attribute <bcp14>MUST</bcp14> be added, as des cribed
in in
<xref target="I-D.ietf-ice-trickle"></xref>, Section 9.3. <xref target="RFC8840" sectionFormat="comma" section="8.2"/>.
If the m= section is bundled into another m= section, both If the "m=" section is bundled into another "m=" section, both
"a=candidate" and "a=end-of-candidates" MUST be "a=candidate" and "a=end-of-candidates" <bcp14>MUST</bcp14> be
omitted.</t> omitted.
<t>For RtpTransceivers that are still present, the "a=rid" <!-- [rfced] Section 5.2.2: We found this RFC Editor Note on
lines MUST stay the same.</t> <https://datatracker.ietf.org/doc/draft-ietf-rtcweb-jsep/writeup/>:
<t>For RtpTransceivers that are still present, any "OLD:
"a=simulcast" line MUST stay the same.</t> o If the m= section is not bundled into another m= section, for each
</list></t> candidate that has been gathered during the most recent gathering
phase (see Section 3.5.1), an "a=candidate" line MUST be added, as
defined in [RFC5245], Section 4.3., paragraph 3. If candidate
gathering for the section has completed, an "a=end-of-candidates"
attribute MUST be added, as described in [I-D.ietf-ice-trickle],
Section 9.3. If the m= section is bundled into another m=
section, both "a=candidate" and "a=end-of-candidates" MUST be
omitted.
NEW:
o If the m= section is not bundled into another m= section, for each
candidate that has been gathered during the most recent gathering
phase (see Section 3.5.1), an "a=candidate" line MUST be added, as
defined in [RFC5245], Section 4.3., paragraph 3. If candidate
gathering for the section has completed, an "a=end-of-candidates"
attribute MUST be added, as described in
[I-D.ietf-mmusic-trickle-ice-sip], Section 8.2. If the m= section is
bundled into another m= section, both "a=candidate" and
"a=end-of-candidates" MUST be omitted."
Please note that the "OLD" text does not match what we found in the
provided draft (i.e., "[RFC5245], Section 4.3., paragraph 3" versus
"[I-D.ietf-mmusic-ice-sip-sdp], Section 4.1"):
o If the m= section is not bundled into another m= section, for each
candidate that has been gathered during the most recent gathering
phase (see Section 3.5.1), an "a=candidate" line MUST be added, as
defined in [I-D.ietf-mmusic-ice-sip-sdp], Section 4.1. If
candidate gathering for the section has completed, an "a=end-of-
candidates" attribute MUST be added, as described in
[I-D.ietf-ice-trickle], Section 9.3. If the m= section is bundled
into another m= section, both "a=candidate" and "a=end-of-
candidates" MUST be omitted.
Please review, and let us know if further changes are needed.
(Note: "[RFC8839]" and "[RFC8840]" are the RFC numbers assigned to
[I-D.ietf-mmusic-ice-sip-sdp] and [I-D.ietf-mmusic-trickle-ice-sip],
respectively.)
Currently:
* If the "m=" section is not bundled into another "m=" section, for each
candidate that has been gathered during the most recent gathering
phase (see Section 3.5.1), an "a=candidate" line MUST be added, as
defined in [RFC8839], Section 5.1. If candidate gathering for the
section has completed, an "a=end-of-candidates" attribute MUST be
added, as described in [RFC8840], Section 8.2. If the "m=" section
is bundled into another "m=" section, both "a=candidate" and "a=end-
of-candidates" MUST be omitted. -->
</li>
<li>For RtpTransceivers that are still present, the "a=rid"
lines <bcp14>MUST</bcp14> stay the same.</li>
<li>For RtpTransceivers that are still present, any
"a=simulcast" line <bcp14>MUST</bcp14> stay the same.</li>
</ul>
<t>If the previous offer was applied using <t>If the previous offer was applied using
setLocalDescription, and a corresponding answer from the setLocalDescription, and a corresponding answer from the
remote side has been applied using setRemoteDescription, remote side has been applied using setRemoteDescription,
meaning the PeerConnection is in the "have-remote-pranswer" meaning the PeerConnection is in the "have-remote-pranswer"
or "stable" states, an offer is generated based on the state or the "stable" state, an offer is generated based on the
negotiated session descriptions by following the steps negotiated session descriptions by following the steps
mentioned for the "have-local-offer" state above.</t> mentioned for the "have-local-offer" state above.</t>
<t>In addition, for each existing, non-recycled, non-rejected <t>In addition, for each existing, non-recycled, non-rejected
m= section in the new offer, the following adjustments are "m=" section in the new offer, the following adjustments are
made based on the contents of the corresponding m= section in made based on the contents of the corresponding "m=" section in
the current local or remote description, as appropriate: the current local or remote description, as appropriate:
<list style="symbols"> </t>
<ul spacing="normal">
<t>The m= line and corresponding "a=rtpmap" and "a=fmtp" <li>The "m=" line and corresponding "a=rtpmap" and "a=fmtp"
lines MUST only include media formats which have not been lines <bcp14>MUST</bcp14> only include media formats that have not b
een
excluded by the codec preferences of the associated excluded by the codec preferences of the associated
transceiver, and MUST include all currently available transceiver and also <bcp14>MUST</bcp14> include all currently avail able
formats. Media formats that were previously offered but are formats. Media formats that were previously offered but are
no longer available (e.g., a shared hardware codec) MAY be no longer available (e.g., a shared hardware codec) <bcp14>MAY</bcp1
excluded.</t> 4> be
excluded.</li>
<t>Unless codec preferences have been set for the <li>Unless codec preferences have been set for the
associated transceiver, the media formats on the m= line associated transceiver, the media formats on the "m=" line
MUST be generated in the same order as in the most recent <bcp14>MUST</bcp14> be generated in the same order as in the most re
cent
answer. Any media formats that were not present in the most answer. Any media formats that were not present in the most
recent answer MUST be added after all existing formats.</t> recent answer <bcp14>MUST</bcp14> be added after all existing format
s.</li>
<t>The RTP header extensions MUST only include those that <li>The RTP header extensions <bcp14>MUST</bcp14> only include those
are present in the most recent answer.</t> that
are present in the most recent answer.</li>
<t>The RTCP feedback mechanisms MUST only include those <li>The RTCP feedback mechanisms <bcp14>MUST</bcp14> only include th
ose
that are present in the most recent answer, except for the that are present in the most recent answer, except for the
case of format-specific mechanisms that are referencing a case of format-specific mechanisms that are referencing a
newly-added media format.</t> newly added media format.</li>
<li>The "a=rtcp" line <bcp14>MUST NOT</bcp14> be added if the most r
<t>The "a=rtcp" line MUST NOT be added if the most recent ecent
answer included an "a=rtcp-mux" line.</t> answer included an "a=rtcp-mux" line.</li>
<li>The "a=rtcp-mux" line <bcp14>MUST</bcp14> be the same as that in
the
most recent answer.</li>
<li>The "a=rtcp-mux-only" line <bcp14>MUST NOT</bcp14> be added.</li
>
<li>The "a=rtcp-rsize" line <bcp14>MUST NOT</bcp14> be added unless
present
in the most recent answer.</li>
<li>An "a=bundle-only" line <bcp14>MUST NOT</bcp14> be added, as ind
icated
in
<xref target="RFC8843" sectionFormat="comma" section="6"/>.
<t>The "a=rtcp-mux" line MUST be the same as that in the <!-- [rfced] Section 5.2.2: We could not find any indication in
most recent answer.</t> RFC 8843 [I-D.ietf-mmusic-sdp-bundle-negotiation], Section 6 that an
"a=bundle-only" line MUST NOT be added. Will this be clear to
readers?
<t>The "a=rtcp-mux-only" line MUST NOT be added.</t> Original:
o An "a=bundle-only" line MUST NOT be added, as indicated in
[I-D.ietf-mmusic-sdp-bundle-negotiation], Section 6.
<t>The "a=rtcp-rsize" line MUST NOT be added unless present Possibly:
in the most recent answer.</t> * An "a=bundle-only" line, as described in [RFC8843], Section 6,
MUST NOT be added. -->
<t>An "a=bundle-only" line MUST NOT be added, as indicated Instead, JSEP implementations <bcp14>MUST</bcp14> simply omit
in
<xref target="I-D.ietf-mmusic-sdp-bundle-negotiation" />,
Section 6. Instead, JSEP implementations MUST simply omit
parameters in the IDENTICAL and TRANSPORT categories for parameters in the IDENTICAL and TRANSPORT categories for
bundled m= sections, as described in bundled "m=" sections, as described in
<xref target="I-D.ietf-mmusic-sdp-bundle-negotiation" />, <xref target="RFC8843" sectionFormat="comma" section="8.1"/>.</li>
Section 8.1.</t> <li>Note that if media "m=" sections are bundled into a data
"m=" section, then certain TRANSPORT and IDENTICAL attributes
<t>Note that if media m= sections are bundled into a data may appear in the data "m=" section even if they would
m= section, then certain TRANSPORT and IDENTICAL attributes otherwise only be appropriate for a media "m=" section (e.g.,
may appear in the data m= section even if they would
otherwise only be appropriate for a media m= section (e.g.,
"a=rtcp-mux"). This cannot happen in initial offers because "a=rtcp-mux"). This cannot happen in initial offers because
in the initial offer JSEP implementations always list media in the initial offer JSEP implementations always list media
m= sections (if any) before the data m= section (if any), "m=" sections (if any) before the data "m=" section (if any),
and at least one of those media m= sections will not have and at least one of those media "m=" sections will not have
the "a=bundle-only" attribute. Therefore, in initial the "a=bundle-only" attribute. Therefore, in initial
offers, any "a=bundle-only" m= sections will be bundled offers, any "a=bundle-only" "m=" sections will be bundled
into a preceding non-bundle-only media m= section.</t> into a preceding non-bundle-only media "m=" section.</li>
</list></t> </ul>
<t>The "a=group:BUNDLE" attribute <bcp14>MUST</bcp14> include the MID
<t>The "a=group:BUNDLE" attribute MUST include the MID
identifiers specified in the bundle group in the most recent identifiers specified in the bundle group in the most recent
answer, minus any m= sections that have been marked as answer, minus any "m=" sections that have been marked as
rejected, plus any newly added or re-enabled m= sections. In rejected, plus any newly added or re-enabled "m=" sections. In
other words, the bundle attribute must contain all m= other words, the bundle attribute must contain all "m="
sections that were previously bundled, as long as they are sections that were previously bundled, as long as they are
still alive, as well as any new m= sections.</t> still alive, as well as any new "m=" sections.</t>
<t>"a=group:LS" attributes are generated in the same way as <t>"a=group:LS" attributes are generated in the same way as
for initial offers, with the additional stipulation that any for initial offers, with the additional stipulation that any
lip sync groups that were present in the most recent answer lip sync groups that were present in the most recent answer
MUST continue to exist and MUST contain any previously <bcp14>MUST</bcp14> continue to exist and <bcp14>MUST</bcp14> contain
existing MID identifiers, as long as the identified m= any previously
existing MID identifiers, as long as the identified "m="
sections still exist and are not rejected, and the group sections still exist and are not rejected, and the group
still contains at least two MID identifiers. This ensures still contains at least two MID identifiers. This ensures
that any synchronized "recvonly" m= sections continue to be that any synchronized "recvonly" "m=" sections continue to be
synchronized in the new offer.</t> synchronized in the new offer.</t>
</section> </section>
<section title="Options Handling" <section anchor="sec.options-handling1" numbered="true" toc="default">
anchor="sec.options-handling1"> <name>Options Handling</name>
<t>The createOffer method takes as a parameter an <t>The createOffer method takes as a parameter an
RTCOfferOptions object. Special processing is performed when RTCOfferOptions object. Special processing is performed when
generating a SDP description if the following options are generating an SDP description if the following options are
present.</t> present.</t>
<section title="IceRestart" anchor="sec.icerestart"> <section anchor="sec.icerestart" numbered="true" toc="default">
<name>IceRestart</name>
<t>If the "IceRestart" option is specified, with a value of <t>If the "IceRestart" option is specified, with a value of
"true", the offer MUST indicate an ICE restart by "true", the offer <bcp14>MUST</bcp14> indicate an ICE restart by
generating new ICE ufrag and pwd attributes, as specified generating new ICE ufrag and pwd attributes, as specified
in in
<xref target="I-D.ietf-mmusic-ice-sip-sdp"></xref>, <xref target="RFC8839" sectionFormat="comma" section="4.4.3.1.1"/>.
Section 3.4.1.1.1. If this If this
option is specified on an initial offer, it has no effect option is specified on an initial offer, it has no effect
(since a new ICE ufrag and pwd are already generated). (since a new ICE ufrag and pwd are already generated).
Similarly, if the ICE configuration has changed, this Similarly, if the ICE configuration has changed, this
option has no effect, since new ufrag and pwd attributes option has no effect, since new ufrag and pwd attributes
will be generated automatically. This option is primarily will be generated automatically. This option is primarily
useful for reestablishing connectivity in cases where useful for reestablishing connectivity in cases where
failures are detected by the application.</t> failures are detected by the application.</t>
</section> </section>
<section title="VoiceActivityDetection" <section anchor="sec.voiceactivitydetection1" numbered="true" toc="def
anchor="sec.voiceactivitydetection1"> ault">
<name>VoiceActivityDetection</name>
<t>Silence suppression, also known as discontinuous <t>Silence suppression, also known as discontinuous
transmission ("DTX"), can reduce the bandwidth used for transmission ("DTX"), can reduce the bandwidth used for
audio by switching to a special encoding when voice audio by switching to a special encoding when voice
activity is not detected, at the cost of some fidelity.</t> activity is not detected, at the cost of some fidelity.</t>
<t>If the "VoiceActivityDetection" option is specified, <t>If the "VoiceActivityDetection" option is specified,
with a value of "true", the offer MUST indicate support for with a value of "true", the offer <bcp14>MUST</bcp14> indicate suppo rt for
silence suppression in the audio it receives by including silence suppression in the audio it receives by including
comfort noise ("CN") codecs for each offered audio codec, comfort noise ("CN") codecs for each offered audio codec,
as specified in as specified in
<xref target="RFC3389"></xref>, Section 5.1, except for <xref target="RFC3389" sectionFormat="comma" section="5.1"/>, except for
codecs that have their own internal silence suppression codecs that have their own internal silence suppression
support. For codecs that have their own internal silence support. For codecs that have their own internal silence
suppression support, the appropriate fmtp parameters for suppression support, the appropriate fmtp parameters for
that codec MUST be specified to indicate that silence that codec <bcp14>MUST</bcp14> be specified to indicate that silence
suppression for received audio is desired. For example, suppression for received audio is desired. For example,
when using the Opus codec when using the Opus codec
<xref target="RFC6716" />, the "usedtx=1" parameter, <xref target="RFC6716" format="default"/>, the "usedtx=1" parameter,
specified in specified in
<xref target="RFC7587" />, would be used in the offer.</t> <xref target="RFC7587" format="default"/>, would be used in the offe
r.</t>
<t>If the "VoiceActivityDetection" option is specified, <t>If the "VoiceActivityDetection" option is specified,
with a value of "false", the JSEP implementation MUST NOT with a value of "false", the JSEP implementation <bcp14>MUST NOT</bc p14>
emit "CN" codecs. For codecs that have their own internal emit "CN" codecs. For codecs that have their own internal
silence suppression support, the appropriate fmtp silence suppression support, the appropriate fmtp
parameters for that codec MUST be specified to indicate parameters for that codec <bcp14>MUST</bcp14> be specified to indica te
that silence suppression for received audio is not desired. that silence suppression for received audio is not desired.
For example, when using the Opus codec, the "usedtx=0" For example, when using the Opus codec, the "usedtx=0"
parameter would be specified in the offer. In addition, the parameter would be specified in the offer. In addition, the
implementation MUST NOT use silence suppression for media implementation <bcp14>MUST NOT</bcp14> use silence suppression for m edia
it generates, regardless of whether the "CN" codecs or it generates, regardless of whether the "CN" codecs or
related fmtp parameters appear in the peer's description. related fmtp parameters appear in the peer's description.
The impact of these rules is that silence suppression in The impact of these rules is that silence suppression in
JSEP depends on mutual agreement of both sides, which JSEP depends on mutual agreement of both sides, which
ensures consistent handling regardless of which codec is ensures consistent handling regardless of which codec is
used.</t> used.</t>
<t>The "VoiceActivityDetection" option does not have any <t>The "VoiceActivityDetection" option does not have any
impact on the setting of the "vad" value in the signaling impact on the setting of the "vad" value in the signaling
of the client to mixer audio level header extension of the client-to-mixer audio level header extension
described in described in
<xref target="RFC6464"></xref>, Section 4.</t> <xref target="RFC6464" sectionFormat="comma" section="4"/>.</t>
</section> </section>
</section> </section>
</section> </section>
<section title="Generating an Answer" <section anchor="sec.generating-an-answer" numbered="true" toc="default">
anchor="sec.generating-an-answer"> <name>Generating an Answer</name>
<t>When createAnswer is called, a new SDP description must be <t>When createAnswer is called, a new SDP description must be
created that is compatible with the supplied remote description created that is compatible with the supplied remote description
as well as the requirements specified in as well as the requirements specified in
<xref target="I-D.ietf-rtcweb-rtp-usage"></xref>. The exact <xref target="RFC8834" format="default"/>. The exact
details of this process are explained below.</t> details of this process are explained below.</t>
<section title="Initial Answers" anchor="sec.initial-answers"> <section anchor="sec.initial-answers" numbered="true" toc="default">
<name>Initial Answers</name>
<t>When createAnswer is called for the first time after a <t>When createAnswer is called for the first time after a
remote description has been provided, the result is known as remote description has been provided, the result is known as
the initial answer. If no remote description has been the initial answer. If no remote description has been
installed, an answer cannot be generated, and an error MUST installed, an answer cannot be generated, and an error <bcp14>MUST</bc p14>
be returned.</t> be returned.</t>
<t>Note that the remote description SDP may not have been <t>Note that the remote description SDP may not have been
created by a JSEP endpoint and may not conform to all the created by a JSEP endpoint and may not conform to all the
requirements listed in requirements listed in
<xref target="sec-create-offer"></xref>. For many cases, this <xref target="sec-create-offer" format="default"/>. For many cases, th is
is not a problem. However, if any mandatory SDP attributes is not a problem. However, if any mandatory SDP attributes
are missing, or functionality listed as mandatory-to-use are missing or functionality listed as mandatory-to-use
above is not present, this MUST be treated as an error, and above is not present, this <bcp14>MUST</bcp14> be treated as an error
MUST cause the affected m= sections to be marked as and
<bcp14>MUST</bcp14> cause the affected "m=" sections to be marked as
rejected.</t> rejected.</t>
<t>The first step in generating an initial answer is to <t>The first step in generating an initial answer is to
generate session-level attributes. The process here is generate session-level attributes. The process here is
identical to that indicated in the initial offers section identical to that indicated in <xref target="sec.initial-offers"/> abo
above, except that the "a=ice-options" line, with the ve, except that the "a=ice-options" line, with the
"trickle" option as specified in "trickle" option as specified in
<xref target="I-D.ietf-ice-trickle"></xref>, Section 3, <xref target="RFC8840" sectionFormat="comma" section="4.1.3"/>
and the "ice2" option as specified in and the "ice2" option as specified in
<xref target="RFC8445"></xref>, Section 10, is <xref target="RFC8445" sectionFormat="comma" section="10"/>, is
only included if such an option was present in the offer.</t> only included if such an option was present in the offer.
<!-- [rfced] Section 5.3.1: We found this RFC Editor Note on
<https://datatracker.ietf.org/doc/draft-ietf-rtcweb-jsep/writeup/>:
"OLD:
The first step in generating an initial answer is to generate
session-level attributes. The process here is identical to that
indicated in the initial offers section above, except that the
"a=ice-options" line, with the "trickle" option as specified in
[I-D.ietf-ice-trickle], Section 4, is only included if such an option
was present in the offer.
NEW:
The first step in generating an initial answer is to generate
session-level attributes. The process here is identical to that
indicated in the initial offers section above, except that the
"a=ice-options" line, with the "trickle" option as specified in
[I-D.ietf-mmusic-trickle-ice-sip], Section 4.1.3, is only included
if such an option was present in the offer."
Please note that the "OLD" text does not match what we found in the
provided draft:
The first step in generating an initial answer is to generate
session-level attributes. The process here is identical to that
indicated in the initial offers section above, except that the
"a=ice-options" line, with the "trickle" option as specified in
[I-D.ietf-ice-trickle], Section 3, and the "ice2" option as specified
in [RFC8445], Section 10, is only included if such an option was
present in the offer.
We updated as follows. As noted previously, (1) for ease of the
reader and (2) to create a usable hyperlink in the .html file, we
changed "the initial offers section above" to "Section 5.2.1 above."
Currently:
The first step in generating an initial answer is to generate
session-level attributes. The process here is identical to that
indicated in Section 5.2.1 above, except that the "a=ice-options"
line, with the "trickle" option as specified in [RFC8840],
Section 4.1.3 and the "ice2" option as specified in [RFC8445],
Section 10, is only included if such an option was present in the
offer. -->
</t>
<t>The next step is to generate session-level lip sync <t>The next step is to generate session-level lip sync
groups, as defined in groups, as defined in
<xref target="RFC5888" />, Section 7. For each group of type <xref target="RFC5888" sectionFormat="comma" section="7"/>. For each g roup of type
"LS" present in the offer, select the local RtpTransceivers "LS" present in the offer, select the local RtpTransceivers
that are referenced by the MID values in the specified group, that are referenced by the MID values in the specified group,
and determine which of them either reference a common local and determine which of them either reference a common local
MediaStream (specified in the calls to MediaStream (specified in the calls to
addTrack/addTransceiver used to create them), or have no addTrack/addTransceiver used to create them) or have no
MediaStream to reference because they were not created by MediaStream to reference because they were not created by
addTrack/addTransceiver. If at least two such RtpTransceivers addTrack/addTransceiver. If at least two such RtpTransceivers
exist, a group of type "LS" with the mid values of these exist, a group of type "LS" with the mid values of these
RtpTransceivers MUST be added. Otherwise the offered "LS" RtpTransceivers <bcp14>MUST</bcp14> be added. Otherwise, the offered "
group MUST be ignored and no corresponding group generated in LS"
group <bcp14>MUST</bcp14> be ignored and no corresponding group genera
ted in
the answer.</t> the answer.</t>
<t>As a simple example, consider the following offer of a <t>As a simple example, consider the following offer of a
single audio and single video track contained in the same single audio and single video track contained in the same
MediaStream. SDP lines not relevant to this example have been MediaStream. SDP lines not relevant to this example have been
removed for clarity. As explained in removed for clarity. As explained in
<xref target="sec-create-offer" />, a group of type "LS" has <xref target="sec-create-offer" format="default"/>, a group of type "L S" has
been added that references each track's RtpTransceiver.</t> been added that references each track's RtpTransceiver.</t>
<sourcecode name="" type="sdp"><![CDATA[
<t>
<figure>
<artwork>
<![CDATA[
a=group:LS a1 v1 a=group:LS a1 v1
m=audio 10000 UDP/TLS/RTP/SAVPF 0 m=audio 10000 UDP/TLS/RTP/SAVPF 0
a=mid:a1 a=mid:a1
a=msid:ms1 a=msid:ms1
m=video 10001 UDP/TLS/RTP/SAVPF 96 m=video 10001 UDP/TLS/RTP/SAVPF 96
a=mid:v1 a=mid:v1
a=msid:ms1 a=msid:ms1 ]]></sourcecode>
]]>
</artwork>
</figure>
</t>
<t>If the answerer uses a single MediaStream when it adds its <t>If the answerer uses a single MediaStream when it adds its
tracks, both of its transceivers will reference this stream, tracks, both of its transceivers will reference this stream,
and so the subsequent answer will contain a "LS" group and so the subsequent answer will contain a "LS" group
identical to that in the offer, as shown below:</t> identical to that in the offer, as shown below:</t>
<sourcecode name="" type="sdp"><![CDATA[
<t>
<figure>
<artwork>
<![CDATA[
a=group:LS a1 v1 a=group:LS a1 v1
m=audio 20000 UDP/TLS/RTP/SAVPF 0 m=audio 20000 UDP/TLS/RTP/SAVPF 0
a=mid:a1 a=mid:a1
a=msid:ms2 a=msid:ms2
m=video 20001 UDP/TLS/RTP/SAVPF 96 m=video 20001 UDP/TLS/RTP/SAVPF 96
a=mid:v1 a=mid:v1
a=msid:ms2 a=msid:ms2 ]]></sourcecode>
]]>
</artwork>
</figure>
</t>
<t>However, if the answerer groups its tracks into separate <t>However, if the answerer groups its tracks into separate
MediaStreams, its transceivers will reference different MediaStreams, its transceivers will reference different
streams, and so the subsequent answer will not contain a "LS" streams, and so the subsequent answer will not contain a "LS"
group.</t> group.</t>
<sourcecode name="" type="sdp"><![CDATA[
<t>
<figure>
<artwork>
<![CDATA[
m=audio 20000 UDP/TLS/RTP/SAVPF 0 m=audio 20000 UDP/TLS/RTP/SAVPF 0
a=mid:a1 a=mid:a1
a=msid:ms2a a=msid:ms2a
m=video 20001 UDP/TLS/RTP/SAVPF 96 m=video 20001 UDP/TLS/RTP/SAVPF 96
a=mid:v1 a=mid:v1
a=msid:ms2b a=msid:ms2b ]]></sourcecode>
]]>
</artwork>
</figure>
</t>
<t>Finally, if the answerer does not add any tracks, its <t>Finally, if the answerer does not add any tracks, its
transceivers will not reference any MediaStreams, causing the transceivers will not reference any MediaStreams, causing the
preferences of the offerer to be maintained, and so the preferences of the offerer to be maintained, and so the
subsequent answer will contain an identical "LS" group.</t> subsequent answer will contain an identical "LS" group.</t>
<sourcecode name="" type="sdp"><![CDATA[
<t>
<figure>
<artwork>
<![CDATA[
a=group:LS a1 v1 a=group:LS a1 v1
m=audio 20000 UDP/TLS/RTP/SAVPF 0 m=audio 20000 UDP/TLS/RTP/SAVPF 0
a=mid:a1 a=mid:a1
a=recvonly a=recvonly
m=video 20001 UDP/TLS/RTP/SAVPF 96 m=video 20001 UDP/TLS/RTP/SAVPF 96
a=mid:v1 a=mid:v1
a=recvonly a=recvonly ]]></sourcecode>
]]> <t>The example in <xref target="sec.detailed-example" format="default"
</artwork> /> shows a more involved case of "LS" group
</figure>
</t>
<t>The
<xref target="sec.detailed-example" /> example later in this
document shows a more involved case of "LS" group
generation.</t> generation.</t>
<t>The next step is to generate "m=" sections for each "m="
<t>The next step is to generate m= sections for each m=
section that is present in the remote offer, as specified in section that is present in the remote offer, as specified in
<xref target="RFC3264"></xref>, Section 6. For the purposes <xref target="RFC3264" sectionFormat="comma" section="6"/>. For the pu rposes
of this discussion, any session-level attributes in the offer of this discussion, any session-level attributes in the offer
that are also valid as media-level attributes are considered that are also valid as media-level attributes are considered
to be present in each m= section. Each offered m= section to be present in each "m=" section. Each offered "m=" section
will have an associated RtpTransceiver, as described in will have an associated RtpTransceiver, as described in
<xref target="sec.applying-a-remote-desc" />. If there are <xref target="sec.applying-a-remote-desc" format="default"/>. If there
more RtpTransceivers than there are m= sections, the are
more RtpTransceivers than there are "m=" sections, the
unmatched RtpTransceivers will need to be associated in a unmatched RtpTransceivers will need to be associated in a
subsequent offer.</t> subsequent offer.</t>
<t>For each offered "m=" section, if any of the following
conditions are true, the corresponding "m=" section in the
answer <bcp14>MUST</bcp14> be marked as rejected by setting the port i
n the
"m=" line to zero, as indicated in
<xref target="RFC3264" sectionFormat="comma" section="6"/>, and furthe
r
processing for this "m=" section can be skipped:
</t>
<ul spacing="normal">
<li>The associated RtpTransceiver has been stopped.</li>
<li>None of the offered media formats are supported and, if
applicable, allowed by codec preferences.
<t>For each offered m= section, if any of the following <!-- [rfced] Section 5.3.1: Does this text mean that the offered
conditions are true, the corresponding m= section in the media formats are allowed (in which case it should say "they are
answer MUST be marked as rejected by setting the port in the allowed") or are not allowed (in which case "and, if applicable"
m= line to zero, as indicated in should be "or, if applicable")?
<xref target="RFC3264"></xref>, Section 6, and further
processing for this m= section can be skipped:
<list style="symbols">
<t>The associated RtpTransceiver has been stopped.</t>
<t>None of the offered media formats are supported and, if
applicable, allowed by codec preferences.</t>
<t>The bundle policy is "max-bundle", and this is not the
first m= section or in the same bundle group as the first
m= section.</t>
<t>The bundle policy is "balanced", and this is not the Original:
first m= section for this media type or in the same bundle o None of the offered media formats are supported and, if
group as the first m= section for this media type.</t> applicable, allowed by codec preferences. -->
<t>This m= section is in a bundle group, and the group's </li>
offerer tagged m= section is being rejected due to one of <li>The bundle policy is "max-bundle", and this is not the
the above reasons. This requires all m= sections in the first "m=" section or in the same bundle group as the first
"m=" section.</li>
<li>The bundle policy is "balanced", and this is not the
first "m=" section for this media type or in the same bundle
group as the first "m=" section for this media type.</li>
<li>This "m=" section is in a bundle group, and the group's
offerer tagged "m=" section is being rejected due to one of
the above reasons. This requires all "m=" sections in the
bundle group to be rejected, as specified in bundle group to be rejected, as specified in
<xref target="I-D.ietf-mmusic-sdp-bundle-negotiation" />, <xref target="RFC8843" sectionFormat="comma" section="7.3.3"/>.</li>
Section 7.3.3.</t> </ul>
</list></t> <t>Otherwise, each "m=" section in the answer should then be
<t>Otherwise, each m= section in the answer should then be
generated as specified in generated as specified in
<xref target="RFC3264"></xref>, Section 6.1. For the m= line <xref target="RFC3264" sectionFormat="comma" section="6.1"/>. For the "m=" line
itself, the following rules must be followed: itself, the following rules must be followed:
<list style="symbols">
<t>The port value would normally be set to the port of the <!-- [rfced] Section 5.3.1: Should "must be followed:" here be "MUST be
default ICE candidate for this m= section, but given that followed:"? (We ask because (1) we see "For the m= line itself, the following
no candidates are available yet, the "dummy" port value of rules MUST be followed:" in Section 5.2.1 and (2) all of the rules
9 (Discard) MUST be used, as indicated in listed below this sentence include a "MUST.")
<xref target="I-D.ietf-ice-trickle"></xref>, Section
5.1.</t>
<t>The &lt;proto&gt; field MUST be set to exactly match the Original:
&lt;proto&gt; field for the corresponding m= line in the For the m= line itself, the
offer.</t> following rules must be followed: -->
<t>If codec preferences have been set for the associated </t>
transceiver, media formats MUST be generated in the <ul spacing="normal">
<li>The port value would normally be set to the port of the
default ICE candidate for this "m=" section, but given that
no candidates are available yet, the "dummy" port value of
9 (Discard) <bcp14>MUST</bcp14> be used, as indicated in
<xref target="RFC8840" sectionFormat="comma" section="4.1.1"/>.</li>
<li>The &lt;proto&gt; field <bcp14>MUST</bcp14> be set to exactly ma
tch the
&lt;proto&gt; field for the corresponding "m=" line in the
offer.</li>
<li>If codec preferences have been set for the associated
transceiver, media formats <bcp14>MUST</bcp14> be generated in the
corresponding order, regardless of what was offered, and corresponding order, regardless of what was offered, and
MUST exclude any codecs not present in the codec <bcp14>MUST</bcp14> exclude any codecs not present in the codec
preferences.</t> preferences.</li>
<li>Otherwise, the media formats on the "m=" line <bcp14>MUST</bcp14
<t>Otherwise, the media formats on the m= line MUST be > be
generated in the same order as those offered in the current generated in the same order as those offered in the current
remote description, excluding any currently unsupported remote description, excluding any currently unsupported
formats. Any currently available media formats that are not formats. Any currently available media formats that are not
present in the current remote description MUST be added present in the current remote description <bcp14>MUST</bcp14> be add
after all existing formats.</t> ed
after all existing formats.</li>
<t>In either case, the media formats in the answer MUST <li>In either case, the media formats in the answer <bcp14>MUST</bcp
include at least one format that is present in the offer, 14>
but MAY include formats that are locally supported but not include at least one format that is present in the offer
but <bcp14>MAY</bcp14> include formats that are locally supported bu
t not
present in the offer, as mentioned in present in the offer, as mentioned in
<xref target="RFC3264" />, Section 6.1. If no common format <xref target="RFC3264" sectionFormat="comma" section="6.1"/>. If no
exists, the m= section is rejected as described above.</t> common format
</list></t> exists, the "m=" section is rejected as described above.</li>
</ul>
<t>The m= line MUST be followed immediately by a "c=" line, <t>The "m=" line <bcp14>MUST</bcp14> be followed immediately by a "c=" line,
as specified in as specified in
<xref target="RFC4566"></xref>, Section 5.7. Again, as no <xref target="RFC4566" sectionFormat="comma" section="5.7"/>. Again, a s no
candidates are available yet, the "c=" line must contain the candidates are available yet, the "c=" line must contain the
"dummy" value "IN IP4 0.0.0.0", as defined in "dummy" value "IN IP4 0.0.0.0", as defined in
<xref target="I-D.ietf-ice-trickle"></xref>, Section 5.1.</t> <xref target="RFC8840" sectionFormat="comma" section="4.1.3"/>.</t>
<t>If the offer supports bundle, all "m=" sections to be
<t>If the offer supports bundle, all m= sections to be
bundled must use the same ICE credentials and candidates; all bundled must use the same ICE credentials and candidates; all
m= sections not being bundled must use unique ICE credentials "m=" sections not being bundled must use unique ICE credentials
and candidates. Each m= section MUST contain the following and candidates. Each "m=" section <bcp14>MUST</bcp14> contain the foll
owing
attributes (which are of attribute types other than IDENTICAL attributes (which are of attribute types other than IDENTICAL
and TRANSPORT): or TRANSPORT):
<list style="symbols"> </t>
<ul spacing="normal">
<t>If and only if present in the offer, an "a=mid" line, as <li>If and only if present in the offer, an "a=mid" line, as
specified in specified in
<xref target="RFC5888"></xref>, Section 9.1. The "mid" <xref target="RFC5888" sectionFormat="comma" section="9.1"/>. The "m
value MUST match that specified in the offer.</t> id"
value <bcp14>MUST</bcp14> match that specified in the offer.</li>
<t>A direction attribute, determined by applying the rules <li>A direction attribute, determined by applying the rules
regarding the offered direction specified in regarding the offered direction specified in
<xref target="RFC3264" />, Section 6.1, and then <xref target="RFC3264" sectionFormat="comma" section="6.1"/>, and th en
intersecting with the direction of the associated intersecting with the direction of the associated
RtpTransceiver. For example, in the case where an m= RtpTransceiver. For example, in the case where an "m="
section is offered as "sendonly", and the local transceiver section is offered as "sendonly" and the local transceiver
is set to "sendrecv", the result in the answer is a is set to "sendrecv", the result in the answer is a
"recvonly" direction.</t> "recvonly" direction.</li>
<li>For each media format on the "m=" line, "a=rtpmap" and "a=fmtp"
<t>For each media format on the m= line, "a=rtpmap" and lines, as specified in
"a=fmtp" lines, as specified in <xref target="RFC4566" sectionFormat="comma" section="6"/> and
<xref target="RFC4566"></xref>, Section 6, and <xref target="RFC3264" sectionFormat="comma" section="6.1"/>.</li>
<xref target="RFC3264"></xref>, Section 6.1.</t> <li>If "rtx" is present in the offer, for each primary codec
<t>If "rtx" is present in the offer, for each primary codec
where RTP retransmission should be used, a corresponding where RTP retransmission should be used, a corresponding
"a=rtpmap" line indicating "rtx" with the clock rate of the "a=rtpmap" line indicating "rtx" with the clock rate of the
primary codec and an "a=fmtp" line that references the primary codec and an "a=fmtp" line that references the
payload type of the primary codec, as specified in payload type of the primary codec, as specified in
<xref target="RFC4588"></xref>, Section 8.1.</t> <xref target="RFC4588" sectionFormat="comma" section="8.1"/>.</li>
<li>For each supported FEC mechanism, "a=rtpmap" and
<t>For each supported FEC mechanism, "a=rtpmap" and
"a=fmtp" lines, as specified in "a=fmtp" lines, as specified in
<xref target="RFC4566"></xref>, Section 6. The FEC <xref target="RFC4566" sectionFormat="comma" section="6"/>. The FEC
mechanisms that MUST be supported are specified in mechanisms that <bcp14>MUST</bcp14> be supported are specified in
<xref target="I-D.ietf-rtcweb-fec"></xref>, Section 6, and <xref target="RFC8854" sectionFormat="comma" section="6"/>, and
specific usage for each media type is outlined in Sections specific usage for each media type is outlined in Sections
4 and 5.</t> <xref target="sec.interface" format="counter"/> and <xref
target="sec.sdp-interaction-procedure" format="counter"/>.</li>
<t>If this m= section is for media with configurable <li>If this "m=" section is for media with configurable
durations of media per packet, e.g., audio, an "a=maxptime" durations of media per packet, e.g., audio, an "a=maxptime"
line, as described in line, as described in
<xref target="sec-create-offer" />.</t> <xref target="sec-create-offer" format="default"/>.</li>
<li>If this "m=" section is for video media and there are
<t>If this m= section is for video media, and there are known limitations on the size of images that can be
known limitations on the size of images which can be
decoded, an "a=imageattr" line, as specified in decoded, an "a=imageattr" line, as specified in
<xref target="sec.imageattr"></xref>.</t> <xref target="sec.imageattr" format="default"/>.</li>
<li>For each supported RTP header extension that is present
<t>For each supported RTP header extension that is present
in the offer, an "a=extmap" line, as specified in in the offer, an "a=extmap" line, as specified in
<xref target="RFC5285"></xref>, Section 5. The list of <xref target="RFC5285" sectionFormat="comma" section="5"/>. The list
header extensions that SHOULD/MUST be supported is of
header extensions that <bcp14>SHOULD</bcp14>/<bcp14>MUST</bcp14> be
supported is
specified in specified in
<xref target="I-D.ietf-rtcweb-rtp-usage"></xref>, Section <xref target="RFC8834" sectionFormat="comma" section="5.2"/>. Any he
5.2. Any header extensions that require encryption MUST be ader extensions that require encryption <bcp14>MUST</bcp14> be
specified as indicated in specified as indicated in
<xref target="RFC6904"></xref>, Section 4.</t> <xref target="RFC6904" sectionFormat="comma" section="4"/>.</li>
<li>For each supported RTCP feedback mechanism that is
<t>For each supported RTCP feedback mechanism that is
present in the offer, an "a=rtcp-fb" line, as specified in present in the offer, an "a=rtcp-fb" line, as specified in
<xref target="RFC4585"></xref>, Section 4.2. The list of <xref target="RFC4585" sectionFormat="comma" section="4.2"/>. The li
RTCP feedback mechanisms that SHOULD/MUST be supported is st of
RTCP feedback mechanisms that <bcp14>SHOULD</bcp14>/<bcp14>MUST</bcp
14> be supported is
specified in specified in
<xref target="I-D.ietf-rtcweb-rtp-usage"></xref>, Section <xref target="RFC8834" sectionFormat="comma" section="5.1"/>.</li>
5.1.</t> <li>
<t>If the RtpTransceiver has a sendrecv or sendonly
<t>If the RtpTransceiver has a sendrecv or sendonly
direction: direction:
<list style="symbols"> </t>
<ul spacing="normal">
<t>For each MediaStream that was associated with the <li>For each MediaStream that was associated with the
transceiver when it was created via addTrack or transceiver when it was created via addTrack or
addTransceiver, an "a=msid" line, as specified in addTransceiver, an "a=msid" line, as specified in
<xref target="I-D.ietf-mmusic-msid"></xref>, Section 2, <xref target="RFC8830" sectionFormat="comma" section="2"/>,
but omitting the "appdata" field.</t> but omitting the "appdata" field.</li>
</list></t> </ul>
</list></t> </li>
</ul>
<t>Each m= section which is not bundled into another m= <t>Each "m=" section that is not bundled into another "m="
section, MUST contain the following attributes (which are of section <bcp14>MUST</bcp14> contain the following attributes (which ar
e of
category IDENTICAL or TRANSPORT):</t> category IDENTICAL or TRANSPORT):</t>
<ul spacing="normal">
<t> <li>"a=ice-ufrag" and "a=ice-pwd" lines, as specified in
<list style="symbols"> <xref target="RFC8839" sectionFormat="comma" section="5.4"/>.</li>
<li>For each desired digest algorithm, one or more
<t>"a=ice-ufrag" and "a=ice-pwd" lines, as specified in
<xref target="I-D.ietf-mmusic-ice-sip-sdp"></xref>,
Section 4.4.</t>
<t>For each desired digest algorithm, one or more
"a=fingerprint" lines for each of the endpoint's "a=fingerprint" lines for each of the endpoint's
certificates, as specified in certificates, as specified in
<xref target="RFC8122"></xref>, Section 5.</t> <xref target="RFC8122" sectionFormat="comma" section="5"/>.</li>
<li>An "a=setup" line, as specified in
<t>An "a=setup" line, as specified in <xref target="RFC4145" sectionFormat="comma" section="4"/> and cla
<xref target="RFC4145"></xref>, Section 4, and clarified rified
for use in DTLS-SRTP scenarios in for use in DTLS-SRTP scenarios in
<xref target="RFC5763"></xref>, Section 5. The role value <xref target="RFC5763" sectionFormat="comma" section="5"/>. The ro
in the answer MUST be "active" or "passive". When the le value
in the answer <bcp14>MUST</bcp14> be "active" or "passive". When t
he
offer contains the "actpass" value, as will always be the offer contains the "actpass" value, as will always be the
case with JSEP endpoints, the answerer SHOULD use the case with JSEP endpoints, the answerer <bcp14>SHOULD</bcp14> use t
"active" role. Offers from non-JSEP endpoints MAY send he
other values for "a=setup", in which case the answer MUST "active" role. Offers from non-JSEP endpoints <bcp14>MAY</bcp14> s
use a value consistent with the value in the offer.</t> end
other values for "a=setup", in which case the answer <bcp14>MUST</
<t>An "a=tls-id" line, as specified in bcp14>
<xref target="I-D.ietf-mmusic-dtls-sdp" />, Section use a value consistent with the value in the offer.</li>
5.3.</t> <li>An "a=tls-id" line, as specified in
<xref target="RFC8842" sectionFormat="comma" section="5.3"/>.</li>
<t>If present in the offer, an "a=rtcp-mux" line, as <li>If present in the offer, an "a=rtcp-mux" line, as
specified in specified in
<xref target="RFC5761"></xref>, Section 5.1.3. Otherwise, <xref target="RFC5761" sectionFormat="comma" section="5.1.3"/>. Ot herwise,
an "a=rtcp" line, as specified in an "a=rtcp" line, as specified in
<xref target="RFC3605"></xref>, Section 2.1, containing <xref target="RFC3605" sectionFormat="comma" section="2.1"/>, cont aining
the dummy value "9 IN IP4 0.0.0.0" (because no candidates the dummy value "9 IN IP4 0.0.0.0" (because no candidates
have yet been gathered).</t> have yet been gathered).</li>
<li>If present in the offer, an "a=rtcp-rsize" line, as
<t>If present in the offer, an "a=rtcp-rsize" line, as
specified in specified in
<xref target="RFC5506"></xref>, Section 5.</t> <xref target="RFC5506" sectionFormat="comma" section="5"/>.</li>
</list> </ul>
</t> <t>If a data channel "m=" section has been offered, an "m="
section <bcp14>MUST</bcp14> also be generated for data. The &lt;media&
<t>If a data channel m= section has been offered, a m= gt;
section MUST also be generated for data. The &lt;media&gt; field <bcp14>MUST</bcp14> be set to "application", and the &lt;proto&g
field MUST be set to "application" and the &lt;proto&gt; and t; and
&lt;fmt&gt; fields MUST be set to exactly match the fields in &lt;fmt&gt; fields <bcp14>MUST</bcp14> be set to exactly match the fie
lds in
the offer.</t> the offer.</t>
<t>Within the data "m=" section, an "a=mid" line <bcp14>MUST</bcp14> b
<t>Within the data m= section, an "a=mid" line MUST be e
generated and included as described above, along with an generated and included as described above, along with an
"a=sctp-port" line referencing the SCTP port number, as "a=sctp-port" line referencing the SCTP port number, as
defined in defined in
<xref target="I-D.ietf-mmusic-sctp-sdp"></xref>, Section 5.1, <xref target="RFC8841" sectionFormat="comma" section="5.1"/>;
and, if appropriate, an "a=max-message-size" line, as defined and, if appropriate, an "a=max-message-size" line, as defined
in in
<xref target="I-D.ietf-mmusic-sctp-sdp"></xref>, Section <xref target="RFC8841" sectionFormat="comma" section="6.1"/>.</t>
6.1.</t>
<t>As discussed above, the following attributes of category <t>As discussed above, the following attributes of category
IDENTICAL or TRANSPORT are included only if the data m= IDENTICAL or TRANSPORT are included only if the data "m="
section is not bundled into another m= section: section is not bundled into another "m=" section:
<list style="symbols"> </t>
<ul spacing="normal">
<t>"a=ice-ufrag"</t> <li>"a=ice-ufrag"</li>
<li>"a=ice-pwd"</li>
<t>"a=ice-pwd"</t> <li>"a=fingerprint"</li>
<li>"a=setup"</li>
<t>"a=fingerprint"</t> <li>"a=tls-id"</li>
</ul>
<t>"a=setup"</t> <t>Note that if media "m=" sections are bundled into a data "m="
<t>"a=tls-id"</t>
</list></t>
<t>Note that if media m= sections are bundled into a data m=
section, then certain TRANSPORT and IDENTICAL attributes may section, then certain TRANSPORT and IDENTICAL attributes may
also appear in the data m= section even if they would also appear in the data "m=" section even if they would
otherwise only be appropriate for a media m= section (e.g., otherwise only be appropriate for a media "m=" section (e.g.,
"a=rtcp-mux").</t> "a=rtcp-mux").</t>
<t>If "a=group" attributes with semantics of "BUNDLE" are <t>If "a=group" attributes with semantics of "BUNDLE" are
offered, corresponding session-level "a=group" attributes offered, corresponding session-level "a=group" attributes
MUST be added as specified in <bcp14>MUST</bcp14> be added as specified in
<xref target="RFC5888"></xref>. These attributes MUST have <xref target="RFC5888" format="default"/>. These attributes <bcp14>MUS
semantics "BUNDLE", and MUST include the all mid identifiers T</bcp14> have
semantics "BUNDLE" and <bcp14>MUST</bcp14> include all mid identifiers
from the offered bundle groups that have not been rejected. from the offered bundle groups that have not been rejected.
Note that regardless of the presence of "a=bundle-only" in Note that regardless of the presence of "a=bundle-only" in
the offer, no m= sections in the answer should have an the offer, no "m=" sections in the answer should have an
"a=bundle-only" line.</t> "a=bundle-only" line.</t>
<t>Attributes that are common between all "m=" sections <bcp14>MAY</bc
p14> be
moved to the session level if explicitly defined to be valid at
the session level.
<t>Attributes that are common between all m= sections MAY be <!-- [rfced] Section 5.3.1: Per "the session level" used elsewhere in this
moved to session-level, if explicitly defined to be valid at document and "ice-options are now at session level" in the Change Log, we
session-level.</t> changed "to session-level" and "at session-level" to "to the session level"
and "at the session level." Please let us know any objections.
Original:
Attributes that are common between all m= sections MAY be moved to
session-level, if explicitly defined to be valid at session-level.
Currently:
Attributes that are common between all "m=" sections MAY be moved to
the session level if explicitly defined to be valid at the session
level. -->
</t>
<t>The attributes prohibited in the creation of offers are <t>The attributes prohibited in the creation of offers are
also prohibited in the creation of answers.</t> also prohibited in the creation of answers.</t>
</section> </section>
<section title="Subsequent Answers" <section anchor="sec.subsequent-answers" numbered="true" toc="default">
anchor="sec.subsequent-answers"> <name>Subsequent Answers</name>
<t>When createAnswer is called a second (or later) time or
<t>When createAnswer is called a second (or later) time, or
is called after a local description has already been is called after a local description has already been
installed, the processing is somewhat different than for an installed, the processing is somewhat different than for an
initial answer.</t> initial answer.</t>
<t>If the previous answer was not applied using <t>If the previous answer was not applied using
setLocalDescription, meaning the PeerConnection is still in setLocalDescription, meaning the PeerConnection is still in
the "have-remote-offer" state, the steps for generating an the "have-remote-offer" state, the steps for generating an
initial answer should be followed, subject to the following initial answer should be followed, subject to the following
restriction: restriction:
<list style="symbols"> </t>
<ul spacing="normal">
<t>The fields of the "o=" line MUST stay the same except <li>The fields of the "o=" line <bcp14>MUST</bcp14> stay the same ex
for the &lt;session-version&gt; field, which MUST increment cept
for the &lt;session-version&gt; field, which <bcp14>MUST</bcp14> inc
rement
if the session description changes in any way from the if the session description changes in any way from the
previously generated answer.</t> previously generated answer.</li>
</list></t> </ul>
<t>If any session description was previously supplied to <t>If any session description was previously supplied to
setLocalDescription, an answer is generated by following the setLocalDescription, an answer is generated by following the
steps in the "have-remote-offer" state above, along with steps in the "have-remote-offer" state above, along with
these exceptions: these exceptions:
<list style="symbols"> </t>
<ul spacing="normal">
<t>The "s=" and "t=" lines MUST stay the same.</t> <li>The "s=" and "t=" lines <bcp14>MUST</bcp14> stay the same.</li>
<li>Each "m=" and "c=" line <bcp14>MUST</bcp14> be filled in with th
<t>Each "m=" and c=" line MUST be filled in with the port e port
and address of the default candidate for the m= section, as and address of the default candidate for the "m=" section, as
described in described in
<xref target="I-D.ietf-mmusic-ice-sip-sdp"></xref>, <xref target="RFC8839" sectionFormat="comma" section="4.2.1.2"/>. No
Section 3.2.1.2. Note that in certain cases, the m= line protocol te that in certain cases, the "m=" line protocol
may not match that of the default candidate, because the m= line may not match that of the default candidate, because the "m=" line
protocol value MUST match what was supplied in the offer, as protocol value <bcp14>MUST</bcp14> match what was supplied in the of
described above.</t> fer, as
described above.</li>
<t>Each "a=ice-ufrag" and "a=ice-pwd" line MUST stay the <li>Each "a=ice-ufrag" and "a=ice-pwd" line <bcp14>MUST</bcp14> stay
same, unless the m= section is restarting, in which case the
same, unless the "m=" section is restarting, in which case
new ICE credentials must be created as specified in new ICE credentials must be created as specified in
<xref target="I-D.ietf-mmusic-ice-sip-sdp"></xref>, <xref target="RFC8839" sectionFormat="comma" section="4.4.1.1.1"/>.
Section 3.4.1.1.1. If the m= If the "m="
section is bundled into another m= section, it still MUST section is bundled into another "m=" section, it still <bcp14>MUST
NOT contain any ICE credentials.</t> NOT</bcp14> contain any ICE credentials.</li>
<li>Each "a=tls-id" line <bcp14>MUST</bcp14> stay the same, unless t
<t>Each "a=tls-id" line MUST stay the same unless the he
offerer's "a=tls-id" line changed, in which case a new offerer's "a=tls-id" line changed, in which case a new
"a=tls-id" value MUST be created, as described in "a=tls-id" value <bcp14>MUST</bcp14> be created, as described in
<xref target="I-D.ietf-mmusic-dtls-sdp" />, Section <xref target="RFC8842" sectionFormat="comma" section="5.2"/>.</li>
5.2.</t> <li>Each "a=setup" line <bcp14>MUST</bcp14> use an "active" or "pass
ive"
<t>Each "a=setup" line MUST use an "active" or "passive"
role value consistent with the existing DTLS association, role value consistent with the existing DTLS association,
if the association is being continued by the offerer.</t> if the association is being continued by the offerer.</li>
<li>RTCP multiplexing must be used, and an "a=rtcp-mux" line
<t>RTCP multiplexing must be used, and an "a=rtcp-mux" line inserted if and only if the "m=" section previously used RTCP
inserted if and only if the m= section previously used RTCP multiplexing.</li>
multiplexing.</t> <li>If the "m=" section is not bundled into another "m=" section
<t>If the m= section is not bundled into another m= section
and RTCP multiplexing is not active, an "a=rtcp" attribute and RTCP multiplexing is not active, an "a=rtcp" attribute
line MUST be filled in with the port and address of the line <bcp14>MUST</bcp14> be filled in with the port and address of t he
default RTCP candidate. If no RTCP candidates have yet been default RTCP candidate. If no RTCP candidates have yet been
gathered, dummy values MUST be used, as described in the gathered, dummy values <bcp14>MUST</bcp14> be used, as described in
initial answer section above.</t> <xref target="sec.initial-answers"/> above.</li>
<t>If the m= section is not bundled into another m= <li>If the "m=" section is not bundled into another "m="
section, for each candidate that has been gathered during section, for each candidate that has been gathered during
the most recent gathering phase (see the most recent gathering phase (see
<xref target="sec.ice-gather-overview"></xref>), an <xref target="sec.ice-gather-overview" format="default"/>), an
"a=candidate" line MUST be added, as defined in "a=candidate" line <bcp14>MUST</bcp14> be added, as defined in
<xref target="I-D.ietf-mmusic-ice-sip-sdp"></xref>, <xref target="RFC8839" sectionFormat="comma" section="5.1"/>.
Section 4.1.
If candidate gathering for the section has completed, an If candidate gathering for the section has completed, an
"a=end-of-candidates" attribute MUST be added, as described "a=end-of-candidates" attribute <bcp14>MUST</bcp14> be added, as des cribed
in in
<xref target="I-D.ietf-ice-trickle"></xref>, Section 9.3. <xref target="RFC8840" sectionFormat="comma" section="8.2"/>.
If the m= section is bundled into another m= section, both If the "m=" section is bundled into another "m=" section, both
"a=candidate" and "a=end-of-candidates" MUST be "a=candidate" and "a=end-of-candidates" <bcp14>MUST</bcp14> be
omitted.</t> omitted.
<t>For RtpTransceivers that are not stopped, the "a=msid" <!-- [rfced] Section 5.3.2: We found this RFC Editor Note on
line(s) MUST stay the same, regardless of changes to the <https://datatracker.ietf.org/doc/draft-ietf-rtcweb-jsep/writeup/>:
"OLD:
o If the m= section is not bundled into another m= section, for each
candidate that has been gathered during the most recent gathering
phase (see Section 3.5.1), an "a=candidate" line MUST be added, as
defined in [RFC5245], Section 4.3., paragraph 3. If candidate
gathering for the section has completed, an "a=end-of-candidates"
attribute MUST be added, as described in [I-D.ietf-ice-trickle],
Section 9.3. If the m= section is bundled into another m=
section, both "a=candidate" and "a=end-of-candidates" MUST be
omitted.
NEW:
o If the m= section is not bundled into another m= section, for each
candidate that has been gathered during the most recent gathering
phase (see Section 3.5.1), an "a=candidate" line MUST be added, as
defined in [RFC5245], Section 4.3., paragraph 3. If candidate
gathering for the section has completed, an "a=end-of-candidates"
attribute MUST be added, as described in
[I-D.ietf-mmusic-trickle-ice-sip], Section 8.2. If the m= section is
bundled into another m= section, both "a=candidate" and
"a=end-of-candidates" MUST be omitted."
Please note that the "OLD" text does not match what we found in the
provided draft (i.e., "[RFC5245], Section 4.3., paragraph 3" versus
"[I-D.ietf-mmusic-ice-sip-sdp], Section 4.1"):
o If the m= section is not bundled into another m= section, for each
candidate that has been gathered during the most recent gathering
phase (see Section 3.5.1), an "a=candidate" line MUST be added, as
defined in [I-D.ietf-mmusic-ice-sip-sdp], Section 4.1. If
candidate gathering for the section has completed, an "a=end-of-
candidates" attribute MUST be added, as described in
[I-D.ietf-ice-trickle], Section 9.3. If the m= section is bundled
into another m= section, both "a=candidate" and "a=end-of-
candidates" MUST be omitted.
Please review, and let us know if further changes are needed. (As
noted previously, "[RFC8839]" and "[RFC8840]" are the RFC numbers assigned
for [I-D.ietf-mmusic-ice-sip-sdp] and
[I-D.ietf-mmusic-trickle-ice-sip], respectively.)
Currently:
* If the "m=" section is not bundled into another "m=" section, for each
candidate that has been gathered during the most recent gathering
phase (see Section 3.5.1), an "a=candidate" line MUST be added, as
defined in [RFC8839], Section 5.1. If candidate gathering for the
section has completed, an "a=end-of-candidates" attribute MUST be
added, as described in [RFC8840], Section 8.2. If the "m=" section
is bundled into another "m=" section, both "a=candidate" and "a=end-
of-candidates" MUST be omitted. -->
</li>
<li>For RtpTransceivers that are not stopped, the "a=msid"
line(s) <bcp14>MUST</bcp14> stay the same, regardless of changes to
the
transceiver's direction or track. If no "a=msid" line is transceiver's direction or track. If no "a=msid" line is
present in the current description, "a=msid" line(s) MUST present in the current description, "a=msid" line(s) <bcp14>MUST</bc p14>
be generated according to the same rules as for an initial be generated according to the same rules as for an initial
answer.</t> answer.</li>
</list></t> </ul>
</section> </section>
<section title="Options Handling" <section anchor="sec.options-handling2" numbered="true" toc="default">
anchor="sec.options-handling2"> <name>Options Handling</name>
<t>The createAnswer method takes as a parameter an <t>The createAnswer method takes as a parameter an
RTCAnswerOptions object. The set of parameters for RTCAnswerOptions object. The set of parameters for
RTCAnswerOptions is different than those supported in RTCAnswerOptions is different than those supported in
RTCOfferOptions; the IceRestart option is unnecessary, as ICE RTCOfferOptions; the IceRestart option is unnecessary, as ICE
credentials will automatically be changed for all m= sections credentials will automatically be changed for all "m=" sections
where the offerer chose to perform ICE restart.</t> where the offerer chose to perform ICE restart.</t>
<t>The following options are supported in <t>The following options are supported in
RTCAnswerOptions.</t> RTCAnswerOptions.</t>
<section title="VoiceActivityDetection" <section anchor="sec.voiceactivitydetection2" numbered="true" toc="def
anchor="sec.voiceactivitydetection2"> ault">
<name>VoiceActivityDetection</name>
<t>Silence suppression in the answer is handled as <t>Silence suppression in the answer is handled as
described in described in
<xref target="sec.voiceactivitydetection1"></xref>, with <xref target="sec.voiceactivitydetection1" format="default"/>, with
one exception: if support for silence suppression was not one exception: if support for silence suppression was not
indicated in the offer, the VoiceActivityDetection indicated in the offer, the VoiceActivityDetection
parameter has no effect, and the answer should be generated parameter has no effect, and the answer should be generated
as if VoiceActivityDetection was set to false. This is done as if VoiceActivityDetection was set to "false". This is done
on a per-codec basis (e.g., if the offerer somehow offered on a per-codec basis (e.g., if the offerer somehow offered
support for CN but set "usedtx=0" for Opus, setting support for CN but set "usedtx=0" for Opus, setting
VoiceActivityDetection to true would result in an answer VoiceActivityDetection to "true" would result in an answer
with CN codecs and "usedtx=0"). The impact of this rule is with CN codecs and "usedtx=0"). The impact of this rule is
that an answerer will not try to use silence suppression that an answerer will not try to use silence suppression
with any endpoint that does not offer it, making silence with any endpoint that does not offer it, making silence
suppression support bilateral even with non-JSEP suppression support bilateral even with non-JSEP
endpoints.</t> endpoints.</t>
</section> </section>
</section> </section>
</section> </section>
<section title="Modifying an Offer or Answer" <section anchor="sec.modifying-sdp" numbered="true" toc="default">
anchor="sec.modifying-sdp"> <name>Modifying an Offer or Answer</name>
<t>The SDP returned from createOffer or createAnswer <bcp14>MUST NOT</bc
<t>The SDP returned from createOffer or createAnswer MUST NOT p14>
be changed before passing it to setLocalDescription. If precise be changed before passing it to setLocalDescription. If precise
control over the SDP is needed, the aforementioned control over the SDP is needed, the aforementioned
createOffer/createAnswer options or RtpTransceiver APIs MUST be createOffer/createAnswer options or RtpTransceiver APIs <bcp14>MUST</bcp 14> be
used.</t> used.</t>
<t>After calling setLocalDescription with an offer or answer, <t>After calling setLocalDescription with an offer or answer,
the application MAY modify the SDP to reduce its capabilities the application <bcp14>MAY</bcp14> modify the SDP to reduce its capabili ties
before sending it to the far side, as long as it follows the before sending it to the far side, as long as it follows the
rules above that define a valid JSEP offer or answer. Likewise, rules above that define a valid JSEP offer or answer. Likewise,
an application that has received an offer or answer from a peer an application that has received an offer or answer from a peer
MAY modify the received SDP, subject to the same constraints, <bcp14>MAY</bcp14> modify the received SDP, subject to the same constrai nts,
before calling setRemoteDescription.</t> before calling setRemoteDescription.</t>
<t>As always, the application is solely responsible for what it <t>As always, the application is solely responsible for what it
sends to the other party, and all incoming SDP will be sends to the other party, and all incoming SDP will be
processed by the JSEP implementation to the extent of its processed by the JSEP implementation to the extent of its
capabilities. It is an error to assume that all SDP is capabilities. It is an error to assume that all SDP is
well-formed; however, one should be able to assume that any well formed; however, one should be able to assume that any
implementation of this specification will be able to process, implementation of this specification will be able to process,
as a remote offer or answer, unmodified SDP coming from any as a remote offer or answer, unmodified SDP coming from any
other implementation of this specification.</t> other implementation of this specification.</t>
</section> </section>
<section title="Processing a Local Description" <section anchor="sec.processing-a-local-desc" numbered="true" toc="default
anchor="sec.processing-a-local-desc"> ">
<name>Processing a Local Description</name>
<t>When a SessionDescription is supplied to <t>When a SessionDescription is supplied to
setLocalDescription, the following steps MUST be performed: setLocalDescription, the following steps <bcp14>MUST</bcp14> be performe
<list style="symbols"> d:
</t>
<t>If the description is of type "rollback", follow the <ul spacing="normal">
<li>If the description is of type "rollback", follow the
processing defined in processing defined in
<xref target="sec.processing-a-rollback" /> and skip the <xref target="sec.processing-a-rollback" format="default"/> and skip t
processing described in the rest of this section.</t> he
processing described in the rest of this section.</li>
<t>Otherwise, the type of the SessionDescription is checked <li>
<t>Otherwise, the type of the SessionDescription is checked
against the current state of the PeerConnection: against the current state of the PeerConnection:
<list style="symbols"> </t>
<ul spacing="normal">
<t>If the type is "offer", the PeerConnection state MUST be <li>If the type is "offer", the PeerConnection state <bcp14>MUST</
either "stable" or "have-local-offer".</t> bcp14> be
either "stable" or "have-local-offer".</li>
<t>If the type is "pranswer" or "answer", the <li>If the type is "pranswer" or "answer", the
PeerConnection state MUST be either "have-remote-offer" or PeerConnection state <bcp14>MUST</bcp14> be either "have-remote-offe
"have-local-pranswer".</t> r" or
</list></t> "have-local-pranswer".</li>
</ul>
<t>If the type is not correct for the current state, </li>
processing MUST stop and an error MUST be returned.</t> <li>If the type is not correct for the current state,
processing <bcp14>MUST</bcp14> stop and an error <bcp14>MUST</bcp14> b
<t>The SessionDescription is then checked to ensure that its e returned.</li>
<li>The SessionDescription is then checked to ensure that its
contents are identical to those generated in the last call to contents are identical to those generated in the last call to
createOffer/createAnswer, and thus have not been altered, as createOffer/createAnswer, and thus have not been altered, as
discussed in discussed in
<xref target="sec.modifying-sdp" />; otherwise, processing <xref target="sec.modifying-sdp" format="default"/>; otherwise, proces
MUST stop and an error MUST be returned.</t> sing
<bcp14>MUST</bcp14> stop and an error <bcp14>MUST</bcp14> be returned.
<t>Next, the SessionDescription is parsed into a data </li>
<li>Next, the SessionDescription is parsed into a data
structure, as described in structure, as described in
<xref target="sec.parsing-a-desc" /> below.</t> <xref target="sec.parsing-a-desc" format="default"/> below.</li>
<li>Finally, the parsed SessionDescription is applied as
<t>Finally, the parsed SessionDescription is applied as
described in described in
<xref target="sec.applying-a-local-desc" /> below.</t> <xref target="sec.applying-a-local-desc" format="default"/> below.</li
</list></t> >
</ul>
</section> </section>
<section title="Processing a Remote Description" <section anchor="sec.processing-a-remote-desc" numbered="true" toc="defaul
anchor="sec.processing-a-remote-desc"> t">
<name>Processing a Remote Description</name>
<t>When a SessionDescription is supplied to <t>When a SessionDescription is supplied to
setRemoteDescription, the following steps MUST be performed: setRemoteDescription, the following steps <bcp14>MUST</bcp14> be perform
<list style="symbols"> ed:
</t>
<t>If the description is of type "rollback", follow the <ul spacing="normal">
<li>If the description is of type "rollback", follow the
processing defined in processing defined in
<xref target="sec.processing-a-rollback" /> and skip the <xref target="sec.processing-a-rollback" format="default"/> and skip t
processing described in the rest of this section.</t> he
processing described in the rest of this section.</li>
<t>Otherwise, the type of the SessionDescription is checked <li>
<t>Otherwise, the type of the SessionDescription is checked
against the current state of the PeerConnection: against the current state of the PeerConnection:
<list style="symbols"> </t>
<ul spacing="normal">
<t>If the type is "offer", the PeerConnection state MUST be <li>If the type is "offer", the PeerConnection state <bcp14>MUST</
either "stable" or "have-remote-offer".</t> bcp14> be
either "stable" or "have-remote-offer".</li>
<t>If the type is "pranswer" or "answer", the <li>If the type is "pranswer" or "answer", the
PeerConnection state MUST be either "have-local-offer" or PeerConnection state <bcp14>MUST</bcp14> be either "have-local-offer
"have-remote-pranswer".</t> " or
</list></t> "have-remote-pranswer".</li>
</ul>
<t>If the type is not correct for the current state, </li>
processing MUST stop and an error MUST be returned.</t> <li>If the type is not correct for the current state,
processing <bcp14>MUST</bcp14> stop and an error <bcp14>MUST</bcp14> b
<t>Next, the SessionDescription is parsed into a data e returned.</li>
<li>Next, the SessionDescription is parsed into a data
structure, as described in structure, as described in
<xref target="sec.parsing-a-desc" /> below. If parsing fails <xref target="sec.parsing-a-desc" format="default"/> below. If parsing
for any reason, processing MUST stop and an error MUST be fails
returned.</t> for any reason, processing <bcp14>MUST</bcp14> stop and an error <bcp1
4>MUST</bcp14> be
<t>Finally, the parsed SessionDescription is applied as returned.</li>
<li>Finally, the parsed SessionDescription is applied as
described in described in
<xref target="sec.applying-a-remote-desc" /> below.</t> <xref target="sec.applying-a-remote-desc" format="default"/> below.</l
</list></t> i>
</ul>
</section> </section>
<section title="Processing a Rollback" <section anchor="sec.processing-a-rollback" numbered="true" toc="default">
anchor="sec.processing-a-rollback"> <name>Processing a Rollback</name>
<t>A rollback may be performed if the PeerConnection is in any <t>A rollback may be performed if the PeerConnection is in any
state except for "stable". This means that both offers and state except for "stable". This means that both offers and
provisional answers can be rolled back. Rollback can only be provisional answers can be rolled back. Rollback can only be
used to cancel proposed changes; there is no support for used to cancel proposed changes; there is no support for
rolling back from a stable state to a previous stable state. If rolling back from a stable state to a previous stable state. If
a rollback is attempted in the "stable" state, processing MUST a rollback is attempted in the "stable" state, processing <bcp14>MUST</b
stop and an error MUST be returned. Note that this implies that cp14>
once the answerer has performed setLocalDescription with his stop and an error <bcp14>MUST</bcp14> be returned. Note that this implie
s that
once the answerer has performed setLocalDescription with its
answer, this cannot be rolled back.</t> answer, this cannot be rolled back.</t>
<t>The effect of rollback <bcp14>MUST</bcp14> be the same regardless of
<t>The effect of rollback MUST be the same regardless of
whether setLocalDescription or setRemoteDescription is whether setLocalDescription or setRemoteDescription is
called.</t> called.</t>
<t>In order to process rollback, a JSEP implementation abandons <t>In order to process rollback, a JSEP implementation abandons
the current offer/answer transaction, sets the signaling state the current offer/answer transaction, sets the signaling state
to "stable", and sets the pending local and/or remote to "stable", and sets the pending local and/or remote
description (see description (see Sections
<xref target="sec.pendinglocaldescription" /> and <xref target="sec.pendinglocaldescription" format="counter"/> and
<xref target="sec.pendingremotedescription" />) to null. Any <xref target="sec.pendingremotedescription" format="counter"/>) to "null
". Any
resources or candidates that were allocated by the abandoned resources or candidates that were allocated by the abandoned
local description are discarded; any media that is received is local description are discarded; any media that is received is
processed according to the previous local and remote processed according to the previous local and remote
descriptions.</t> descriptions.</t>
<t>A rollback disassociates any RtpTransceivers that were <t>A rollback disassociates any RtpTransceivers that were
associated with m= sections by the application of the associated with "m=" sections by the application of the
rolled-back session description (see rolled-back session description (see Sections
<xref target="sec.applying-a-remote-desc" /> and <xref target="sec.applying-a-remote-desc" format="counter"/> and
<xref target="sec.applying-a-local-desc" />). This means that <xref target="sec.applying-a-local-desc" format="counter"/>).
<!-- [rfced] Section 5.7: Please confirm that Section 5.9 is the
correct section to cite here. We easily found relevant text in
Section 5.10 but not in Section 5.9.
Original:
A rollback disassociates any RtpTransceivers that were associated
with m= sections by the application of the rolled-back session
description (see Section 5.10 and Section 5.9). -->
This means that
some RtpTransceivers that were previously associated will no some RtpTransceivers that were previously associated will no
longer be associated with any m= section; in such cases, the longer be associated with any "m=" section; in such cases, the
value of the RtpTransceiver's mid property MUST be set to null, value of the RtpTransceiver's mid property <bcp14>MUST</bcp14> be set to
and the mapping between the transceiver and its m= section "null",
index MUST be discarded. RtpTransceivers that were created by and the mapping between the transceiver and its "m=" section
applying a remote offer that was subsequently rolled back MUST index <bcp14>MUST</bcp14> be discarded. RtpTransceivers that were create
be stopped and removed from the PeerConnection. However, a d by
RtpTransceiver MUST NOT be removed if a track was attached to applying a remote offer that was subsequently rolled back <bcp14>MUST</b
cp14>
be stopped and removed from the PeerConnection. However, an
RtpTransceiver <bcp14>MUST NOT</bcp14> be removed if a track was attache
d to
the RtpTransceiver via the addTrack method. This is so that an the RtpTransceiver via the addTrack method. This is so that an
application may call addTrack, then call setRemoteDescription application may call addTrack, then call setRemoteDescription
with an offer, then roll back that offer, then call createOffer with an offer, then roll back that offer, then call createOffer
and have a m= section for the added track appear in the and have an "m=" section for the added track appear in the
generated offer.</t> generated offer.</t>
</section> </section>
<section title="Parsing a Session Description" <section anchor="sec.parsing-a-desc" numbered="true" toc="default">
anchor="sec.parsing-a-desc"> <name>Parsing a Session Description</name>
<t>The SDP contained in the session description object consists <t>The SDP contained in the session description object consists
of a sequence of text lines, each containing a key-value of a sequence of text lines, each containing a key-value
expression, as described in expression, as described in
<xref target="RFC4566" />, Section 5. The SDP is read, <xref target="RFC4566" sectionFormat="comma" section="5"/>. The SDP is r
line-by-line, and converted to a data structure that contains ead,
line by line, and converted to a data structure that contains
the deserialized information. However, SDP allows many types of the deserialized information. However, SDP allows many types of
lines, not all of which are relevant to JSEP applications. For lines, not all of which are relevant to JSEP applications. For
each line, the implementation will first ensure it is each line, the implementation will first ensure that it is
syntactically correct according to its defining ABNF, check syntactically correct according to its defining ABNF, check
that it conforms to that it conforms to the semantics used in
<xref target="RFC4566" /> and <xref target="RFC4566" format="default"/> and
<xref target="RFC3264" /> semantics, and then either parse and <xref target="RFC3264" format="default"/>, and then either parse and
store or discard the provided value, as described below.</t> store or discard the provided value, as described below.</t>
<t>If any line is not well formed or cannot be parsed as
<t>If any line is not well-formed, or cannot be parsed as described, the parser <bcp14>MUST</bcp14> stop with an error and reject
described, the parser MUST stop with an error and reject the the
session description, even if the value is to be discarded. This session description, even if the value is to be discarded. This
ensures that implementations do not accidentally misinterpret ensures that implementations do not accidentally misinterpret
ambiguous SDP.</t> ambiguous SDP.</t>
<section title="Session-Level Parsing" <section anchor="sec.session-level-parse" numbered="true" toc="default">
anchor="sec.session-level-parse"> <name>Session-Level Parsing</name>
<t>First, the session-level lines are checked and parsed. <t>First, the session-level lines are checked and parsed.
These lines MUST occur in a specific order, and with a These lines <bcp14>MUST</bcp14> occur in a specific order, and with a
specific syntax, as defined in specific syntax, as defined in
<xref target="RFC4566" />, Section 5. Note that while the <xref target="RFC4566" sectionFormat="comma" section="5"/>. Note that
specific line types (e.g. "v=", "c=") MUST occur in the while the
specific line types (e.g., "v=", "c=") <bcp14>MUST</bcp14> occur in th
e
defined order, lines of the same type (typically "a=") can defined order, lines of the same type (typically "a=") can
occur in any order.</t> occur in any order.</t>
<t>The following non-attribute lines are not meaningful in <t>The following non-attribute lines are not meaningful in
the JSEP context and MAY be discarded once they have been the JSEP context and <bcp14>MAY</bcp14> be discarded once they have be en
checked. checked.
<list> </t>
<ul spacing="normal">
<t>The "c=" line MUST be checked for syntax but its value <li>The "c=" line <bcp14>MUST</bcp14> be checked for syntax, but its
value
is only used for ICE mismatch detection, as defined in is only used for ICE mismatch detection, as defined in
<xref target="RFC8445" />, Section 5.4. Note that JSEP <xref target="RFC8445" sectionFormat="comma" section="5.4"/>. Note t hat JSEP
implementations should never encounter this condition implementations should never encounter this condition
because ICE is required for WebRTC.</t> because ICE is required for WebRTC.</li>
<li>The "i=", "u=", "e=", "p=", "t=", "r=", "z=", and "k="
lines are not used by this specification; they <bcp14>MUST</bcp14> b
e
checked for syntax, but their values are not used.
<t>The "i=", "u=", "e=", "p=", "t=", "r=", "z=", and "k=" <!-- [rfced] Section 5.8.1: We found this sentence confusing; should
lines are not used by this specification; they MUST be "t=" remain in this list of lines not used by this specification?
checked for syntax but their values are not used.</t>
</list></t>
<t>The remaining non-attribute lines are processed as Original:
follows: The "i=", "u=", "e=", "p=", "t=", "r=", "z=", and "k=" lines are
<list> not used by this specification; they MUST be checked for syntax
but their values are not used.
<t>The "v=" line MUST have a version of 0, as specified in We ask because we see (for example):
<xref target="RFC4566" />, Section 5.1.</t>
<t>The "o=" line MUST be parsed as specified in o A "t=" line MUST be added, as specified in [RFC4566], Section 5.9;
<xref target="RFC4566" />, Section 5.2.</t> both <start-time> and <stop-time> SHOULD be set to zero, e.g. "t=0
0".
...
o The "s=" and "t=" lines MUST stay the same.
...
t=0 0 -->
<t>The "b=" line, if present, MUST be parsed as specified </li>
</ul>
<t>The remaining non-attribute lines are processed as
follows:
</t>
<ul spacing="normal">
<li>The "v=" line <bcp14>MUST</bcp14> have a version of 0, as specif
ied in
<xref target="RFC4566" sectionFormat="comma" section="5.1"/>.</li>
<li>The "o=" line <bcp14>MUST</bcp14> be parsed as specified in
<xref target="RFC4566" sectionFormat="comma" section="5.2"/>.</li>
<li>The "b=" line, if present, <bcp14>MUST</bcp14> be parsed as spec
ified
in in
<xref target="RFC4566" />, Section 5.8, and the bwtype and <xref target="RFC4566" sectionFormat="comma" section="5.8"/>, and th
bandwidth values stored.</t> e bwtype and
</list></t> bandwidth values stored.</li>
</ul>
<t>Finally, the attribute lines are processed. Specific <t>Finally, the attribute lines are processed. Specific
processing MUST be applied for the following session-level processing <bcp14>MUST</bcp14> be applied for the following session-le vel
attribute ("a=") lines: attribute ("a=") lines:
<list style="symbols"> </t>
<ul spacing="normal">
<t>Any "a=group" lines are parsed as specified in <li>Any "a=group" lines are parsed as specified in
<xref target="RFC5888" />, Section 5, and the group's <xref target="RFC5888" sectionFormat="comma" section="5"/>, and the
semantics and mids are stored.</t> group's
semantics and mids are stored.</li>
<t>If present, a single "a=ice-lite" line is parsed as <li>If present, a single "a=ice-lite" line is parsed as
specified in
<xref target="I-D.ietf-mmusic-ice-sip-sdp" />,
Section 4.3, and a value
indicating the presence of ice-lite is stored.</t>
<t>If present, a single "a=ice-ufrag" line is parsed as
specified in specified in
<xref target="I-D.ietf-mmusic-ice-sip-sdp" />, <xref target="RFC8839" sectionFormat="comma" section="5.3"/>, and a
Section 4.4, and the ufrag value is stored.</t> value
indicating the presence of ice-lite is stored.</li>
<t>If present, a single "a=ice-pwd" line is parsed as <li>If present, a single "a=ice-ufrag" line is parsed as
specified in specified in
<xref target="I-D.ietf-mmusic-ice-sip-sdp" />, <xref target="RFC8839" sectionFormat="comma" section="5.4"/>, and th
Section 4.4, and the password value is stored.</t> e ufrag value is stored.</li>
<li>If present, a single "a=ice-pwd" line is parsed as
<t>If present, a single "a=ice-options" line is parsed as
specified in specified in
<xref target="I-D.ietf-mmusic-ice-sip-sdp" />, <xref target="RFC8839" sectionFormat="comma" section="5.4"/>, and th
Section 4.6, and the set of specified options is stored.</t> e password value is stored.</li>
<li>If present, a single "a=ice-options" line is parsed as
<t>Any "a=fingerprint" lines are parsed as specified in
<xref target="RFC8122" />, Section 5, and the set of
fingerprint and algorithm values is stored.</t>
<t>If present, a single "a=setup" line is parsed as
specified in specified in
<xref target="RFC4145" />, Section 4, and the setup value <xref target="RFC8839" sectionFormat="comma" section="5.6"/>, and th
is stored.</t> e set of specified options is stored.</li>
<li>Any "a=fingerprint" lines are parsed as specified in
<t>If present, a single "a=tls-id" line is parsed as <xref target="RFC8122" sectionFormat="comma" section="5"/>, and the
set of
fingerprint and algorithm values is stored.</li>
<li>If present, a single "a=setup" line is parsed as
specified in specified in
<xref target="I-D.ietf-mmusic-dtls-sdp" /> Section 5, and <xref target="RFC4145" sectionFormat="comma" section="4"/>, and the
the tls-id value is stored.</t> setup value
is stored.</li>
<t>Any "a=identity" lines are parsed and the identity <li>If present, a single "a=tls-id" line is parsed as
values stored for subsequent verification, as specified specified in <xref target="RFC8842" sectionFormat="comma" section="5
<xref target="I-D.ietf-rtcweb-security-arch" />, Section "/>, and
5.</t> the tls-id value is stored.</li>
<li>Any "a=identity" lines are parsed and the identity
<t>Any "a=extmap" lines are parsed as specified in values stored for subsequent verification, as specified in
<xref target="RFC5285" />, Section 5, and their values are <xref target="RFC8827" sectionFormat="comma" section="5"/>.</li>
stored.</t> <li>Any "a=extmap" lines are parsed as specified in
</list></t> <xref target="RFC5285" sectionFormat="comma" section="5"/>, and thei
r values are
stored.</li>
</ul>
<t>Other attributes that are not relevant to JSEP may also be <t>Other attributes that are not relevant to JSEP may also be
present, and implementations SHOULD process any that they present, and implementations <bcp14>SHOULD</bcp14> process any that th ey
recognize. As required by recognize. As required by
<xref target="RFC4566"></xref>, Section 5.13, unknown <xref target="RFC4566" sectionFormat="comma" section="5.13"/>, unknown
attribute lines MUST be ignored.</t> attribute lines <bcp14>MUST</bcp14> be ignored.</t>
<t>Once all the session-level lines have been parsed, <t>Once all the session-level lines have been parsed,
processing continues with the lines in m= sections.</t> processing continues with the lines in "m=" sections.</t>
</section> </section>
<section title="Media Section Parsing" <section anchor="sec.media-level-parse" numbered="true" toc="default">
anchor="sec.media-level-parse"> <name>Media Section Parsing</name>
<t>Like the session-level lines, the media section lines <bcp14>MUST</
<t>Like the session-level lines, the media section lines MUST bcp14>
occur in the specific order and with the specific syntax occur in the specific order and with the specific syntax
defined in defined in
<xref target="RFC4566" />, Section 5.</t> <xref target="RFC4566" sectionFormat="comma" section="5"/>.</t>
<t>The "m=" line itself <bcp14>MUST</bcp14> be parsed as described in
<t>The "m=" line itself MUST be parsed as described in <xref target="RFC4566" sectionFormat="comma" section="5.14"/>, and the
<xref target="RFC4566" />, Section 5.14, and the media, port, media, port,
proto, and fmt values stored.</t> proto, and fmt values stored.</t>
<t>Following the "m=" line, specific processing <bcp14>MUST</bcp14> be
<t>Following the "m=" line, specific processing MUST be
applied for the following non-attribute lines: applied for the following non-attribute lines:
<list style="symbols"> </t>
<ul spacing="normal">
<t>As with the "c=" line at the session level, the "c=" <li>As with the "c=" line at the session level, the "c="
line MUST be parsed according to line <bcp14>MUST</bcp14> be parsed according to
<xref target="RFC4566" />, Section 5.7, but its value is <xref target="RFC4566" sectionFormat="comma" section="5.7"/>, but it
not used.</t> s value is
not used.</li>
<t>The "b=" line, if present, MUST be parsed as specified <li>The "b=" line, if present, <bcp14>MUST</bcp14> be parsed as spec
ified
in in
<xref target="RFC4566" />, Section 5.8, and the bwtype and <xref target="RFC4566" sectionFormat="comma" section="5.8"/>, and th
bandwidth values stored.</t> e bwtype and
</list></t> bandwidth values stored.</li>
</ul>
<t>Specific processing MUST also be applied for the following <t>Specific processing <bcp14>MUST</bcp14> also be applied for the fol
lowing
attribute lines: attribute lines:
<list style="symbols"> </t>
<ul spacing="normal">
<t>If present, a single "a=ice-ufrag" line is parsed as <li>If present, a single "a=ice-ufrag" line is parsed as
specified in specified in
<xref target="I-D.ietf-mmusic-ice-sip-sdp" />, <xref target="RFC8839" sectionFormat="comma" section="5.4"/>, and th
Section 4.4, and the ufrag value is stored.</t> e ufrag value is stored.</li>
<li>If present, a single "a=ice-pwd" line is parsed as
<t>If present, a single "a=ice-pwd" line is parsed as
specified in specified in
<xref target="I-D.ietf-mmusic-ice-sip-sdp" />, <xref target="RFC8839" sectionFormat="comma" section="5.4"/>, and th
Section 4.4, and the password value is stored.</t> e password value is stored.</li>
<li>If present, a single "a=ice-options" line is parsed as
<t>If present, a single "a=ice-options" line is parsed as
specified in specified in
<xref target="I-D.ietf-mmusic-ice-sip-sdp" />, Section 4.6, <xref target="RFC8839" sectionFormat="comma" section="5.6"/>,
and the set of specified options is stored.</t> and the set of specified options is stored.</li>
<li>Any "a=candidate" attributes <bcp14>MUST</bcp14> be parsed as sp
<t>Any "a=candidate" attributes MUST be parsed as specified ecified
in in
<xref target="I-D.ietf-mmusic-ice-sip-sdp" />, <xref target="RFC8839" sectionFormat="comma" section="5.1"/>, and th
Section 4.1, and their values stored.</t> eir values stored.</li>
<li>Any "a=remote-candidates" attributes <bcp14>MUST</bcp14> be pars
<t>Any "a=remote-candidates" attributes MUST be parsed as ed as
specified in specified in
<xref target="I-D.ietf-mmusic-ice-sip-sdp" />, <xref target="RFC8839" sectionFormat="comma" section="5.2"/>, but th
Section 4.2, but their values are ignored.</t> eir values are ignored.</li>
<t>If present, a single "a=end-of-candidates" attribute
MUST be parsed as specified in
<xref target="I-D.ietf-ice-trickle" />, Section 8.2, and
its presence or absence flagged and stored.</t>
<t>Any "a=fingerprint" lines are parsed as specified in
<xref target="RFC8122" />, Section 5, and the set of
fingerprint and algorithm values is stored.</t>
</list></t>
<li>If present, a single "a=end-of-candidates" attribute
<bcp14>MUST</bcp14> be parsed as specified in
<xref target="RFC8840" sectionFormat="comma" section="8.1"/>, and
its presence or absence flagged and stored.</li>
<li>Any "a=fingerprint" lines are parsed as specified in
<xref target="RFC8122" sectionFormat="comma" section="5"/>, and the
set of
fingerprint and algorithm values is stored.</li>
</ul>
<t>If the "m=" proto value indicates use of RTP, as described <t>If the "m=" proto value indicates use of RTP, as described
in in
<xref target="sec.profile-names" /> above, the following <xref target="sec.profile-names" format="default"/> above, the followi
attribute lines MUST be processed: ng
<list style="symbols"> attribute lines <bcp14>MUST</bcp14> be processed:
</t>
<t>The "m=" fmt value MUST be parsed as specified in <ul spacing="normal">
<xref target="RFC4566" />, Section 5.14, and the individual <li>The "m=" fmt value <bcp14>MUST</bcp14> be parsed as specified in
values stored.</t> <xref target="RFC4566" sectionFormat="comma" section="5.14"/>, and t
he individual
<t>Any "a=rtpmap" or "a=fmtp" lines MUST be parsed as values stored.</li>
<li>Any "a=rtpmap" or "a=fmtp" lines <bcp14>MUST</bcp14> be parsed a
s
specified in specified in
<xref target="RFC4566" />, Section 6, and their values <xref target="RFC4566" sectionFormat="comma" section="6"/>, and thei
stored.</t> r values
stored.</li>
<t>If present, a single "a=ptime" line MUST be parsed as <li>If present, a single "a=ptime" line <bcp14>MUST</bcp14> be parse
d as
described in described in
<xref target="RFC4566" />, Section 6, and its value <xref target="RFC4566" sectionFormat="comma" section="6"/>, and its
stored.</t> value
stored.</li>
<t>If present, a single "a=maxptime" line MUST be parsed as <li>If present, a single "a=maxptime" line <bcp14>MUST</bcp14> be pa
rsed as
described in described in
<xref target="RFC4566" />, Section 6, and its value <xref target="RFC4566" sectionFormat="comma" section="6"/>, and its
stored.</t> value
stored.</li>
<t>If present, a single direction attribute line (e.g. <li>If present, a single direction attribute line (e.g.,
"a=sendrecv") MUST be parsed as described in "a=sendrecv") <bcp14>MUST</bcp14> be parsed as described in
<xref target="RFC4566" />, Section 6, and its value <xref target="RFC4566" sectionFormat="comma" section="6"/>, and its
stored.</t> value
stored.</li>
<t>Any "a=ssrc" attributes MUST be parsed as specified in <li>Any "a=ssrc" attributes <bcp14>MUST</bcp14> be parsed as specifi
<xref target="RFC5576" />, Section 4.1, and their values ed in
stored.</t> <xref target="RFC5576" sectionFormat="comma" section="4.1"/>, and th
eir values
<t>Any "a=extmap" attributes MUST be parsed as specified in stored.</li>
<li>Any "a=extmap" attributes <bcp14>MUST</bcp14> be parsed as speci
<xref target="RFC5285" />, Section 5, and their values fied in
stored.</t>
<t>Any "a=rtcp-fb" attributes MUST be parsed as specified <xref target="RFC5285" sectionFormat="comma" section="5"/>, and thei
r values
stored.</li>
<li>Any "a=rtcp-fb" attributes <bcp14>MUST</bcp14> be parsed as spec
ified
in in
<xref target="RFC4585" />, Section 4.2., and their values <xref target="RFC4585" sectionFormat="comma" section="4.2"/>, and th
stored.</t> eir values
stored.</li>
<t>If present, a single "a=rtcp-mux" attribute MUST be <li>If present, a single "a=rtcp-mux" attribute <bcp14>MUST</bcp14>
be
parsed as specified in parsed as specified in
<xref target="RFC5761"></xref>, Section 5.1.3, and its <xref target="RFC5761" sectionFormat="comma" section="5.1.3"/>, and
presence or absence flagged and stored.</t> its
presence or absence flagged and stored.</li>
<t>If present, a single "a=rtcp-mux-only" attribute MUST be <li>If present, a single "a=rtcp-mux-only" attribute <bcp14>MUST</bc
p14> be
parsed as specified in parsed as specified in
<xref target="I-D.ietf-mmusic-mux-exclusive" />, Section 3, <xref target="RFC8858" sectionFormat="comma" section="3"/>,
and its presence or absence flagged and stored.</t> and its presence or absence flagged and stored.</li>
<li>If present, a single "a=rtcp-rsize" attribute <bcp14>MUST</bcp14
<t>If present, a single "a=rtcp-rsize" attribute MUST be > be
parsed as specified in parsed as specified in
<xref target="RFC5506" />, Section 5, and its presence or <xref target="RFC5506" sectionFormat="comma" section="5"/>, and its
absence flagged and stored.</t> presence or
absence flagged and stored.</li>
<t>If present, a single "a=rtcp" attribute MUST be parsed <li>If present, a single "a=rtcp" attribute <bcp14>MUST</bcp14> be p
arsed
as specified in as specified in
<xref target="RFC3605" />, Section 2.1, but its value is <xref target="RFC3605" sectionFormat="comma" section="2.1"/>, but it s value is
ignored, as this information is superfluous when using ignored, as this information is superfluous when using
ICE.</t> ICE.</li>
<li>If present, "a=msid" attributes <bcp14>MUST</bcp14> be parsed as
<t>If present, "a=msid" attributes MUST be parsed as
specified in specified in
<xref target="I-D.ietf-mmusic-msid" />, Section 3.2, and <xref target="RFC8830" sectionFormat="comma" section="3.2"/>, and
their values stored, ignoring any "appdata" field. If no "a=msid" their values stored, ignoring any "appdata" field. If no "a=msid"
attributes are present, a random msid-id value is generated for a attributes are present, a random msid-id value is generated for a
"default" MediaStream for the session, if not already present, and "default" MediaStream for the session, if not already present, and
this value is stored.</t> this value is stored.</li>
<li>Any "a=imageattr" attributes <bcp14>MUST</bcp14> be parsed as sp
<t>Any "a=imageattr" attributes MUST be parsed as specified ecified
in in
<xref target="RFC6236" />, Section 3, and their values <xref target="RFC6236" sectionFormat="comma" section="3"/>, and thei
stored.</t> r values
stored.</li>
<t>Any "a=rid" lines MUST be parsed as specified in <li>Any "a=rid" lines <bcp14>MUST</bcp14> be parsed as specified in
<xref target="I-D.ietf-mmusic-rid"></xref>, Section 10, and <xref target="RFC8851" sectionFormat="comma" section="10"/>, and
their values stored.</t> their values stored.</li>
<li>If present, a single "a=simulcast" line <bcp14>MUST</bcp14> be p
<t>If present, a single "a=simulcast" line MUST be parsed arsed
as specified in as specified in
<xref target="I-D.ietf-mmusic-sdp-simulcast"></xref>, and <xref target="RFC8853" format="default"/>, and
its values stored.</t> its values stored.</li>
</list></t> </ul>
<t>Otherwise, if the "m=" proto value indicates use of SCTP, <t>Otherwise, if the "m=" proto value indicates use of SCTP,
the following attribute lines MUST be processed: the following attribute lines <bcp14>MUST</bcp14> be processed:
<list style="symbols"> </t>
<ul spacing="normal">
<t>The "m=" fmt value MUST be parsed as specified in <li>The "m=" fmt value <bcp14>MUST</bcp14> be parsed as specified in
<xref target="I-D.ietf-mmusic-sctp-sdp" />, Section 4.3, <xref target="RFC8841" sectionFormat="comma" section="4.3"/>,
and the application protocol value stored.</t> and the application protocol value stored.</li>
<li>An "a=sctp-port" attribute <bcp14>MUST</bcp14> be present, and i
<t>An "a=sctp-port" attribute MUST be present, and it MUST t <bcp14>MUST</bcp14>
be parsed as specified in be parsed as specified in
<xref target="I-D.ietf-mmusic-sctp-sdp" />, Section 5.2, <xref target="RFC8841" sectionFormat="comma" section="5.2"/>,
and the value stored.</t> and the value stored.</li>
<li>If present, a single "a=max-message-size" attribute <bcp14>MUST<
<t>If present, a single "a=max-message-size" attribute MUST /bcp14>
be parsed as specified in be parsed as specified in
<xref target="I-D.ietf-mmusic-sctp-sdp" />, Section 6, and <xref target="RFC8841" sectionFormat="comma" section="6"/>, and
the value stored. Otherwise, use the specified default.</t> the value stored. Otherwise, use the specified default.</li>
</list></t> </ul>
<t>Other attributes that are not relevant to JSEP may also be <t>Other attributes that are not relevant to JSEP may also be
present, and implementations SHOULD process any that they present, and implementations <bcp14>SHOULD</bcp14> process any that th ey
recognize. As required by recognize. As required by
<xref target="RFC4566"></xref>, Section 5.13, unknown <xref target="RFC4566" sectionFormat="comma" section="5.13"/>, unknown
attribute lines MUST be ignored.</t> attribute lines <bcp14>MUST</bcp14> be ignored.</t>
</section> </section>
<section title="Semantics Verification"> <section numbered="true" toc="default">
<name>Semantics Verification</name>
<t>Assuming parsing completes successfully, the parsed <t>Assuming that parsing completes successfully, the parsed
description is then evaluated to ensure internal consistency description is then evaluated to ensure internal consistency
as well as proper support for mandatory features. as well as proper support for mandatory features.
Specifically, the following checks are performed: Specifically, the following checks are performed:
<list style="symbols"> </t>
<ul spacing="normal">
<t>For each m= section, valid values for each of the <li>
<t>For each "m=" section, valid values for each of the
mandatory-to-use features enumerated in mandatory-to-use features enumerated in
<xref target="sec.usage-requirements" /> MUST be present. <xref target="sec.usage-requirements" format="default"/> <bcp14>MUST
These values MAY either be present at the media level, or </bcp14> be present.
These values <bcp14>MAY</bcp14> be either present at the media level
or
inherited from the session level. inherited from the session level.
<list style="symbols"> </t>
<ul spacing="normal">
<t>ICE ufrag and password values, which MUST comply with <li>ICE ufrag and password values, which <bcp14>MUST</bcp14> com
ply with
the size limits specified in the size limits specified in
<xref target="I-D.ietf-mmusic-ice-sip-sdp" />, Section 4.4.</t> <xref target="RFC8839" sectionFormat="comma" section="5.4"/>.</li>
<li>A tls-id value, which <bcp14>MUST</bcp14> be set according t
<t>tls-id value, which MUST be set according to o
<xref target="I-D.ietf-mmusic-dtls-sdp" />, Section 5. If <xref target="RFC8842" sectionFormat="comma" section="5"/>. If
this is a re-offer or a response to a re-offer and the this is a re-offer or a response to a re-offer and the
tls-id value is different from that presently in use, the tls-id value is different from that presently in use, the
DTLS connection is not being continued and the remote DTLS connection is not being continued and the remote
description MUST be part of an ICE restart, together with description <bcp14>MUST</bcp14> be part of an ICE restart, togethe
new ufrag and password values.</t> r with
new ufrag and password values.</li>
<t>DTLS setup value, which MUST be set according to the <li>A DTLS setup value, which <bcp14>MUST</bcp14> be set accordi
rules specified in ng to the
<xref target="RFC5763" />, Section 5 and MUST be rules specified in
consistent with the selected role of the current DTLS <xref target="RFC5763" sectionFormat="comma" section="5"/> and <bc
connection, if one exists and is being continued.</t> p14>MUST</bcp14> be
consistent with the selected role of the current DTLS
<t>DTLS fingerprint values, where at least one connection, if one exists and is being continued.</li>
fingerprint MUST be present.</t> <li>DTLS fingerprint values, where at least one
</list></t> fingerprint <bcp14>MUST</bcp14> be present.</li>
</ul>
<t>All RID values referenced in an "a=simulcast" line MUST </li>
exist as "a=rid" lines.</t> <li>All RID values referenced in an "a=simulcast" line <bcp14>MUST</
bcp14>
<t>Each m= section is also checked to ensure prohibited exist as "a=rid" lines.</li>
features are not used.</t> <li>Each "m=" section is also checked to ensure that prohibited
features are not used.</li>
<t>If the RTP/RTCP multiplexing policy is "require", each <li>If the RTP/RTCP multiplexing policy is "require", each
m= section MUST contain an "a=rtcp-mux" attribute. If an m= "m=" section <bcp14>MUST</bcp14> contain an "a=rtcp-mux" attribute.
section contains an "a=rtcp-mux-only" attribute, that If an "m="
section MUST also contain an "a=rtcp-mux" attribute.</t> section contains an "a=rtcp-mux-only" attribute, that
section <bcp14>MUST</bcp14> also contain an "a=rtcp-mux" attribute.<
<t>If an m= section was present in the previous answer, the /li>
state of RTP/RTCP multiplexing MUST match what was <li>If an "m=" section was present in the previous answer, the
previously negotiated.</t> state of RTP/RTCP multiplexing <bcp14>MUST</bcp14> match what was
</list></t> previously negotiated.</li>
</ul>
<t>If this session description is of type "pranswer" or <t>If this session description is of type "pranswer" or
"answer", the following additional checks are applied: "answer", the following additional checks are applied:
<list style="symbols"> </t>
<ul spacing="normal">
<t>The session description must follow the rules defined in <li>The session description must follow the rules defined in
<xref target="RFC3264" />, Section 6, including the
requirement that the number of m= sections MUST exactly
match the number of m= sections in the associated
offer.</t>
<t>For each m= section, the media type and protocol values
MUST exactly match the media type and protocol values in
the corresponding m= section in the associated offer.</t>
</list></t>
<t>If any of the preceding checks failed, processing MUST
stop and an error MUST be returned.</t>
</section>
</section>
<section title="Applying a Local Description"
anchor="sec.applying-a-local-desc">
<t>The following steps are performed at the media engine level
to apply a local description. If an error is returned, the
session MUST be restored to the state it was in before
performing these steps.</t>
<t>First, m= sections are processed. For each m= section, the
following steps MUST be performed; if any parameters are out of
bounds, or cannot be applied, processing MUST stop and an error
MUST be returned.
<list style="symbols">
<t>If this m= section is new, begin gathering candidates for
it, as defined in
<xref target="RFC8445" />, Section 5.1.1, unless it is
definitively being bundled (either this is an offer and the
m= section is marked bundle-only, or it is an answer and the
m= section is bundled into into another m= section.)</t>
<t>Or, if the ICE ufrag and password values have changed,
trigger the ICE agent to start an ICE restart as described in
<xref target="RFC8445" />, Section 9, and begin
gathering new candidates for the m= section. If this
description is an answer, also start checks on that media
section.</t>
<t>If the m= section proto value indicates use of RTP:
<list style="symbols">
<t>If there is no RtpTransceiver associated with this m=
section, find one and associate it with this m= section
according to the following steps. Note that this situation
will only occur when applying an offer.
<list style="symbols">
<t>Find the RtpTransceiver that corresponds to this m=
section, using the mapping between transceivers and m=
section indices established when creating the offer.</t>
<t>Set the value of this RtpTransceiver's mid property to
the MID of the m= section.</t>
</list></t>
<t>If RTCP mux is indicated, prepare to demux RTP and RTCP
from the RTP ICE component, as specified in
<xref target="RFC5761" />, Section 5.1.3.</t>
<t>For each specified RTP header extension, establish a
mapping between the extension ID and URI, as described in
<xref target="RFC5285" />, Section 6.</t>
<t>If the MID header extension is supported, prepare to
demux RTP streams intended for this m= section based on the
MID header extension, as described in
<xref target="I-D.ietf-mmusic-sdp-bundle-negotiation" />,
Section 15.</t>
<t>For each specified media format, establish a mapping
between the payload type and the actual media format, as
described in
<xref target="RFC3264" />, Section 6.1. In addition,
prepare to demux RTP streams intended for this m= section
based on the media formats supported by this m= section, as
described in
<xref target="I-D.ietf-mmusic-sdp-bundle-negotiation" />,
Section 10.2.</t>
<t>For each specified "rtx" media format, establish a
mapping between the RTX payload type and its associated
primary payload type, as described in
<xref target="RFC4588" />, Sections 8.6 and 8.7.</t>
<t>If the directional attribute is of type "sendrecv" or
"recvonly", enable receipt and decoding of media.</t>
</list></t>
</list></t>
<t>Finally, if this description is of type "pranswer" or
"answer", follow the processing defined in
<xref target="sec.applying-an-answer" /> below.</t>
</section>
<section title="Applying a Remote Description"
anchor="sec.applying-a-remote-desc">
<t>The following steps are performed to apply a remote
description. If an error is returned, the session MUST be
restored to the state it was in before performing these
steps.</t>
<t>If the answer contains any "a=ice-options" attributes where
"trickle" is listed as an attribute, update the PeerConnection
canTrickle property to be true. Otherwise, set this property to
false.</t>
<t>The following steps MUST be performed for attributes at the
session level; if any parameters are out of bounds, or cannot
be applied, processing MUST stop and an error MUST be returned.
<list style="symbols">
<t>For any specified "CT" bandwidth value, set this as the
limit for the maximum total bitrate for all m= sections, as
specified in
<xref target="RFC4566"></xref>, Section 5.8. Within this
overall limit, the implementation can dynamically decide how
to best allocate the available bandwidth between m= sections,
respecting any specific limits that have been specified for
individual m= sections.</t>
<t>For any specified "RR" or "RS" bandwidth values, handle as
specified in
<xref target="RFC3556"></xref>, Section 2.</t>
<t>Any "AS" bandwidth value MUST be ignored, as the meaning
of this construct at the session level is not well
defined.</t>
</list></t>
<t>For each m= section, the following steps MUST be performed;
if any parameters are out of bounds, or cannot be applied,
processing MUST stop and an error MUST be returned.
<list style="symbols">
<t>If the ICE ufrag or password changed from the previous
remote description:
<list style="symbols">
<t>If the description is of type "offer", the
implementation MUST note that an ICE restart is needed, as
described in
<xref target="I-D.ietf-mmusic-ice-sip-sdp" />,
Section 3.4.1.1.1</t>
<t>If the description is of type "answer" or "pranswer",
then check to see if the current local description is an
ICE restart, and if not, generate an error. If the
PeerConnection state is "have-remote-pranswer", and the ICE
ufrag or password changed from the previous provisional
answer, then signal the ICE agent to discard any previous
ICE check list state for the m= section. Finally, signal
the ICE agent to begin checks.</t>
</list></t>
<t>If the current local description indicates an ICE restart,
and either the ICE ufrag or password has not changed from the
previous remote description, as prescribed by
<xref target="RFC8445" />, Section 9, generate an
error.</t>
<t>Configure the ICE components associated with this media
section to use the supplied ICE remote ufrag and password for
their connectivity checks.</t>
<t>Pair any supplied ICE candidates with any gathered local
candidates, as described in
<xref target="RFC8445" />, Section 6.1.2, and start
connectivity checks with the appropriate credentials.</t>
<t>If an "a=end-of-candidates" attribute is present, process
the end-of-candidates indication as described in
<xref target="I-D.ietf-ice-trickle" />, Section 11.</t>
<t>If the m= section proto value indicates use of RTP:
<list style="symbols">
<t>If the m= section is being recycled (see
<xref target="sec.subsequent-offers"></xref>), dissociate
the currently associated RtpTransceiver by setting its mid
property to null, and discard the mapping between the
transceiver and its m= section index.</t>
<t>If the m= section is not associated with any
RtpTransceiver (possibly because it was dissociated in the
previous step), either find an RtpTransceiver or create one
according to the following steps:
<list style="symbols">
<t>If the m= section is sendrecv or recvonly, and there
are RtpTransceivers of the same type that were added to
the PeerConnection by addTrack and are not associated
with any m= section and are not stopped, find the first
(according to the canonical order described in
<xref target="sec.initial-offers" />) such
RtpTransceiver.</t>
<t>If no RtpTransceiver was found in the previous step,
create one with a recvonly direction.</t>
<t>Associate the found or created RtpTransceiver with the
m= section by setting the value of the RtpTransceiver's
mid property to the MID of the m= section, and establish
a mapping between the transceiver and the index of the m=
section. If the m= section does not include a MID (i.e.,
the remote endpoint does not support the MID extension),
generate a value for the RtpTransceiver mid property,
following the guidance for "a=mid" mentioned in
<xref target="sec.initial-offers" />.</t>
</list></t>
<t>For each specified media format that is also supported
by the local implementation, establish a mapping between
the specified payload type and the media format, as
described in
<xref target="RFC3264" />, Section 6.1. Specifically, this
means that the implementation records the payload type to
be used in outgoing RTP packets when sending each specified
media format, as well as the relative preference for each
format that is indicated in their ordering. If any
indicated media format is not supported by the local
implementation, it MUST be ignored.</t>
<t>For each specified "rtx" media format, establish a
mapping between the RTX payload type and its associated
primary payload type, as described in
<xref target="RFC4588" />, Section 4. If any referenced
primary payload types are not present, this MUST result in
an error. Note that RTX payload types may refer to primary
payload types which are not supported by the local media
implementation, in which case, the RTX payload type MUST
also be ignored.</t>
<t>For each specified fmtp parameter that is supported by
the local implementation, enable them on the associated
media formats.</t>
<t>For each specified SSRC that is signaled in the m=
section, prepare to demux RTP streams intended for this m=
section using that SSRC, as described in
<xref target="I-D.ietf-mmusic-sdp-bundle-negotiation" />,
Section 10.2.</t>
<t>For each specified RTP header extension that is also
supported by the local implementation, establish a mapping
between the extension ID and URI, as described in
<xref target="RFC5285" />, Section 5. Specifically, this
means that the implementation records the extension ID to
be used in outgoing RTP packets when sending each specified
header extension. If any indicated RTP header extension is
not supported by the local implementation, it MUST be
ignored.</t>
<t>For each specified RTCP feedback mechanism that is
supported by the local implementation, enable them on the
associated media formats.</t>
<t>For any specified "TIAS" bandwidth value, set this value
as a constraint on the maximum RTP bitrate to be used when
sending media, as specified in
<xref target="RFC3890"></xref>. If a "TIAS" value is not
present, but an "AS" value is specified, generate a "TIAS"
value using this formula:
<list style="format">
<t>TIAS = AS * 1000 * 0.95 - (50 * 40 * 8)</t>
</list>The 50 is based on 50 packets per second, the 40 is
based on an estimate of total header size, the 1000 changes
the unit from kbps to bps (as required by TIAS), and the
0.95 is to allocate 5% to RTCP. "TIAS" is used in
preference to "AS" because it provides more accurate
control of bandwidth.</t>
<t>For any "RR" or "RS" bandwidth values, handle as
specified in
<xref target="RFC3556"></xref>, Section 2.</t>
<t>Any specified "CT" bandwidth value MUST be ignored, as <xref target="RFC3264" sectionFormat="comma" section="6"/>, includin
the meaning of this construct at the media level is not g the
well defined.</t> requirement that the number of "m=" sections <bcp14>MUST</bcp14> exa
ctly
match the number of "m=" sections in the associated
offer.</li>
<li>For each "m=" section, the media type and protocol values
<bcp14>MUST</bcp14> exactly match the media type and protocol values
in
the corresponding "m=" section in the associated offer.</li>
</ul>
<t>If any of the preceding checks failed, processing <bcp14>MUST</bcp1
4>
stop and an error <bcp14>MUST</bcp14> be returned.</t>
</section>
</section>
<section anchor="sec.applying-a-local-desc" numbered="true" toc="default"
>
<name>Applying a Local Description</name>
<t>The following steps are performed at the media engine level
to apply a local description. If an error is returned, the
session <bcp14>MUST</bcp14> be restored to the state it was in before
performing these steps.</t>
<t>First, "m=" sections are processed. For each "m=" section, the
following steps <bcp14>MUST</bcp14> be performed; if any parameters are
out of
bounds or cannot be applied, processing <bcp14>MUST</bcp14> stop and an
error
<bcp14>MUST</bcp14> be returned.
</t>
<ul spacing="normal">
<li>If this "m=" section is new, begin gathering candidates for
it, as defined in
<xref target="RFC8445" sectionFormat="comma" section="5.1.1"/>, unless
it is
definitively being bundled (either (1) this is an offer and the
"m=" section is marked bundle-only or (2)&nbsp;it is an answer and the
"m=" section is bundled into another "m=" section).</li>
<li>Or, if the ICE ufrag and password values have changed,
trigger the ICE agent to start an ICE restart as described in
<xref target="RFC8445" sectionFormat="comma" section="9"/>, and begin
gathering new candidates for the "m=" section. If this
description is an answer, also start checks on that media
section.</li>
<li>
<t>If the "m=" section proto value indicates use of RTP:
</t>
<ul spacing="normal">
<li>
<t>If there is no RtpTransceiver associated with this "m="
section, find one and associate it with this "m=" section
according to the following steps. Note that this situation
will only occur when applying an offer.
</t>
<ul spacing="normal">
<li>Find the RtpTransceiver that corresponds to this "m="
section, using the mapping between transceivers and "m="
section indices established when creating the offer.</li>
<li>Set the value of this RtpTransceiver's mid property to
the MID of the "m=" section.</li>
</ul>
</li>
<li>If RTCP mux is indicated, prepare to demux RTP and RTCP
from the RTP ICE component, as specified in
<xref target="RFC5761" sectionFormat="comma" section="5.1.3"/>.</li>
<li>For each specified RTP header extension, establish a
mapping between the extension ID and URI, as described in
<xref target="RFC5285" sectionFormat="comma" section="6"/>.</li>
<li>If the MID header extension is supported, prepare to
demux RTP streams intended for this "m=" section based on the
MID header extension, as described in
<xref target="RFC8843" sectionFormat="comma" section="15"/>.</li>
<li>For each specified media format, establish a mapping
between the payload type and the actual media format, as
described in
<xref target="RFC3264" sectionFormat="comma" section="6.1"/>. In add
ition,
prepare to demux RTP streams intended for this "m=" section
based on the media formats supported by this "m=" section, as
described in
<xref target="RFC8843" sectionFormat="comma" section="9.2"/>.
<t>If the m= section is of type audio: <!-- [rfced] Sections 5.9, 5.10, 5.11, and 6:
<list style="symbols"> RFC 8843 [I-D.ietf-mmusic-sdp-bundle-negotiation] does not have a Section 8.2
or a Section 10.2. We have updated these to refer to 7.2 and 9.2,
respectively. Please review.
<t>For each specified "CN" media format, configure Original:
silence suppression for all supported media formats with In addition, prepare to demux RTP
the same clockrate, as described in streams intended for this m= section based on the media formats
<xref target="RFC3389" />, Section 5, except for formats supported by this m= section, as described in
that have their own internal silence suppression [I-D.ietf-mmusic-sdp-bundle-negotiation], Section 10.2.
mechanisms. Silence suppression for such formats (e.g., ...
Opus) is controlled via fmtp parameters, as discussed in * For each specified SSRC that is signaled in the m= section,
<xref target="sec.voiceactivitydetection1" />.</t> prepare to demux RTP streams intended for this m= section using
that SSRC, as described in
[I-D.ietf-mmusic-sdp-bundle-negotiation], Section 10.2.
...
If the answer contains valid bundle groups, discard any ICE
components for the m= sections that will be bundled onto the primary
ICE components in each bundle, and begin muxing these m= sections
accordingly, as described in
[I-D.ietf-mmusic-sdp-bundle-negotiation], Section 8.2.
...
When bundling, associating incoming RTP/RTCP with the proper m=
section is defined in [I-D.ietf-mmusic-sdp-bundle-negotiation],
Section 10.2. -->
<t>For each specified "telephone-event" media format, </li>
enable DTMF transmission for all supported media formats <li>For each specified "rtx" media format, establish a
with the same clockrate, as described in mapping between the RTX payload type and its associated
<xref target="RFC4733" />, Section 2.5.1.2. If there are primary payload type, as described in
any supported media formats that do not have a Sections <xref target="RFC4588" section="8.6"
corresponding telephone-event format, disable DTMF sectionFormat="bare"/> and <xref target="RFC4588" section="8.7"
transmission for those formats.</t> sectionFormat="bare"/> of <xref target="RFC4588"/>.
<t>For any specified "ptime" value, configure the <!-- [rfced] Sections 5.9 and 5.11: We had to change "[RFC4588],
available media formats to use the specified packet size Sections 8.6 and 8.7" to "Sections 8.6 and 8.7 of [RFC4588]"
when sending. If the specified size is not supported for (in Section 5.9) and "[RFC3264], Sections 6.1 and 7" to "Sections 6.1
a media format, use the next closest value instead.</t> and 7 of [RFC3264]" (in Section 5.11) in order to get the hyperlinks
</list></t> to work properly in the .html/.pdf files. Please let us know any concerns.
</list></t>
</list></t>
<t>Finally, if this description is of type "pranswer" or Original:
"answer", follow the processing defined in * For each specified "rtx" media format, establish a mapping
<xref target="sec.applying-an-answer" /> below.</t> between the RTX payload type and its associated primary payload
</section> type, as described in [RFC4588], Sections 8.6 and 8.7.
<section title="Applying an Answer" ...
anchor="sec.applying-an-answer"> * If the directional attribute in the answer indicates that the
JSEP implementation should be sending media ("sendonly" for
local answers, "recvonly" for remote answers, or "sendrecv" for
either type of answer), choose the media format to send as the
most preferred media format from the remote description that is
also locally supported, as discussed in [RFC3264], Sections 6.1
and 7, and start transmitting RTP media using that format once
the underlying transport layers have been established.
<t>In addition to the steps mentioned above for processing a Currently:
local or remote description, the following steps are performed - For each specified "rtx" media format, establish a mapping
when processing a description of type "pranswer" or between the RTX payload type and its associated primary payload
"answer".</t> type, as described in Sections 8.6 and 8.7 of [RFC4588].
...
- If the directional attribute in the answer indicates that the
JSEP implementation should be sending media ("sendonly" for
local answers, "recvonly" for remote answers, or "sendrecv" for
either type of answer), choose the media format to send as the
most preferred media format from the remote description that is
also locally supported, as discussed in Sections 6.1 and 7 of
[RFC3264], and start transmitting RTP media using that format
once the underlying transport layers have been established. -->
<t>For each m= section, the following steps MUST be performed: </li>
<list style="symbols"> <li>If the directional attribute is of type "sendrecv" or
"recvonly", enable receipt and decoding of media.</li>
</ul>
</li>
</ul>
<t>Finally, if this description is of type "pranswer" or
"answer", follow the processing defined in
<xref target="sec.applying-an-answer" format="default"/> below.</t>
</section>
<section anchor="sec.applying-a-remote-desc" numbered="true" toc="default
">
<name>Applying a Remote Description</name>
<t>The following steps are performed to apply a remote
description. If an error is returned, the session <bcp14>MUST</bcp14> be
restored to the state it was in before performing these
steps.</t>
<t>If the answer contains any "a=ice-options" attributes where
"trickle" is listed as an attribute, update the PeerConnection
canTrickle property to be "true". Otherwise, set this property to
"false".</t>
<t>The following steps <bcp14>MUST</bcp14> be performed for attributes a
t the
session level; if any parameters are out of bounds or cannot
be applied, processing <bcp14>MUST</bcp14> stop and an error <bcp14>MUST
</bcp14> be returned.
<t>If the m= section has been rejected (i.e. port is set to </t>
zero in the answer), stop any reception or transmission of <ul spacing="normal">
media for this section, and, unless a non-rejected m= section <li>For any specified "CT" bandwidth value, set this value as the
is bundled with this m= section, discard any associated ICE limit for the maximum total bitrate for all "m=" sections, as
components, as described in specified in
<xref target="I-D.ietf-mmusic-ice-sip-sdp" />, Section 3.4.3.1.</t> <xref target="RFC4566" sectionFormat="comma" section="5.8"/>. Within t
his
overall limit, the implementation can dynamically decide how
to best allocate the available bandwidth between "m=" sections,
respecting any specific limits that have been specified for
individual "m=" sections.</li>
<li>For any specified "RR" or "RS" bandwidth values, handle as
specified in
<xref target="RFC3556" sectionFormat="comma" section="2"/>.</li>
<li>Any "AS" bandwidth value <bcp14>MUST</bcp14> be ignored, as the me
aning
of this construct at the session level is not well
defined.
<t>If the remote DTLS fingerprint has been changed or the <!-- [rfced] Section 5.10: For ease of the reader, we suggest adding
tls-id has changed, tear down the DTLS connection. This a citation that provides the definition of "AS." Please let us know
includes the case when the PeerConnection state is if we may update as follows.
"have-remote-pranswer". If a DTLS connection needs to be torn
down but the answer does not indicate an ICE restart or, in
the case of "have-remote-pranswer", new ICE credentials, an
error MUST be generated. If an ICE restart is performed
without a change in tls-id or fingerprint, then the same DTLS
connection is continued over the new ICE channel. Note that
although JSEP requires that answerers change the tls-id value
if and only if the offerer does, non-JSEP answerers are
permitted to change the tls-id as long as the offer contained
an ICE restart. Thus, JSEP implementations which process DTLS
data prior to receiving an answer MUST be prepared to receive
either a ClientHello or data from the previous DTLS
connection.</t>
<t>If no valid DTLS connection exists, prepare to start a Original:
DTLS connection, using the specified roles and fingerprints, o Any "AS" bandwidth value MUST be ignored, as the meaning of this
on any underlying ICE components, once they are active.</t> construct at the session level is not well defined.
<t>If the m= section proto value indicates use of RTP: Suggested:
<list style="symbols"> * Any "AS" bandwidth value ([RFC4566], Section 5.8) MUST be
ignored, as the meaning of this construct at the session level is
not well defined. -->
<t>If the m= section references RTCP feedback mechanisms </li>
that were not present in the corresponding m= section in </ul>
the offer, this indicates a negotiation problem and MUST <t>For each "m=" section, the following steps <bcp14>MUST</bcp14> be per
result in an error. However, new media formats and new RTP formed;
header extension values are permitted in the answer, as if any parameters are out of bounds or cannot be applied,
described in processing <bcp14>MUST</bcp14> stop and an error <bcp14>MUST</bcp14> be
<xref target="RFC3264" />, Section 7, and returned.
<xref target="RFC5285" />, Section 6.</t> </t>
<ul spacing="normal">
<li>
<t>If the ICE ufrag or password changed from the previous
remote description:
</t>
<ul spacing="normal">
<li>If the description is of type "offer", the
implementation <bcp14>MUST</bcp14> note that an ICE restart is neede
d, as
described in
<xref target="RFC8839" sectionFormat="comma" section="4.4.1.1.1"/>.<
/li>
<li>If the description is of type "answer" or "pranswer",
then check to see if the current local description is an
ICE restart, and if not, generate an error. If the
PeerConnection state is "have-remote-pranswer" and the ICE
ufrag or password changed from the previous provisional
answer, then signal the ICE agent to discard any previous
ICE check list state for the "m=" section. Finally, signal
the ICE agent to begin checks.
<t>If the m= section has RTCP mux enabled, discard the RTCP <!-- [rfced] Other documents in this cluster spell "checklist" as one
ICE component, if one exists, and begin or continue muxing word. May we change "check list" in this document to "checklist"? -->
RTCP over the RTP ICE component, as specified in
<xref target="RFC5761" />, Section 5.1.3. Otherwise,
prepare to transmit RTCP over the RTCP ICE component; if no
RTCP ICE component exists, because RTCP mux was previously
enabled, this MUST result in an error.</t>
<t>If the m= section has reduced-size RTCP enabled, </li>
configure the RTCP transmission for this m= section to use </ul>
reduced-size RTCP, as specified in </li>
<xref target="RFC5506" />.</t> <li>If the current local description indicates an ICE restart
and either the ICE ufrag or password has not changed from the
previous remote description, as prescribed by
<xref target="RFC8445" sectionFormat="comma" section="9"/>, generate a
n
error.
<t>If the directional attribute in the answer indicates <!-- [rfced] Section 5.10: We found this sentence confusing, as we
that the JSEP implementation should be sending media could not tell what "as prescribed by [RFC8445], Section 9" and
("sendonly" for local answers, "recvonly" for remote "generate an error" refer to. Please confirm that the citation is
answers, or "sendrecv" for either type of answer), choose correct and will be clear to readers.
the media format to send as the most preferred media format
from the remote description that is also locally supported,
as discussed in
<xref target="RFC3264" />, Sections 6.1 and 7, and start
transmitting RTP media using that format once the
underlying transport layers have been established. If an
SSRC has not already been chosen for this outgoing RTP
stream, choose a random one. If media is already being
transmitted, the same SSRC SHOULD be used unless the
clockrate of the new codec is different, in which case a
new SSRC MUST be chosen, as specified in
<xref target="RFC7160" />, Section 3.1.</t>
<t>The payload type mapping from the remote description is Original:
used to determine payload types for the outgoing RTP o If the current local description indicates an ICE restart, and
streams, including the payload type for the send media either the ICE ufrag or password has not changed from the previous
format chosen above. Any RTP header extensions that were remote description, as prescribed by [RFC8445], Section 9,
negotiated should be included in the outgoing RTP streams, generate an error. -->
using the extension mapping from the remote description; if
the RID header extension has been negotiated, and RID
values are specified, include the RID header extension in
the outgoing RTP streams, as indicated in
<xref target="I-D.ietf-mmusic-rid"></xref>, Section 4.</t>
<t>If the m= section is of type audio, and silence </li>
suppression was configured for the send media format as a <li>Configure the ICE components associated with this media
result of processing the remote description, and is also section to use the supplied ICE remote ufrag and password for
enabled for that format in the local description, use their connectivity checks.</li>
silence suppression for outgoing media, in accordance with <li>Pair any supplied ICE candidates with any gathered local
the guidance in candidates, as described in
<xref target="sec.voiceactivitydetection1" />. If these <xref target="RFC8445" sectionFormat="comma" section="6.1.2"/>, and st
conditions are not met, silence suppression MUST NOT be art
used for outgoing media.</t> connectivity checks with the appropriate credentials.</li>
<t>If simulcast has been negotiated, send the number of <li>If an "a=end-of-candidates" attribute is present, process
Source RTP Streams as specified in the end-of-candidates indication as described in
<xref target="I-D.ietf-mmusic-sdp-simulcast"></xref>, <xref target="RFC8838" sectionFormat="comma" section="14"/>.</li>
Section 6.2.2.</t> <li>
<t>If the "m=" section proto value indicates use of RTP:
</t>
<ul spacing="normal">
<li>If the "m=" section is being recycled (see
<xref target="sec.subsequent-offers" format="default"/>), dissociat
e
the currently associated RtpTransceiver by setting its mid
property to "null", and discard the mapping between the
transceiver and its "m=" section index.</li>
<li>
<t>If the "m=" section is not associated with any
RtpTransceiver (possibly because it was dissociated in the
previous step), either find an RtpTransceiver or create one
according to the following steps:
</t>
<ul spacing="normal">
<li>If the "m=" section is sendrecv or recvonly, and there
are RtpTransceivers of the same type that were added to
the PeerConnection by addTrack and are not associated
with any "m=" section and are not stopped, find the first
(according to the canonical order described in
<xref target="sec.initial-offers" format="default"/>) such
RtpTransceiver.</li>
<li>If no RtpTransceiver was found in the previous step,
create one with a recvonly direction.</li>
<li>Associate the found or created RtpTransceiver with the
"m=" section by setting the value of the RtpTransceiver's
mid property to the MID of the "m=" section, and establish
a mapping between the transceiver and the index of the "m="
section. If the "m=" section does not include a MID (i.e.,
the remote endpoint does not support the MID extension),
generate a value for the RtpTransceiver mid property,
following the guidance for "a=mid" mentioned in
<xref target="sec.initial-offers" format="default"/>.</li>
</ul>
</li>
<li>For each specified media format that is also supported
by the local implementation, establish a mapping between
the specified payload type and the media format, as
described in
<xref target="RFC3264" sectionFormat="comma" section="6.1"/>. Speci
fically, this
means that the implementation records the payload type to
be used in outgoing RTP packets when sending each specified
media format, as well as the relative preference for each
format that is indicated in their ordering. If any
indicated media format is not supported by the local
implementation, it <bcp14>MUST</bcp14> be ignored.</li>
<li>For each specified "rtx" media format, establish a
mapping between the RTX payload type and its associated
primary payload type, as described in
<xref target="RFC4588" sectionFormat="comma" section="4"/>. If any
referenced
primary payload types are not present, this <bcp14>MUST</bcp14> res
ult in
an error. Note that RTX payload types may refer to primary
payload types that are not supported by the local media
implementation, in which case the RTX payload type <bcp14>MUST</bcp
14>
also be ignored.</li>
<li>For each specified fmtp parameter that is supported by
the local implementation, enable them on the associated
media formats.</li>
<li>For each specified Synchronization Source (SSRC) that is sign
aled in the "m="
section, prepare to demux RTP streams intended for this "m="
section using that SSRC, as described in
<xref target="RFC8843" sectionFormat="comma" section="9.2"/>.</li>
<li>For each specified RTP header extension that is also
supported by the local implementation, establish a mapping
between the extension ID and URI, as described in
<xref target="RFC5285" sectionFormat="comma" section="5"/>. Specifi
cally, this
means that the implementation records the extension ID to
be used in outgoing RTP packets when sending each specified
header extension. If any indicated RTP header extension is
not supported by the local implementation, it <bcp14>MUST</bcp14> b
e
ignored.</li>
<li>For each specified RTCP feedback mechanism that is
supported by the local implementation, enable them on the
associated media formats.</li>
<li>
<t>For any specified "TIAS" ("Transport
Independent Application Specific Maximum") bandwidth value, set this value
as a constraint on the maximum RTP bitrate to be used when
sending media, as specified in
<xref target="RFC3890" format="default"/>. If a "TIAS" value is not
present but an "AS" value is specified, generate a "TIAS"
value using this formula:
</t>
<ul empty="true">
<li>TIAS = AS * 1000 * 0.95 - (50 * 40 * 8)</li>
</ul>
<t>
The "50" is based on 50 packets per second, the "40" is
based on an estimate of total header size, the "1000" changes
the unit from kbps to bps (as required by TIAS), and the
"0.95" is to allocate 5% to RTCP.
<t>If the send media format chosen above has a <!-- [rfced] Section 5.10: For ease of the reader, should the "8"
corresponding "rtx" media format, or a FEC mechanism has also be explained (e.g., possibly "bytes to bits")? We ask because
been negotiated, establish a Redundancy RTP Stream with a explanations are provided for the other four numbers.
random SSRC for each Source RTP Stream, and start or
continue transmitting RTX/FEC packets as needed.</t>
<t>If the send media format chosen above has a Original:
corresponding "red" media format of the same clockrate, TIAS = AS * 1000 * 0.95 - (50 * 40 * 8)
allow redundant encoding using the specified format for
resiliency purposes, as discussed in
<xref target="I-D.ietf-rtcweb-fec" />, Section 3.2. Note
that unlike RTX or FEC media formats, the "red" format is
transmitted on the Source RTP Stream, not the Redundancy
RTP Stream.</t>
<t>Enable the RTCP feedback mechanisms referenced in the The 50 is based on 50 packets per second, the 40 is based on an
media section for all Source RTP Streams using the estimate of total header size, the 1000 changes the unit from
specified media formats. Specifically, begin or continue kbps to bps (as required by TIAS), and the 0.95 is to allocate
sending the requested feedback types and reacting to 5% to RTCP.-->
received feedback, as specified in
<xref target="RFC4585" />, Section 4.2. When sending RTCP
feedback, follow the rules and recommendations from
<xref target="RFC8108"></xref> Section 5.4.1, to select
which SSRC to use.</t>
<t>If the directional attribute in the answer indicates "TIAS" is used in
that the JSEP implementation should not be sending media preference to "AS" because it provides more accurate
("recvonly" for local answers, "sendonly" for remote control of bandwidth.</t>
answers, or "inactive" for either type of answer) stop </li>
transmitting all RTP media, but continue sending RTCP, as <li>For any "RR" or "RS" bandwidth values, handle as
described in specified in
<xref target="RFC3264" />, Section 5.1.</t> <xref target="RFC3556" sectionFormat="comma" section="2"/>.</li>
</list></t> <li>Any specified "CT" bandwidth value <bcp14>MUST</bcp14> be ign
ored, as
the meaning of this construct at the media level is not
well defined.</li>
<li>
<t>If the "m=" section is of type "audio":
</t>
<ul spacing="normal">
<li>For each specified "CN" media format, configure
silence suppression for all supported media formats with
the same clock rate, as described in
<xref target="RFC3389" sectionFormat="comma" section="5"/>, excep
t for formats
that have their own internal silence suppression
mechanisms. Silence suppression for such formats (e.g.,
Opus) is controlled via fmtp parameters, as discussed in
<xref target="sec.voiceactivitydetection1" format="default"/>.</l
i>
<li>For each specified "telephone-event" media format,
enable dual-tone multifrequency (DTMF) transmission for all suppo
rted media formats
with the same clock rate, as described in
<xref target="RFC4733" sectionFormat="comma" section="2.5.1.2"/>.
If there are
any supported media formats that do not have a
corresponding telephone-event format, disable DTMF
transmission for those formats.</li>
<li>For any specified "ptime" value, configure the
available media formats to use the specified packet size
when sending. If the specified size is not supported for
a media format, use the next closest value instead.</li>
</ul>
</li>
</ul>
</li>
</ul>
<t>Finally, if this description is of type "pranswer" or
"answer", follow the processing defined in
<xref target="sec.applying-an-answer" format="default"/> below.</t>
</section>
<section anchor="sec.applying-an-answer" numbered="true" toc="default">
<name>Applying an Answer</name>
<t>In addition to the steps mentioned above for processing a
local or remote description, the following steps are performed
when processing a description of type "pranswer" or
"answer".</t>
<t>For each "m=" section, the following steps <bcp14>MUST</bcp14> be pe
rformed:
</t>
<ul spacing="normal">
<li>If the "m=" section has been rejected (i.e., the port value is se
t to
zero in the answer), stop any reception or transmission of
media for this section, and, unless a non-rejected "m=" section
is bundled with this "m=" section, discard any associated ICE
components, as described in
<xref target="RFC8839" sectionFormat="comma" section="4.4.3.1"/>.</li
>
<li>If the remote DTLS fingerprint has been changed or the
tls-id has changed, tear down the DTLS connection. This
includes the case when the PeerConnection state is
"have-remote-pranswer". If a DTLS connection needs to be torn
down but the answer does not indicate an ICE restart or, in
the case of "have-remote-pranswer", new ICE credentials, an
error <bcp14>MUST</bcp14> be generated. If an ICE restart is perform
ed
without a change in tls-id or fingerprint, then the same DTLS
connection is continued over the new ICE channel. Note that
although JSEP requires that answerers change the tls-id value
if and only if the offerer does, non-JSEP answerers are
permitted to change the tls-id as long as the offer contained
an ICE restart. Thus, JSEP implementations that process DTLS
data prior to receiving an answer <bcp14>MUST</bcp14> be prepared to
receive
either a ClientHello or data from the previous DTLS
connection.</li>
<li>If no valid DTLS connection exists, prepare to start a
DTLS connection, using the specified roles and fingerprints,
on any underlying ICE components, once they are active.</li>
<li>
<t>If the "m=" section proto value indicates use of RTP:
</t>
<ul spacing="normal">
<li>If the "m=" section references RTCP feedback mechanisms
that were not present in the corresponding "m=" section in
the offer, this indicates a negotiation problem and <bcp14>MUST</b
cp14>
result in an error. However, new media formats and new RTP
header extension values are permitted in the answer, as
described in
<xref target="RFC3264" sectionFormat="comma" section="7"/> and
<xref target="RFC5285" sectionFormat="comma" section="6"/>.</li>
<li>If the "m=" section has RTCP mux enabled, discard the RTCP
ICE component, if one exists, and begin or continue muxing
RTCP over the RTP ICE component, as specified in
<xref target="RFC5761" sectionFormat="comma" section="5.1.3"/>. Ot
herwise,
prepare to transmit RTCP over the RTCP ICE component; if no
RTCP ICE component exists because RTCP mux was previously
enabled, this <bcp14>MUST</bcp14> result in an error.</li>
<li>If the "m=" section has Reduced-Size RTCP enabled,
configure the RTCP transmission for this "m=" section to use
Reduced-Size RTCP, as specified in
<xref target="RFC5506" format="default"/>.</li>
<li>If the directional attribute in the answer indicates
that the JSEP implementation should be sending media
("sendonly" for local answers, "recvonly" for remote
answers, or "sendrecv" for either type of answer), choose
the media format to send as the most preferred media format
from the remote description that is also locally supported,
as discussed in Sections <xref target="RFC3264" section="6.1"
sectionFormat="bare"/> and <xref target="RFC3264" section="7" sectionFormat="
bare"/> of <xref target="RFC3264"/>, and start
transmitting RTP media using that format once the
underlying transport layers have been established. If an
SSRC has not already been chosen for this outgoing RTP
stream, choose a random one. If media is already being
transmitted, the same SSRC <bcp14>SHOULD</bcp14> be used unless th
e
clock rate of the new codec is different, in which case a
new SSRC <bcp14>MUST</bcp14> be chosen, as specified in
<xref target="RFC7160" sectionFormat="comma" section="3.1"/>.</li>
<li>The payload type mapping from the remote description is
used to determine payload types for the outgoing RTP
streams, including the payload type for the send media
format chosen above. Any RTP header extensions that were
negotiated should be included in the outgoing RTP streams,
using the extension mapping from the remote description; if
the RID header extension has been negotiated and RID
values are specified, include the RID header extension in
the outgoing RTP streams, as indicated in
<xref target="RFC8851" sectionFormat="comma" section="4"/>.</li>
<li>If (1) the "m=" section is of type "audio" and (2)&nbsp;sile
nce
suppression was configured for the send media format as a
result of processing the remote description and is also
enabled for that format in the local description, use
silence suppression for outgoing media, in accordance with
the guidance in
<xref target="sec.voiceactivitydetection1" format="default"/>.
<t>If the m= section proto value indicates use of SCTP: <!-- [rfced] Section 5.11: As it appears that "is also enabled"
<list style="symbols"> refers to silence suppression, we updated this sentence as follows.
Please let us know if this is incorrect.
<t>If an SCTP association exists, and the remote SCTP port Original:
has changed, discard the existing SCTP association. This * If the m= section is of type audio, and silence suppression was
includes the case when the PeerConnection state is configured for the send media format as a result of processing
"have-remote-pranswer".</t> the remote description, and is also enabled for that format in
the local description, use silence suppression for outgoing
media, in accordance with the guidance in Section 5.2.3.2.
<t>If no valid SCTP association exists, prepare to initiate Currently:
a SCTP association over the associated ICE component and - If (1) the "m=" section is of type "audio" and (2) silence
DTLS connection, using the local SCTP port value from the suppression was configured for the send media format as a
local description, and the remote SCTP port value from the result of processing the remote description and is also enabled
remote description, as described in for that format in the local description, use silence
<xref target="I-D.ietf-mmusic-sctp-sdp" />, Section suppression for outgoing media, in accordance with the guidance
10.2.</t> in Section 5.2.3.2. -->
</list></t>
</list></t>
<t>If the answer contains valid bundle groups, discard any ICE If these
components for the m= sections that will be bundled onto the conditions are not met, silence suppression <bcp14>MUST NOT</bcp14
primary ICE components in each bundle, and begin muxing these > be
m= sections accordingly, as described in used for outgoing media.</li>
<xref target="I-D.ietf-mmusic-sdp-bundle-negotiation" />, <li>If simulcast has been negotiated, send the number of
Section 8.2.</t> Source RTP Streams as specified in
<xref target="RFC8853" sectionFormat="comma" section="6.2.2"/>.
<t>If the description is of type "answer", and there are still <!-- [rfced] Section 5.11: Please (1) confirm that Section 6.2.2 of
remaining candidates in the ICE candidate pool, discard RFC 8853 [I-D.ietf-mmusic-sdp-simulcast] (with version -14 being the latest
them.</t> for this draft) is the correct section to cite here and (2) clarify
</section> the meaning of "send the number of Source RTP Streams as specified in"
</section> (perhaps "... the specified number of ..." or "... the appropriate
<section title="Processing RTP/RTCP" anchor="sec.rtp.demux"> number of ..."?).
<t>When bundling, associating incoming RTP/RTCP with the proper Original:
m= section is defined in * If simulcast has been negotiated, send the number of Source RTP
<xref target="I-D.ietf-mmusic-sdp-bundle-negotiation" />, Section Streams as specified in [I-D.ietf-mmusic-sdp-simulcast],
10.2. When not bundling, the proper m= section is clear from the Section 6.2.2. -->
ICE component over which the RTP/RTCP is received.</t>
<t>Once the proper m= section(s) are known, RTP/RTCP is delivered </li>
to the RtpTransceiver(s) associated with the m= section(s) and <li>If the send media format chosen above has a
further processing of the RTP/RTCP is done at the RtpTransceiver corresponding "rtx" media format or a FEC mechanism has
level. This includes using RID been negotiated, establish a redundancy RTP stream with a
<xref target="I-D.ietf-mmusic-rid" /> to distinguish between random SSRC for each Source RTP Stream, and start or
multiple Encoded Streams, as well as determine which Source RTP continue transmitting RTX/FEC packets as needed.</li>
stream should be repaired by a given Redundancy RTP stream.</t> <li>If the send media format chosen above has a
</section> corresponding "red" media format of the same clock rate,
<section title="Examples" anchor="sec.examples"> allow redundant encoding using the specified format for
resiliency purposes, as discussed in
<xref target="RFC8854" sectionFormat="comma" section="3.2"/>. Note
that unlike RTX or FEC media formats, the "red" format is
transmitted on the Source RTP Stream, not the redundancy
RTP stream.</li>
<li>Enable the RTCP feedback mechanisms referenced in the
media section for all Source RTP Streams using the
specified media formats. Specifically, begin or continue
sending the requested feedback types and reacting to
received feedback, as specified in
<xref target="RFC4585" sectionFormat="comma" section="4.2"/>. When
sending RTCP
feedback, follow the rules and recommendations from
<xref target="RFC8108" sectionFormat="comma" section="5.4.1"/> to select
which SSRC to use.</li>
<li>If the directional attribute in the answer indicates
that the JSEP implementation should not be sending media
("recvonly" for local answers, "sendonly" for remote
answers, or "inactive" for either type of answer), stop
transmitting all RTP media, but continue sending RTCP, as
described in
<xref target="RFC3264" sectionFormat="comma" section="5.1"/>.</li>
</ul>
</li>
<li>
<t>If the "m=" section proto value indicates use of SCTP:
</t>
<ul spacing="normal">
<li>If an SCTP association exists and the remote SCTP port
has changed, discard the existing SCTP association. This
includes the case when the PeerConnection state is
"have-remote-pranswer".</li>
<li>If no valid SCTP association exists, prepare to initiate
an SCTP association over the associated ICE component and
DTLS connection, using the local SCTP port value from the
local description and the remote SCTP port value from the
remote description, as described in
<xref target="RFC8841" sectionFormat="comma" section="10.2"/>.</li
>
</ul>
</li>
</ul>
<t>If the answer contains valid bundle groups, discard any ICE
components for the "m=" sections that will be bundled onto the
primary ICE components in each bundle, and begin muxing these
"m=" sections accordingly, as described in
<xref target="RFC8843" sectionFormat="comma" section="7.2"/>.</t>
<t>If the description is of type "answer" and there are still
remaining candidates in the ICE candidate pool, discard
them.</t>
</section>
</section>
<section anchor="sec.rtp.demux" numbered="true" toc="default">
<name>Processing RTP/RTCP</name>
<t>When bundling, associating incoming RTP/RTCP with the proper
"m=" section is defined in
<xref target="RFC8843" sectionFormat="comma" section="9.2"/>. When not b
undling, the proper "m=" section is clear from the
ICE component over which the RTP/RTCP is received.</t>
<t>Once the proper "m=" section or sections are known, RTP/RTCP is deliv
ered
to the RtpTransceiver(s) associated with the "m=" section(s) and
further processing of the RTP/RTCP is done at the RtpTransceiver
level. This includes using RID
<xref target="RFC8851" format="default"/> to distinguish between
multiple encoded streams, as well as to determine which Source RTP
stream should be repaired by a given redundancy RTP stream.</t>
</section>
<section anchor="sec.examples" numbered="true" toc="default">
<name>Examples</name>
<t>Note that this example section shows several SDP fragments. To
accommodate RFC line-length restrictions, some of the SDP lines have bee
n split
into multiple lines, where leading whitespace indicates that a
line is a continuation of the previous line. In addition, some
blank lines have been added to improve readability but are not
valid in SDP.</t>
<t>More examples of SDP for WebRTC call flows, including examples
with IPv6 addresses, can be found in
<xref target="I-D.ietf-rtcweb-sdp" format="default"/>.</t>
<section anchor="sec.simple-examples" numbered="true" toc="default">
<name>Simple Example</name>
<t>This section shows a very simple example that sets up a
minimal audio/video call between two JSEP endpoints without
using Trickle ICE. The example in the following section
provides a more detailed example of what could happen in a JSEP
session.</t>
<t>The code flow below shows Alice's endpoint initiating the
session to Bob's endpoint. The messages from the JavaScript
application in Alice's browser to the JavaScript in Bob's
browser, abbreviated as "AliceJS" and "BobJS", respectively, are
assumed to flow over some signaling protocol via a web server.
The JavaScript on both Alice's side and Bob's side waits for
all candidates before sending the offer or answer, so the
offers and answers are complete; Trickle ICE is not used. The
user agents (JSEP implementations) in Alice's and Bob's browsers,
abbreviated as "AliceUA" and "BobUA", respectively, are using the
default bundle policy of "balanced", and the default RTCP mux
policy of "require".
<t>Note that this example section shows several SDP fragments. To <!-- [rfced] Section 7.1: We had trouble following the meaning of
format in 72 columns, some of the lines in SDP have been split this sentence. If the suggested text is not correct, please clarify.
into multiple lines, where leading whitespace indicates that a
line is a continuation of the previous line. In addition, some
blank lines have been added to improve readability but are not
valid in SDP.</t>
<t>More examples of SDP for WebRTC call flows, including examples Original:
with IPv6 addresses, can be found in The user agents (JSEP implementations) in Alice and Bob's
<xref target="I-D.ietf-rtcweb-sdp"></xref>.</t> browsers, abbreviated as AliceUA and BobUA respectively, are using
<section title="Simple Example" anchor="sec.simple-examples"> the default bundle policy of "balanced", and the default RTCP mux
policy of "require".
<t>This section shows a very simple example that sets up a Suggested:
minimal audio / video call between two JSEP endpoints without The user agents (JSEP implementations) in Alice's and Bob's
using trickle ICE. The example in the following section browsers, abbreviated as "AliceUA" and "BobUA", respectively, are
provides a more detailed example of what could happen in a JSEP using the default bundle policy of "balanced" (AliceUA) and the
session.</t> default RTCP mux policy of "require" (BobUA). -->
<t>The code flow below shows Alice's endpoint initiating the </t>
session to Bob's endpoint. The messages from the JavaScript <!-- [rfced] Throughout: <artwork> has been converted to <sourcecode> as
application in Alice's browser to the JavaScript in Bob's applicable and we set the "type" attribute. Please review and let us know
browser, abbreviated as AliceJS and BobJS respectively, are corrections are needed.
assumed to flow over some signaling protocol via a web server. -->
The JavaScript on both Alice's side and Bob's side waits for
all candidates before sending the offer or answer, so the
offers and answers are complete; trickle ICE is not used. The
user agents (JSEP implementations) in Alice and Bob's browsers,
abbreviated as AliceUA and BobUA respectively, are using the
default bundle policy of "balanced", and the default RTCP mux
policy of "require".</t>
<t> <sourcecode name="" type="pseudocode"><![CDATA[
<figure>
<artwork>
<![CDATA[
// set up local media state // set up local media state
AliceJS->AliceUA: create new PeerConnection AliceJS->AliceUA: create new PeerConnection
AliceJS->AliceUA: addTrack with two tracks: audio and video AliceJS->AliceUA: addTrack with two tracks: audio and video
AliceJS->AliceUA: createOffer to get offer AliceJS->AliceUA: createOffer to get offer
AliceJS->AliceUA: setLocalDescription with offer AliceJS->AliceUA: setLocalDescription with offer
AliceUA->AliceJS: multiple onicecandidate events with candidates AliceUA->AliceJS: multiple onicecandidate events with candidates
// wait for ICE gathering to complete // wait for ICE gathering to complete
AliceUA->AliceJS: onicecandidate event with null candidate AliceUA->AliceJS: onicecandidate event with null candidate
AliceJS->AliceUA: get |offer-A1| from pendingLocalDescription AliceJS->AliceUA: get |offer-A1| from pendingLocalDescription
skipping to change at line 4192 skipping to change at line 4718
// Bob accepts call // Bob accepts call
BobJS->BobUA: addTrack with local tracks BobJS->BobUA: addTrack with local tracks
BobJS->BobUA: createAnswer BobJS->BobUA: createAnswer
BobJS->BobUA: setLocalDescription with answer BobJS->BobUA: setLocalDescription with answer
BobUA->BobJS: multiple onicecandidate events with candidates BobUA->BobJS: multiple onicecandidate events with candidates
// wait for ICE gathering to complete // wait for ICE gathering to complete
BobUA->BobJS: onicecandidate event with null candidate BobUA->BobJS: onicecandidate event with null candidate
BobJS->BobUA: get |answer-A1| from currentLocalDescription BobJS->BobUA: get |answer-A1| from currentLocalDescription
// |answer-A1| is sent over signaling protocol to Alice // |answer-A1| is sent over signaling protocol
// to Alice
BobJS->WebServer: signaling with |answer-A1| BobJS->WebServer: signaling with |answer-A1|
WebServer->AliceJS: signaling with |answer-A1| WebServer->AliceJS: signaling with |answer-A1|
// |answer-A1| arrives at Alice // |answer-A1| arrives at Alice
AliceJS->AliceUA: setRemoteDescription with |answer-A1| AliceJS->AliceUA: setRemoteDescription with |answer-A1|
AliceUA->AliceJS: ontrack events for audio and video tracks AliceUA->AliceJS: ontrack events for audio and video tracks
// media flows // media flows
BobUA->AliceUA: media sent from Bob to Alice BobUA->AliceUA: media sent from Bob to Alice
AliceUA->BobUA: media sent from Alice to Bob AliceUA->BobUA: media sent from Alice to Bob ]]></sourcecode>
]]>
</artwork>
</figure>
</t>
<t>The SDP for |offer-A1| looks like:</t>
<t> <t>The SDP for |offer-A1| looks like:</t>
<figure> <sourcecode name="offer-A1" type="sdp"><![CDATA[
<artwork alt="offer-A1">
<![CDATA[
v=0 v=0
o=- 4962303333179871722 1 IN IP4 0.0.0.0 o=- 4962303333179871722 1 IN IP4 0.0.0.0
s=- s=-
t=0 0 t=0 0
a=ice-options:trickle ice2 a=ice-options:trickle ice2
a=group:BUNDLE a1 v1 a=group:BUNDLE a1 v1
a=group:LS a1 v1 a=group:LS a1 v1
m=audio 10100 UDP/TLS/RTP/SAVPF 96 0 8 97 98 m=audio 10100 UDP/TLS/RTP/SAVPF 96 0 8 97 98
c=IN IP4 203.0.113.100 c=IN IP4 203.0.113.100
skipping to change at line 4280 skipping to change at line 4799
a=fingerprint:sha-256 a=fingerprint:sha-256
19:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04: 19:E2:1C:3B:4B:9F:81:E6:B8:5C:F4:A5:A8:D8:73:04:
BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2 BB:05:2F:70:9F:04:A9:0E:05:E9:26:33:E8:70:88:A2
a=setup:actpass a=setup:actpass
a=tls-id:91bbf309c0990a6bec11e38ba2933cee a=tls-id:91bbf309c0990a6bec11e38ba2933cee
a=rtcp:10103 IN IP4 203.0.113.100 a=rtcp:10103 IN IP4 203.0.113.100
a=rtcp-mux a=rtcp-mux
a=rtcp-rsize a=rtcp-rsize
a=candidate:1 1 udp 2113929471 203.0.113.100 10102 typ host a=candidate:1 1 udp 2113929471 203.0.113.100 10102 typ host
a=candidate:1 2 udp 2113929470 203.0.113.100 10103 typ host a=candidate:1 2 udp 2113929470 203.0.113.100 10103 typ host
a=end-of-candidates a=end-of-candidates ]]></sourcecode>
]]>
</artwork>
</figure>
</t>
<t>The SDP for |answer-A1| looks like:</t> <!-- [rfced] Sections 7.1, 7.2, and 7.3: Please confirm that the
"=" signs in these ten "=rtpmap:103" entries do not need to be
preceded by an "a" line identifier. We ask because these are the
only "=rtpmap:" SDP lines that begin with "=" instead of "a=".
<t> Example from original:
<figure> =rtpmap:103 rtx/90000 -->
<artwork alt="answer-A1">
<![CDATA[ <t>The SDP for |answer-A1| looks like:</t>
<sourcecode name="answer-A1" type="sdp"><![CDATA[
v=0 v=0
o=- 6729291447651054566 1 IN IP4 0.0.0.0 o=- 6729291447651054566 1 IN IP4 0.0.0.0
s=- s=-
t=0 0 t=0 0
a=ice-options:trickle ice2 a=ice-options:trickle ice2
a=group:BUNDLE a1 v1 a=group:BUNDLE a1 v1
a=group:LS a1 v1 a=group:LS a1 v1
m=audio 10200 UDP/TLS/RTP/SAVPF 96 0 8 97 98 m=audio 10200 UDP/TLS/RTP/SAVPF 96 0 8 97 98
c=IN IP4 203.0.113.200 c=IN IP4 203.0.113.200
skipping to change at line 4343 skipping to change at line 4862
a=fmtp:101 packetization-mode=1;profile-level-id=42e01f a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
a=rtpmap:102 rtx/90000 a=rtpmap:102 rtx/90000
a=fmtp:102 apt=100 a=fmtp:102 apt=100
=rtpmap:103 rtx/90000 =rtpmap:103 rtx/90000
a=fmtp:103 apt=101 a=fmtp:103 apt=101
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=rtcp-fb:100 ccm fir a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli a=rtcp-fb:100 nack pli
a=msid:61317484-2ed4-49d7-9eb7-1414322a7aae a=msid:61317484-2ed4-49d7-9eb7-1414322a7aae ]]></sourcecode>
]]> </section>
</artwork> <section anchor="sec.detailed-example" numbered="true" toc="default">
</figure> <name>Detailed Example</name>
</t> <t>This section shows a more involved example of a session
</section> between two JSEP endpoints. Trickle ICE is used in full trickle
<section title="Detailed Example" anchor="sec.detailed-example"> mode, with a bundle policy of "max-bundle", an RTCP mux policy
of "require", and a single TURN server. Initially, both Alice
<t>This section shows a more involved example of a session and Bob establish an audio channel and a data channel. Later,
between two JSEP endpoints. Trickle ICE is used in full trickle Bob adds two video flows -- one for his video feed and one for
mode, with a bundle policy of "max-bundle", an RTCP mux policy screen sharing, both supporting FEC -- with the video feed
of "require", and a single TURN server. Initially, both Alice configured for simulcast. Alice accepts these video flows but
and Bob establish an audio channel and a data channel. Later, does not add video flows of her own, so they are handled as
Bob adds two video flows, one for his video feed, and one for recvonly. Alice also specifies a maximum video decoder
screensharing, both supporting FEC, and with the video feed resolution.</t>
configured for simulcast. Alice accepts these video flows, but
does not add video flows of her own, so they are handled as
recvonly. Alice also specifies a maximum video decoder
resolution.</t>
<t> <sourcecode name="" type="pseudocode"><![CDATA[
<figure>
<artwork>
<![CDATA[
// set up local media state // set up local media state
AliceJS->AliceUA: create new PeerConnection AliceJS->AliceUA: create new PeerConnection
AliceJS->AliceUA: addTrack with an audio track AliceJS->AliceUA: addTrack with an audio track
AliceJS->AliceUA: createDataChannel to get data channel AliceJS->AliceUA: createDataChannel to get data channel
AliceJS->AliceUA: createOffer to get |offer-B1| AliceJS->AliceUA: createOffer to get |offer-B1|
AliceJS->AliceUA: setLocalDescription with |offer-B1| AliceJS->AliceUA: setLocalDescription with |offer-B1|
// |offer-B1| is sent over signaling protocol to Bob // |offer-B1| is sent over signaling protocol to Bob
AliceJS->WebServer: signaling with |offer-B1| AliceJS->WebServer: signaling with |offer-B1|
WebServer->BobJS: signaling with |offer-B1| WebServer->BobJS: signaling with |offer-B1|
// |offer-B1| arrives at Bob // |offer-B1| arrives at Bob
BobJS->BobUA: create a PeerConnection BobJS->BobUA: create a PeerConnection
BobJS->BobUA: setRemoteDescription with |offer-B1| BobJS->BobUA: setRemoteDescription with |offer-B1|
BobUA->BobJS: ontrack with audio track from Alice BobUA->BobJS: ontrack event with audio track from Alice
// candidates are sent to Bob // candidates are sent to Bob
AliceUA->AliceJS: onicecandidate (host) |offer-B1-candidate-1| AliceUA->AliceJS: onicecandidate (host) |offer-B1-candidate-1|
AliceJS->WebServer: signaling with |offer-B1-candidate-1| AliceJS->WebServer: signaling with |offer-B1-candidate-1|
AliceUA->AliceJS: onicecandidate (srflx) |offer-B1-candidate-2| AliceUA->AliceJS: onicecandidate (srflx) |offer-B1-candidate-2|
AliceJS->WebServer: signaling with |offer-B1-candidate-2| AliceJS->WebServer: signaling with |offer-B1-candidate-2|
AliceUA->AliceJS: onicecandidate (relay) |offer-B1-candidate-3| AliceUA->AliceJS: onicecandidate (relay) |offer-B1-candidate-3|
AliceJS->WebServer: signaling with |offer-B1-candidate-3| AliceJS->WebServer: signaling with |offer-B1-candidate-3|
WebServer->BobJS: signaling with |offer-B1-candidate-1| WebServer->BobJS: signaling with |offer-B1-candidate-1|
skipping to change at line 4435 skipping to change at line 4947
// data channel opens // data channel opens
BobUA->BobJS: ondatachannel event BobUA->BobJS: ondatachannel event
AliceUA->AliceJS: ondatachannel event AliceUA->AliceJS: ondatachannel event
BobUA->BobJS: onopen BobUA->BobJS: onopen
AliceUA->AliceJS: onopen AliceUA->AliceJS: onopen
// media is flowing between endpoints // media is flowing between endpoints
BobUA->AliceUA: audio+data sent from Bob to Alice BobUA->AliceUA: audio+data sent from Bob to Alice
AliceUA->BobUA: audio+data sent from Alice to Bob AliceUA->BobUA: audio+data sent from Alice to Bob
// some time later Bob adds two video streams // some time later, Bob adds two video streams
// note, no candidates exchanged, because of bundle // note: no candidates exchanged, because of bundle
BobJS->BobUA: addTrack with first video stream BobJS->BobUA: addTrack with first video stream
BobJS->BobUA: addTrack with second video stream BobJS->BobUA: addTrack with second video stream
BobJS->BobUA: createOffer to get |offer-B2| BobJS->BobUA: createOffer to get |offer-B2|
BobJS->BobUA: setLocalDescription with |offer-B2| BobJS->BobUA: setLocalDescription with |offer-B2|
// |offer-B2| is sent to Alice // |offer-B2| is sent to Alice
BobJS->WebServer: signaling with |offer-B2| BobJS->WebServer: signaling with |offer-B2|
WebServer->AliceJS: signaling with |offer-B2| WebServer->AliceJS: signaling with |offer-B2|
AliceJS->AliceUA: setRemoteDescription with |offer-B2| AliceJS->AliceUA: setRemoteDescription with |offer-B2|
AliceUA->AliceJS: ontrack event with first video track AliceUA->AliceJS: ontrack event with first video track
AliceUA->AliceJS: ontrack event with second video track AliceUA->AliceJS: ontrack event with second video track
AliceJS->AliceUA: createAnswer to get |answer-B2| AliceJS->AliceUA: createAnswer to get |answer-B2|
AliceJS->AliceUA: setLocalDescription with |answer-B2| AliceJS->AliceUA: setLocalDescription with |answer-B2|
// |answer-B2| is sent over signaling protocol to Bob // |answer-B2| is sent over signaling protocol
// to Bob
AliceJS->WebServer: signaling with |answer-B2| AliceJS->WebServer: signaling with |answer-B2|
WebServer->BobJS: signaling with |answer-B2| WebServer->BobJS: signaling with |answer-B2|
BobJS->BobUA: setRemoteDescription with |answer-B2| BobJS->BobUA: setRemoteDescription with |answer-B2|
// media is flowing between endpoints // media is flowing between endpoints
BobUA->AliceUA: audio+video+data sent from Bob to Alice BobUA->AliceUA: audio+video+data sent from Bob to Alice
AliceUA->BobUA: audio+video+data sent from Alice to Bob AliceUA->BobUA: audio+video+data sent from Alice to Bob ]]></sourcecode>
]]>
</artwork>
</figure>
</t>
<t>The SDP for |offer-B1| looks like:</t> <!-- [rfced] Section 7.2: We changed "ontrack with" to "ontrack event
with" per the other three "ontrack event with" items. Please let us
know if this is incorrect.
<t> Original:
<figure> BobUA->BobJS: ontrack with audio track from Alice
<artwork alt="offer-B1">
<![CDATA[ Currently:
BobUA->BobJS: ontrack event with audio track from Alice -->
<t>The SDP for |offer-B1| looks like:</t>
<sourcecode name="offer-B1" type="sdp"><![CDATA[
v=0 v=0
o=- 4962303333179871723 1 IN IP4 0.0.0.0 o=- 4962303333179871723 1 IN IP4 0.0.0.0
s=- s=-
t=0 0 t=0 0
a=ice-options:trickle ice2 a=ice-options:trickle ice2
a=group:BUNDLE a1 d1 a=group:BUNDLE a1 d1
m=audio 9 UDP/TLS/RTP/SAVPF 96 0 8 97 98 m=audio 9 UDP/TLS/RTP/SAVPF 96 0 8 97 98
c=IN IP4 0.0.0.0 c=IN IP4 0.0.0.0
a=mid:a1 a=mid:a1
skipping to change at line 4508 skipping to change at line 5023
a=tls-id:17f0f4ba8a5f1213faca591b58ba52a7 a=tls-id:17f0f4ba8a5f1213faca591b58ba52a7
a=rtcp-mux a=rtcp-mux
a=rtcp-mux-only a=rtcp-mux-only
a=rtcp-rsize a=rtcp-rsize
m=application 0 UDP/DTLS/SCTP webrtc-datachannel m=application 0 UDP/DTLS/SCTP webrtc-datachannel
c=IN IP4 0.0.0.0 c=IN IP4 0.0.0.0
a=mid:d1 a=mid:d1
a=sctp-port:5000 a=sctp-port:5000
a=max-message-size:65536 a=max-message-size:65536
a=bundle-only a=bundle-only ]]></sourcecode>
]]>
</artwork>
</figure>
</t>
<t>|offer-B1-candidate-1| looks like:</t> <t>|offer-B1-candidate-1| looks like:</t>
<t> <sourcecode name="offer-B1-candidate-1" type="sdp"><![CDATA[
<figure>
<artwork alt="offer-B1-candidate-1">
<![CDATA[
ufrag ATEn ufrag ATEn
index 0 index 0
mid a1 mid a1
attr candidate:1 1 udp 2113929471 203.0.113.100 10100 typ host attr candidate:1 1 udp 2113929471 203.0.113.100 10100 typ host ]]></sourcecode>
]]> <t>|offer-B1-candidate-2| looks like:</t>
</artwork> <sourcecode name="offer-B1-candidate-2" type="sdp"><![CDATA[
</figure>
</t>
<t>|offer-B1-candidate-2| looks like:</t>
<t>
<figure>
<artwork alt="offer-B1-candidate-2">
<![CDATA[
ufrag ATEn ufrag ATEn
index 0 index 0
mid a1 mid a1
attr candidate:1 1 udp 1845494015 198.51.100.100 11100 typ srflx attr candidate:1 1 udp 1845494015 198.51.100.100 11100 typ srflx
raddr 203.0.113.100 rport 10100 raddr 203.0.113.100 rport 10100 ]]></sourcecode>
]]> <t>|offer-B1-candidate-3| looks like:</t>
</artwork> <sourcecode name="offer-B1-candidate-3" type="sdp"><![CDATA[
</figure>
</t>
<t>|offer-B1-candidate-3| looks like:</t>
<t>
<figure>
<artwork alt="offer-B1-candidate-3">
<![CDATA[
ufrag ATEn ufrag ATEn
index 0 index 0
mid a1 mid a1
attr candidate:1 1 udp 255 192.0.2.100 12100 typ relay attr candidate:1 1 udp 255 192.0.2.100 12100 typ relay
raddr 198.51.100.100 rport 11100 raddr 198.51.100.100 rport 11100 ]]></sourcecode>
]]> <t>The SDP for |answer-B1| looks like:</t>
</artwork> <sourcecode name="answer-B1" type="sdp"><![CDATA[
</figure>
</t>
<t>The SDP for |answer-B1| looks like:</t>
<t>
<figure>
<artwork alt="answer-B1">
<![CDATA[
v=0 v=0
o=- 7729291447651054566 1 IN IP4 0.0.0.0 o=- 7729291447651054566 1 IN IP4 0.0.0.0
s=- s=-
t=0 0 t=0 0
a=ice-options:trickle ice2 a=ice-options:trickle ice2
a=group:BUNDLE a1 d1 a=group:BUNDLE a1 d1
m=audio 9 UDP/TLS/RTP/SAVPF 96 0 8 97 98 m=audio 9 UDP/TLS/RTP/SAVPF 96 0 8 97 98
c=IN IP4 0.0.0.0 c=IN IP4 0.0.0.0
a=mid:a1 a=mid:a1
skipping to change at line 4604 skipping to change at line 5085
a=setup:active a=setup:active
a=tls-id:7a25ab85b195acaf3121f5a8ab4f0f71 a=tls-id:7a25ab85b195acaf3121f5a8ab4f0f71
a=rtcp-mux a=rtcp-mux
a=rtcp-mux-only a=rtcp-mux-only
a=rtcp-rsize a=rtcp-rsize
m=application 9 UDP/DTLS/SCTP webrtc-datachannel m=application 9 UDP/DTLS/SCTP webrtc-datachannel
c=IN IP4 0.0.0.0 c=IN IP4 0.0.0.0
a=mid:d1 a=mid:d1
a=sctp-port:5000 a=sctp-port:5000
a=max-message-size:65536 a=max-message-size:65536 ]]></sourcecode>
]]>
</artwork>
</figure>
</t>
<t>|answer-B1-candidate-1| looks like:</t> <t>|answer-B1-candidate-1| looks like:</t>
<t> <sourcecode name="answer-B1-candidate-1" type="sdp"><![CDATA[
<figure>
<artwork alt="answer-B1-candidate-1">
<![CDATA[
ufrag 7sFv ufrag 7sFv
index 0 index 0
mid a1 mid a1
attr candidate:1 1 udp 2113929471 203.0.113.200 10200 typ host attr candidate:1 1 udp 2113929471 203.0.113.200 10200 typ host ]]></sourcecode>
]]> <t>|answer-B1-candidate-2| looks like:</t>
</artwork> <sourcecode name="answer-B1-candidate-2" type="sdp"><![CDATA[
</figure>
</t>
<t>|answer-B1-candidate-2| looks like:</t>
<t>
<figure>
<artwork alt="answer-B1-candidate-2">
<![CDATA[
ufrag 7sFv ufrag 7sFv
index 0 index 0
mid a1 mid a1
attr candidate:1 1 udp 1845494015 198.51.100.200 11200 typ srflx attr candidate:1 1 udp 1845494015 198.51.100.200 11200 typ srflx
raddr 203.0.113.200 rport 10200 raddr 203.0.113.200 rport 10200 ]]></sourcecode>
]]> <t>|answer-B1-candidate-3| looks like:</t>
</artwork> <sourcecode name="answer-B1-candidate-3" type="sdp"><![CDATA[
</figure>
</t>
<t>|answer-B1-candidate-3| looks like:</t>
<t>
<figure>
<artwork alt="answer-B1-candidate-3">
<![CDATA[
ufrag 7sFv ufrag 7sFv
index 0 index 0
mid a1 mid a1
attr candidate:1 1 udp 255 192.0.2.200 12200 typ relay attr candidate:1 1 udp 255 192.0.2.200 12200 typ relay
raddr 198.51.100.200 rport 11200 raddr 198.51.100.200 rport 11200 ]]></sourcecode>
]]> <t>The SDP for |offer-B2| is shown below. In addition to the
</artwork> new "m=" sections for video, both of which are offering FEC and
</figure> one of which is offering simulcast, note the increment of the
</t> version number in the "o=" line; changes to the "c=" line,
indicating the local candidate that was selected; and the
<t>The SDP for |offer-B2| is shown below. In addition to the inclusion of gathered candidates as a=candidate lines.</t>
new m= sections for video, both of which are offering FEC, and <sourcecode name="offer-B2" type="sdp"><![CDATA[
one of which is offering simulcast, note the increment of the
version number in the o= line, changes to the c= line,
indicating the local candidate that was selected, and the
inclusion of gathered candidates as a=candidate lines.</t>
<t>
<figure>
<artwork alt="offer-B2">
<![CDATA[
v=0 v=0
o=- 7729291447651054566 2 IN IP4 0.0.0.0 o=- 7729291447651054566 2 IN IP4 0.0.0.0
s=- s=-
t=0 0 t=0 0
a=ice-options:trickle ice2 a=ice-options:trickle ice2
a=group:BUNDLE a1 d1 v1 v2 a=group:BUNDLE a1 d1 v1 v2
a=group:LS a1 v1 a=group:LS a1 v1
m=audio 12200 UDP/TLS/RTP/SAVPF 96 0 8 97 98 m=audio 12200 UDP/TLS/RTP/SAVPF 96 0 8 97 98
c=IN IP4 192.0.2.200 c=IN IP4 192.0.2.200
skipping to change at line 4754 skipping to change at line 5201
a=rtpmap:102 rtx/90000 a=rtpmap:102 rtx/90000
a=fmtp:102 apt=100 a=fmtp:102 apt=100
=rtpmap:103 rtx/90000 =rtpmap:103 rtx/90000
a=fmtp:103 apt=101 a=fmtp:103 apt=101
a=rtpmap:104 flexfec/90000 a=rtpmap:104 flexfec/90000
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=rtcp-fb:100 ccm fir a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli a=rtcp-fb:100 nack pli
a=msid:81317484-2ed4-49d7-9eb7-1414322a7aae a=msid:81317484-2ed4-49d7-9eb7-1414322a7aae ]]></sourcecode>
]]> <t>The SDP for |answer-B2| is shown below. In addition to the
</artwork> acceptance of the video "m=" sections, the use of a=recvonly to
</figure> indicate one-way video, and the use of a=imageattr to limit the
</t> received resolution, note the use of setup:passive to maintain
the existing DTLS roles.</t>
<t>The SDP for |answer-B2| is shown below. In addition to the <sourcecode name="answer-B2" type="sdp"><![CDATA[
acceptance of the video m= sections, the use of a=recvonly to
indicate one-way video, and the use of a=imageattr to limit the
received resolution, note the use of setup:passive to maintain
the existing DTLS roles.</t>
<t>
<figure>
<artwork alt="answer-B2">
<![CDATA[
v=0 v=0
o=- 4962303333179871723 2 IN IP4 0.0.0.0 o=- 4962303333179871723 2 IN IP4 0.0.0.0
s=- s=-
t=0 0 t=0 0
a=ice-options:trickle ice2 a=ice-options:trickle ice2
a=group:BUNDLE a1 d1 v1 v2 a=group:BUNDLE a1 d1 v1 v2
a=group:LS a1 v1 a=group:LS a1 v1
m=audio 12100 UDP/TLS/RTP/SAVPF 96 0 8 97 98 m=audio 12100 UDP/TLS/RTP/SAVPF 96 0 8 97 98
c=IN IP4 192.0.2.100 c=IN IP4 192.0.2.100
skipping to change at line 4850 skipping to change at line 5288
a=fmtp:101 packetization-mode=1;profile-level-id=42e01f a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
a=rtpmap:102 rtx/90000 a=rtpmap:102 rtx/90000
a=fmtp:102 apt=100 a=fmtp:102 apt=100
=rtpmap:103 rtx/90000 =rtpmap:103 rtx/90000
a=fmtp:103 apt=101 a=fmtp:103 apt=101
a=imageattr:100 recv [x=[48:1920],y=[48:1080],q=1.0] a=imageattr:100 recv [x=[48:1920],y=[48:1080],q=1.0]
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=rtcp-fb:100 ccm fir a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli a=rtcp-fb:100 nack pli ]]></sourcecode>
]]> </section>
</artwork> <section anchor="sec.warmup-example" numbered="true" toc="default">
</figure> <name>Early Transport Warmup Example</name>
</t> <t>This example demonstrates the early-warmup technique
</section> described in
<section title="Early Transport Warmup Example" <xref target="sec.use-of-provisional-answer" format="default"/>. Here,
anchor="sec.warmup-example"> Alice's
endpoint sends an offer to Bob's endpoint to start an
<t>This example demonstrates the early warmup technique audio/video call. Bob immediately responds with an answer that
described in accepts the audio/video "m=" sections but marks them as sendonly
<xref target="sec.use-of-provisional-answer" />. Here, Alice's (from his perspective), meaning that Alice will not yet send
endpoint sends an offer to Bob's endpoint to start an media. This allows the JSEP implementation to start negotiating
audio/video call. Bob immediately responds with an answer that ICE and DTLS immediately. Bob's endpoint then prompts him to
accepts the audio/video m= sections, but marks them as sendonly answer the call, and when he does, his endpoint sends a second
(from his perspective), meaning that Alice will not yet send offer, which enables the audio and video "m=" sections, and
media. This allows the JSEP implementation to start negotiating thereby bidirectional media transmission. The advantage of such
ICE and DTLS immediately. Bob's endpoint then prompts him to a flow is that as soon as the first answer is received, the
answer the call, and when he does, his endpoint sends a second implementation can proceed with ICE and DTLS negotiation and
offer which enables the audio and video m= sections, and establish the session transport. If the transport setup
thereby bidirectional media transmission. The advantage of such completes before the second offer is sent, then media can be
a flow is that as soon as the first answer is received, the transmitted by the callee immediately upon
implementation can proceed with ICE and DTLS negotiation and answering the call, minimizing perceived post-dial delay. The
establish the session transport. If the transport setup second offer/answer exchange can also change the preferred
completes before the second offer is sent, then media can be codecs or other session parameters.</t>
transmitted immediately by the callee immediately upon <t>This example also makes use of the "relay" ICE candidate
answering the call, minimizing perceived post-dial-delay. The policy described in
second offer/answer exchange can also change the preferred <xref target="sec.ice-candidate-policy" format="default"/> to minimize
codecs or other session parameters.</t> the ICE
gathering and checking needed.</t>
<t>This example also makes use of the "relay" ICE candidate
policy described in
<xref target="sec.ice-candidate-policy" /> to minimize the ICE
gathering and checking needed.</t>
<t> <sourcecode name="" type="pseudocode"><![CDATA[
<figure>
<artwork>
<![CDATA[
// set up local media state // set up local media state
AliceJS->AliceUA: create new PeerConnection with "relay" ICE policy AliceJS->AliceUA: create new PeerConnection with "relay" ICE policy
AliceJS->AliceUA: addTrack with two tracks: audio and video AliceJS->AliceUA: addTrack with two tracks: audio and video
AliceJS->AliceUA: createOffer to get |offer-C1| AliceJS->AliceUA: createOffer to get |offer-C1|
AliceJS->AliceUA: setLocalDescription with |offer-C1| AliceJS->AliceUA: setLocalDescription with |offer-C1|
// |offer-C1| is sent over signaling protocol to Bob // |offer-C1| is sent over signaling protocol to Bob
AliceJS->WebServer: signaling with |offer-C1| AliceJS->WebServer: signaling with |offer-C1|
WebServer->BobJS: signaling with |offer-C1| WebServer->BobJS: signaling with |offer-C1|
skipping to change at line 4911 skipping to change at line 5340
BobJS->BobUA: setRemoteDescription with |offer-C1| BobJS->BobUA: setRemoteDescription with |offer-C1|
BobUA->BobJS: ontrack events for audio and video BobUA->BobJS: ontrack events for audio and video
// a relay candidate is sent to Bob // a relay candidate is sent to Bob
AliceUA->AliceJS: onicecandidate (relay) |offer-C1-candidate-1| AliceUA->AliceJS: onicecandidate (relay) |offer-C1-candidate-1|
AliceJS->WebServer: signaling with |offer-C1-candidate-1| AliceJS->WebServer: signaling with |offer-C1-candidate-1|
WebServer->BobJS: signaling with |offer-C1-candidate-1| WebServer->BobJS: signaling with |offer-C1-candidate-1|
BobJS->BobUA: addIceCandidate with |offer-C1-candidate-1| BobJS->BobUA: addIceCandidate with |offer-C1-candidate-1|
// Bob prepares an early answer to warmup the transport // Bob prepares an early answer to warm up the
// transport
BobJS->BobUA: addTransceiver with null audio and video tracks BobJS->BobUA: addTransceiver with null audio and video tracks
BobJS->BobUA: transceiver.setDirection(sendonly) for both BobJS->BobUA: transceiver.setDirection(sendonly) for both
BobJS->BobUA: createAnswer BobJS->BobUA: createAnswer
BobJS->BobUA: setLocalDescription with answer BobJS->BobUA: setLocalDescription with answer
// |answer-C1| is sent over signaling protocol to Alice // |answer-C1| is sent over signaling protocol
// to Alice
BobJS->WebServer: signaling with |answer-C1| BobJS->WebServer: signaling with |answer-C1|
WebServer->AliceJS: signaling with |answer-C1| WebServer->AliceJS: signaling with |answer-C1|
// |answer-C1| (sendonly) arrives at Alice // |answer-C1| (sendonly) arrives at Alice
AliceJS->AliceUA: setRemoteDescription with |answer-C1| AliceJS->AliceUA: setRemoteDescription with |answer-C1|
AliceUA->AliceJS: ontrack events for audio and video AliceUA->AliceJS: ontrack events for audio and video
// a relay candidate is sent to Alice // a relay candidate is sent to Alice
BobUA->BobJS: onicecandidate (relay) |answer-B1-candidate-1| BobUA->BobJS: onicecandidate (relay) |answer-B1-candidate-1|
BobJS->WebServer: signaling with |answer-B1-candidate-1| BobJS->WebServer: signaling with |answer-B1-candidate-1|
WebServer->AliceJS: signaling with |answer-B1-candidate-1| WebServer->AliceJS: signaling with |answer-B1-candidate-1|
AliceJS->AliceUA: addIceCandidate with |answer-B1-candidate-1| AliceJS->AliceUA: addIceCandidate with |answer-B1-candidate-1|
// ICE and DTLS establish while call is ringing // ICE and DTLS establish while call is ringing
// Bob accepts call, starts media, and sends new offer // Bob accepts call, starts media, and sends
// new offer
BobJS->BobUA: transceiver.setTrack with audio and video tracks BobJS->BobUA: transceiver.setTrack with audio and video tracks
BobUA->AliceUA: media sent from Bob to Alice BobUA->AliceUA: media sent from Bob to Alice
BobJS->BobUA: transceiver.setDirection(sendrecv) for both BobJS->BobUA: transceiver.setDirection(sendrecv) for both
transceivers transceivers
BobJS->BobUA: createOffer BobJS->BobUA: createOffer
BobJS->BobUA: setLocalDescription with offer BobJS->BobUA: setLocalDescription with offer
// |offer-C2| is sent over signaling protocol to Alice // |offer-C2| is sent over signaling protocol
// to Alice
BobJS->WebServer: signaling with |offer-C2| BobJS->WebServer: signaling with |offer-C2|
WebServer->AliceJS: signaling with |offer-C2| WebServer->AliceJS: signaling with |offer-C2|
// |offer-C2| (sendrecv) arrives at Alice // |offer-C2| (sendrecv) arrives at Alice
AliceJS->AliceUA: setRemoteDescription with |offer-C2| AliceJS->AliceUA: setRemoteDescription with |offer-C2|
AliceJS->AliceUA: createAnswer AliceJS->AliceUA: createAnswer
AliceJS->AliceUA: setLocalDescription with |answer-C2| AliceJS->AliceUA: setLocalDescription with |answer-C2|
AliceUA->BobUA: media sent from Alice to Bob AliceUA->BobUA: media sent from Alice to Bob
// |answer-C2| is sent over signaling protocol to Bob // |answer-C2| is sent over signaling protocol
// to Bob
AliceJS->WebServer: signaling with |answer-C2| AliceJS->WebServer: signaling with |answer-C2|
WebServer->BobJS: signaling with |answer-C2| WebServer->BobJS: signaling with |answer-C2|
BobJS->BobUA: setRemoteDescription with |answer-C2| BobJS->BobUA: setRemoteDescription with |answer-C2| ]]></sourcecode>
]]> <t>The SDP for |offer-C1| looks like:</t>
</artwork> <sourcecode name="offer-C1" type="sdp"><![CDATA[
</figure>
</t>
<t>The SDP for |offer-C1| looks like:</t>
<t>
<figure>
<artwork alt="offer-C1">
<![CDATA[
v=0 v=0
o=- 1070771854436052752 1 IN IP4 0.0.0.0 o=- 1070771854436052752 1 IN IP4 0.0.0.0
s=- s=-
t=0 0 t=0 0
a=ice-options:trickle ice2 a=ice-options:trickle ice2
a=group:BUNDLE a1 v1 a=group:BUNDLE a1 v1
a=group:LS a1 v1 a=group:LS a1 v1
m=audio 9 UDP/TLS/RTP/SAVPF 96 0 8 97 98 m=audio 9 UDP/TLS/RTP/SAVPF 96 0 8 97 98
c=IN IP4 0.0.0.0 c=IN IP4 0.0.0.0
skipping to change at line 5018 skipping to change at line 5443
a=rtpmap:102 rtx/90000 a=rtpmap:102 rtx/90000
a=fmtp:102 apt=100 a=fmtp:102 apt=100
=rtpmap:103 rtx/90000 =rtpmap:103 rtx/90000
a=fmtp:103 apt=101 a=fmtp:103 apt=101
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=rtcp-fb:100 ccm fir a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli a=rtcp-fb:100 nack pli
a=msid:bbce3ba6-abfc-ac63-d00a-e15b286f8fce a=msid:bbce3ba6-abfc-ac63-d00a-e15b286f8fce
a=bundle-only a=bundle-only ]]></sourcecode>
]]> <t>|offer-C1-candidate-1| looks like:</t>
</artwork> <sourcecode name="offer-C1-candidate-1" type="sdp"><![CDATA[
</figure>
</t>
<t>|offer-C1-candidate-1| looks like:</t>
<t>
<figure>
<artwork alt="offer-C1-candidate-1">
<![CDATA[
ufrag 4ZcD ufrag 4ZcD
index 0 index 0
mid a1 mid a1
attr candidate:1 1 udp 255 192.0.2.100 12100 typ relay attr candidate:1 1 udp 255 192.0.2.100 12100 typ relay
raddr 0.0.0.0 rport 0 raddr 0.0.0.0 rport 0 ]]></sourcecode>
]]> <t>The SDP for |answer-C1| looks like:</t>
</artwork> <sourcecode name="answer-C1" type="sdp"><![CDATA[
</figure>
</t>
<t>The SDP for |answer-C1| looks like:</t>
<t>
<figure>
<artwork alt="answer-C1">
<![CDATA[
v=0 v=0
o=- 6386516489780559513 1 IN IP4 0.0.0.0 o=- 6386516489780559513 1 IN IP4 0.0.0.0
s=- s=-
t=0 0 t=0 0
a=ice-options:trickle ice2 a=ice-options:trickle ice2
a=group:BUNDLE a1 v1 a=group:BUNDLE a1 v1
a=group:LS a1 v1 a=group:LS a1 v1
m=audio 9 UDP/TLS/RTP/SAVPF 96 0 8 97 98 m=audio 9 UDP/TLS/RTP/SAVPF 96 0 8 97 98
c=IN IP4 0.0.0.0 c=IN IP4 0.0.0.0
skipping to change at line 5096 skipping to change at line 5503
a=fmtp:101 packetization-mode=1;profile-level-id=42e01f a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
a=rtpmap:102 rtx/90000 a=rtpmap:102 rtx/90000
a=fmtp:102 apt=100 a=fmtp:102 apt=100
=rtpmap:103 rtx/90000 =rtpmap:103 rtx/90000
a=fmtp:103 apt=101 a=fmtp:103 apt=101
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=rtcp-fb:100 ccm fir a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli a=rtcp-fb:100 nack pli
a=msid:751f239e-4ae0-c549-aa3d-890de772998b a=msid:751f239e-4ae0-c549-aa3d-890de772998b ]]></sourcecode>
]]> <t>|answer-C1-candidate-1| looks like:</t>
</artwork> <sourcecode name="answer-C1-candidate-1" type="sdp"><![CDATA[
</figure>
</t>
<t>|answer-C1-candidate-1| looks like:</t>
<t>
<figure>
<artwork alt="answer-C1-candidate-1">
<![CDATA[
ufrag TpaA ufrag TpaA
index 0 index 0
mid a1 mid a1
attr candidate:1 1 udp 255 192.0.2.200 12200 typ relay attr candidate:1 1 udp 255 192.0.2.200 12200 typ relay
raddr 0.0.0.0 rport 0 raddr 0.0.0.0 rport 0 ]]></sourcecode>
]]> <t>The SDP for |offer-C2| looks like:</t>
</artwork> <sourcecode name="offer-C2" type="sdp"><![CDATA[
</figure>
</t>
<t>The SDP for |offer-C2| looks like:</t>
<t>
<figure>
<artwork alt="offer-C2">
<![CDATA[
v=0 v=0
o=- 6386516489780559513 2 IN IP4 0.0.0.0 o=- 6386516489780559513 2 IN IP4 0.0.0.0
s=- s=-
t=0 0 t=0 0
a=ice-options:trickle ice2 a=ice-options:trickle ice2
a=group:BUNDLE a1 v1 a=group:BUNDLE a1 v1
a=group:LS a1 v1 a=group:LS a1 v1
m=audio 12200 UDP/TLS/RTP/SAVPF 96 0 8 97 98 m=audio 12200 UDP/TLS/RTP/SAVPF 96 0 8 97 98
c=IN IP4 192.0.2.200 c=IN IP4 192.0.2.200
skipping to change at line 5177 skipping to change at line 5566
a=fmtp:101 packetization-mode=1;profile-level-id=42e01f a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
a=rtpmap:102 rtx/90000 a=rtpmap:102 rtx/90000
a=fmtp:102 apt=100 a=fmtp:102 apt=100
=rtpmap:103 rtx/90000 =rtpmap:103 rtx/90000
a=fmtp:103 apt=101 a=fmtp:103 apt=101
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=rtcp-fb:100 ccm fir a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli a=rtcp-fb:100 nack pli
a=msid:751f239e-4ae0-c549-aa3d-890de772998b a=msid:751f239e-4ae0-c549-aa3d-890de772998b ]]></sourcecode>
]]> <t>The SDP for |answer-C2| looks like:</t>
</artwork> <sourcecode name="answer-C2" type="sdp"><![CDATA[
</figure>
</t>
<t>The SDP for |answer-C2| looks like:</t>
<t>
<figure>
<artwork alt="answer-C2">
<![CDATA[
v=0 v=0
o=- 1070771854436052752 2 IN IP4 0.0.0.0 o=- 1070771854436052752 2 IN IP4 0.0.0.0
s=- s=-
t=0 0 t=0 0
a=ice-options:trickle ice2 a=ice-options:trickle ice2
a=group:BUNDLE a1 v1 a=group:BUNDLE a1 v1
a=group:LS a1 v1 a=group:LS a1 v1
m=audio 12100 UDP/TLS/RTP/SAVPF 96 0 8 97 98 m=audio 12100 UDP/TLS/RTP/SAVPF 96 0 8 97 98
c=IN IP4 192.0.2.100 c=IN IP4 192.0.2.100
skipping to change at line 5242 skipping to change at line 5622
a=fmtp:101 packetization-mode=1;profile-level-id=42e01f a=fmtp:101 packetization-mode=1;profile-level-id=42e01f
a=rtpmap:102 rtx/90000 a=rtpmap:102 rtx/90000
a=fmtp:102 apt=100 a=fmtp:102 apt=100
=rtpmap:103 rtx/90000 =rtpmap:103 rtx/90000
a=fmtp:103 apt=101 a=fmtp:103 apt=101
a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=rtcp-fb:100 ccm fir a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli a=rtcp-fb:100 nack pli
a=msid:bbce3ba6-abfc-ac63-d00a-e15b286f8fce a=msid:bbce3ba6-abfc-ac63-d00a-e15b286f8fce ]]></sourcecode>
]]> </section>
</artwork> </section>
</figure> <section anchor="sec.security-considerations" numbered="true" toc="defaul
</t> t">
</section> <name>Security Considerations</name>
</section> <t>The IETF has published separate documents
<section title="Security Considerations" <xref target="RFC8827" format="default"/>
anchor="sec.security-considerations"> <xref target="RFC8826" format="default"/> describing the security
architecture for WebRTC as a whole. The remainder of this section
<t>The IETF has published separate documents describes security considerations for this document.</t>
<xref target="I-D.ietf-rtcweb-security-arch" /> <t>While formally the JSEP interface is an API, it is better to
<xref target="I-D.ietf-rtcweb-security" /> describing the security think of it as an Internet protocol, with the application
architecture for WebRTC as a whole. The remainder of this section JavaScript being untrustworthy from the perspective of the JSEP
describes security considerations for this document.</t> implementation. Thus, the threat model of
<xref target="RFC3552" format="default"/> applies. In particular, JavaSc
<t>While formally the JSEP interface is an API, it is better to ript can
think of it as an Internet protocol, with the application call the API in any order and with any inputs, including
JavaScript being untrustworthy from the perspective of the JSEP malicious ones. This is particularly relevant when we consider
implementation. Thus, the threat model of the SDP that is passed to setLocalDescription(). While correct
<xref target="RFC3552" /> applies. In particular, JavaScript can API usage requires that the application pass in SDP that was
call the API in any order and with any inputs, including derived from createOffer() or createAnswer(), there is no
malicious ones. This is particularly relevant when we consider guarantee that applications do so. The JSEP implementation <bcp14>MUST</
the SDP which is passed to setLocalDescription(). While correct bcp14>
API usage requires that the application pass in SDP which was be prepared for the JavaScript to pass in bogus data instead.</t>
derived from createOffer() or createAnswer(), there is no <t>Conversely, the application programmer needs to be aware that
guarantee that applications do so. The JSEP implementation MUST the JavaScript does not have complete control of endpoint
be prepared for the JavaScript to pass in bogus data instead.</t> behavior. One case that bears particular mention is that editing
ICE candidates out of the SDP or suppressing trickled candidates
<t>Conversely, the application programmer needs to be aware that does not have the expected behavior: implementations will still
the JavaScript does not have complete control of endpoint perform checks from those candidates even if they are not sent to
behavior. One case that bears particular mention is that editing the other side. Thus, for instance, it is not possible to prevent
ICE candidates out of the SDP or suppressing trickled candidates the remote peer from learning your public IP address by removing
does not have the expected behavior: implementations will still server-reflexive candidates. Applications that wish to conceal
perform checks from those candidates even if they are not sent to their public IP address should instead configure the ICE agent to
the other side. Thus, for instance, it is not possible to prevent use only relay candidates.</t>
the remote peer from learning your public IP address by removing </section>
server reflexive candidates. Applications which wish to conceal <section anchor="sec.iana-considerations" numbered="true" toc="default">
their public IP address should instead configure the ICE agent to <name>IANA Considerations</name>
use only relay candidates.</t> <t>This document has no IANA actions.</t>
</section> </section>
<section title="IANA Considerations" </middle>
anchor="sec.iana-considerations"> <back>
<!-- draft-ietf-rtcweb-sdp ("Publication Requested") -->
<t>This document requires no actions from IANA.</t> <displayreference target="I-D.ietf-rtcweb-sdp" to="SDP4WebRTC"/>
</section> <references>
<section title="Acknowledgements" anchor="sec.acknowledgements"> <name>References</name>
<references>
<t>Harald Alvestrand, Taylor Brandstetter, Suhas Nandakumar, and <name>Normative References</name>
Peter Thatcher provided significant text for this draft. Bernard
Aboba, Adam Bergkvist, Dan Burnett, Ben Campbell, Alissa Cooper,
Richard Ejzak, Stefan Hakansson, Ted Hardie, Christer Holmberg
Andrew Hutton, Randell Jesup, Matthew Kaufman, Anant Narayanan,
Adam Roach, Robert Sparks, Neil Stratford, Martin Thomson, Sean
Turner, and Magnus Westerlund all provided valuable feedback on
this proposal.</t>
</section>
</middle>
<back>
<references title="Normative References">
<?rfc include='reference.I-D.ietf-avtext-rid'?>
<?rfc include='reference.I-D.ietf-ice-trickle'?>
<?rfc include='reference.I-D.ietf-mmusic-dtls-sdp'?>
<?rfc include='reference.I-D.ietf-mmusic-ice-sip-sdp'?>
<?rfc include='reference.I-D.ietf-mmusic-msid'?>
<?rfc include='reference.I-D.ietf-mmusic-mux-exclusive'?>
<?rfc include='reference.I-D.ietf-mmusic-rid'?>
<?rfc include='reference.I-D.ietf-mmusic-sctp-sdp'?>
<?rfc include='reference.I-D.ietf-mmusic-sdp-bundle-negotiation'?>
<?rfc include='reference.I-D.ietf-mmusic-sdp-mux-attributes'?>
<?rfc include='reference.I-D.ietf-mmusic-sdp-simulcast'?>
<?rfc include='reference.I-D.ietf-rtcweb-fec'?>
<?rfc include='reference.I-D.ietf-rtcweb-rtp-usage'?>
<?rfc include='reference.I-D.ietf-rtcweb-security'?>
<?rfc include='reference.I-D.ietf-rtcweb-security-arch'?>
<?rfc include='reference.RFC.2119.xml'?>
<?rfc include='reference.RFC.3261.xml'?>
<?rfc include='reference.RFC.3264.xml'?>
<?rfc include='reference.RFC.3552.xml'?>
<?rfc include='reference.RFC.3605.xml'?>
<?rfc include='reference.RFC.3890.xml'?>
<?rfc include='reference.RFC.4145.xml'?>
<?rfc include='reference.RFC.4566.xml'?>
<?rfc include='reference.RFC.4585.xml'?>
<?rfc include='reference.RFC.5124.xml'?>
<?rfc include='reference.RFC.5285.xml'?>
<?rfc include='reference.RFC.5761.xml'?>
<?rfc include='reference.RFC.5888.xml'?>
<?rfc include='reference.RFC.6236.xml'?>
<?rfc include='reference.RFC.6347.xml'?>
<?rfc include='reference.RFC.6716.xml'?>
<?rfc include='reference.RFC.6904.xml'?>
<?rfc include='reference.RFC.7160.xml'?>
<?rfc include='reference.RFC.7587.xml'?>
<?rfc include='reference.RFC.7742.xml'?>
<?rfc include='reference.RFC.7850.xml'?>
<?rfc include='reference.RFC.7874.xml'?>
<?rfc include='reference.RFC.8108.xml'?>
<?rfc include='reference.RFC.8122.xml'?>
<?rfc include='reference.RFC.8445.xml'?>
<?rfc include='reference.RFC.3711.xml'?>
</references>
<references title="Informative References">
<?rfc include='reference.I-D.ietf-rtcweb-ip-handling'?>
<?rfc include='reference.I-D.ietf-mmusic-trickle-ice-sip'?>
<?rfc include='reference.I-D.ietf-rtcweb-sdp'?>
<?rfc include='reference.RFC.3389.xml'?>
<?rfc include='reference.RFC.3960.xml'?>
<?rfc include='reference.RFC.4568.xml'?>
<?rfc include='reference.RFC.4588.xml'?>
<?rfc include='reference.RFC.4733.xml'?>
<?rfc include='reference.RFC.5245.xml'?>
<?rfc include='reference.RFC.5506.xml'?>
<?rfc include='reference.RFC.5576.xml'?>
<?rfc include='reference.RFC.5763.xml'?>
<?rfc include='reference.RFC.5764.xml'?>
<?rfc include='reference.RFC.6464.xml'?>
<?rfc include='reference.RFC.6544.xml'?>
<?rfc include='reference.RFC.3556.xml'?>
<reference anchor="W3C.webrtc"
target="https://www.w3.org/TR/2017/WD-webrtc-20170515/">
<front>
<title>WebRTC 1.0: Real-time Communication Between
Browsers</title>
<author fullname="Adam Bergkvist" initials="A."
surname="Bergkvist">
<organization>Ericsson</organization>
</author>
<author fullname="Daniel C. Burnett" initials="D."
surname="Burnett">
<organization></organization>
</author>
<author fullname="Cullen Jennings" initials="C."
surname="Jennings">
<organization>Cisco</organization>
</author>
<author fullname="Anant Narayanan" initials="A."
surname="Narayanan">
<organization>Mozilla</organization>
</author>
<author fullname="Bernard Aboba" initials="B."
surname="Aboba">
<organization>Microsoft Corporation</organization>
</author>
<author fullname="Taylor Brandstetter" initials="T."
surname="Brandstetter">
<organization>Google</organization>
</author>
<date day="15" month="May" year="2017" />
</front>
<seriesInfo name="World Wide Web Consortium WD"
value="WD-webrtc-20170515" />
<format target="https://www.w3.org/TR/2017/WD-webrtc-20170515/"
type="HTML" />
</reference>
<reference anchor="TS26.114"
target="http://www.3gpp.org/DynaReport/26114.htm">
<front>
<title>3rd Generation Partnership Project; Technical
Specification Group Services and System Aspects; IP
Multimedia Subsystem (IMS); Multimedia Telephony; Media
handling and interaction (Release 12)</title>
<author>
<organization>3GPP TS 26.114 V12.8.0</organization>
</author>
<date year="2014" month="December" />
</front>
</reference>
</references>
<section title="Appendix A" anchor="sec.appendix-a">
<t>For the syntax validation performed in
<xref target="sec.parsing-a-desc" />, the following list of ABNF
definitions is used:</t>
<texttable anchor="sdp-abnf" title="SDP ABNF References">
<ttcol align='left'>Attribute</ttcol>
<ttcol align='left'>Reference</ttcol>
<c>ptime</c>
<c>
<xref target="RFC4566" /> Section 9</c>
<c>maxptime</c>
<c>
<xref target="RFC4566" /> Section 9</c>
<c>rtpmap</c>
<c>
<xref target="RFC4566" /> Section 9</c>
<c>recvonly</c>
<c>
<xref target="RFC4566" /> Section 9</c>
<c>sendrecv</c>
<c>
<xref target="RFC4566" /> Section 9</c>
<c>sendonly</c>
<c>
<xref target="RFC4566" /> Section 9</c>
<c>inactive</c>
<c>
<xref target="RFC4566" /> Section 9</c>
<c>framerate</c>
<c>
<xref target="RFC4566" /> Section 9</c>
<c>fmtp</c>
<c>
<xref target="RFC4566" /> Section 9</c>
<c>quality</c>
<c>
<xref target="RFC4566" /> Section 9</c>
<c>rtcp</c>
<c>
<xref target="RFC3605" /> Section 2.1</c>
<c>setup</c>
<c>
<xref target="RFC4145" /> Sections 3, 4, and 5</c>
<c>connection</c>
<c>
<xref target="RFC4145" /> Sections 3, 4, and 5</c>
<c>fingerprint</c>
<c>
<xref target="RFC8122" /> Section 5</c>
<c>rtcp-fb</c>
<c>
<xref target="RFC4585" /> Section 4.2</c>
<c>extmap</c>
<c>
<xref target="RFC5285" /> Section 7</c>
<c>mid</c>
<c>
<xref target="RFC5888" /> Sections 4 and 5</c>
<c>group</c>
<c>
<xref target="RFC5888" /> Sections 4 and 5</c>
<c>imageattr</c>
<c>
<xref target="RFC6236" /> Section 3.1</c>
<c>extmap (encrypt option)</c>
<c>
<xref target="RFC6904" /> Section 4</c>
<c>candidate</c>
<c>
<xref target="I-D.ietf-mmusic-ice-sip-sdp" /> Section 4.1</c>
<c>remote-candidates</c>
<c>
<xref target="I-D.ietf-mmusic-ice-sip-sdp" /> Section 4.2</c>
<c>ice-lite</c>
<c>
<xref target="I-D.ietf-mmusic-ice-sip-sdp" /> Section 4.3</c>
<c>ice-ufrag</c>
<c>
<xref target="I-D.ietf-mmusic-ice-sip-sdp" /> Section 4.4</c>
<c>ice-pwd</c>
<c>
<xref target="I-D.ietf-mmusic-ice-sip-sdp" /> Section 4.4</c>
<c>ice-options</c>
<c>
<xref target="I-D.ietf-mmusic-ice-sip-sdp" /> Section 4.6</c>
<c>msid</c>
<c>
<xref target="I-D.ietf-mmusic-msid" /> Section 2</c>
<c>rid</c>
<c>
<xref target="I-D.ietf-mmusic-rid" /> Section 10</c>
<c>simulcast</c>
<c>
<xref target="I-D.ietf-mmusic-sdp-simulcast" /> Section 6.1</c>
<c>tls-id</c>
<c>
<xref target="I-D.ietf-mmusic-dtls-sdp" /> Section 4</c>
</texttable>
</section>
<section title="Change log" anchor="sec.change-log">
<t>Note to RFC Editor: Please remove this section before
publication.</t>
<t>Changes in draft-26:</t>
<t>
<list style="symbols">
<t>Update guidance on generation of the m= proto value to be
consistent with ice-sip-sdp.</t>
</list>
</t>
<t>Changes in draft-25:</t>
<t>
<list style="symbols">
<t>Remove MSID track ID from offers and answers.</t>
<t>Add note about rejecting all m= sections in a BUNDLE group.</t>
<t>Update ICE references to RFC 8445 and mention ice2.</t>
</list>
</t>
<t>Changes in draft-24:</t>
<t>
<list style="symbols">
<t>Clarify that rounding is permitted when trying to maintain
aspect ratio.</t>
<t>Update tls-id handling to match what is specified in
dtls-sdp.</t>
</list>
</t>
<t>Changes in draft-23:</t>
<t>
<list style="symbols">
<t>Clarify rollback handling, and treat it similarly to other
setLocal/setRemote usages.</t>
<t>Adopt a first-fit policy for handling multiple remote
a=imageattr attributes.</t>
<t>Clarify that a session description with zero m= sections
is legal.</t>
</list>
</t>
<t>Changes in draft-22:</t>
<t>
<list style="symbols">
<t>Clarify currentDirection versus direction.</t>
<t>Correct session-id text so that it aligns with RFC
3264.</t>
<t>Clarify that generated ICE candidate objects must have all
four fields.</t>
<t>Make rollback work from any state besides stable and
regardless of whether setLocalDescription or
setRemoteDescription is used.</t>
<t>Allow modifying SDP before sending or after receiving
either offers or answers (previously this was forbidden for
answers).</t>
<t>Provide rationale for several design choices.</t>
</list>
</t>
<t>Changes in draft-21:</t>
<t>
<list style="symbols">
<t>Change dtls-id to tls-id to match MMUSIC draft.</t>
<t>Replace regular expression for proto field with a list and
clarify that the answer must exactly match the offer.</t>
<t>Remove text about how to error check on setLocal because
local descriptions cannot be changed.</t>
<t>Rework silence suppression support to always require that
both sides agree to silence suppression or none is used.</t>
<t>Remove instructions to parse "a=ssrc-group".</t>
<t>Allow the addition of new codecs in answers and in
subsequent offers.</t>
<t>Clarify imageattr processing. Replace use of [x=0,y=0]
with direction indicators.</t>
<t>Document when early media can occur.</t>
<t>Fix ICE default port handling when bundle-only is
used.</t>
<t>Forbid duplicating IDENTICAL/TRANSPORT attributes when you
are bundling.</t>
<t>Clarify the number of components to gather when bundle is
involved.</t>
<t>Explicitly state that PTs and SSRCs are to be used for
demuxing.</t>
<t>Update guidance on "a=setup" line. This should now match
the MMUSIC draft.</t>
<t>Update guidance on certificate/digest matching to conform
to RFC8122.</t>
<t>Update examples.</t>
</list>
</t>
<t>Changes in draft-20:</t>
<t>
<list style="symbols">
<t>Remove Appendix-B.</t>
</list>
</t>
<t>Changes in draft-19:</t>
<t>
<list style="symbols">
<t>Examples are now machine-generated for correctness, and
use IETF-approved example IP addresses.</t>
<t>Add early transport warmup example, and add missing
attributes to existing examples.</t>
<t>Only send "a=rtcp-mux-only" and "a=bundle-only" on new m=
sections.</t>
<t>Update references.</t>
<t>Add coverage of a=identity.</t>
<t>Explain the lipsync group algorithm more thoroughly.</t>
<t>Remove unnecessary list of MTI specs.</t>
<t>Allow codecs which weren't offered to appear in answers
and which weren't selected to appear in subsequent
offers.</t>
<t>Codec preferences now are applied on both initial and
subsequent offers and answers.</t>
<t>Clarify a=msid handling for recvonly m= sections.</t>
<t>Clarify behavior of attributes for bundle-only data
channels.</t>
<t>Allow media attributes to appear in data m= sections when
all the media m= sections are bundle-only.</t>
<t>Use consistent terminology for JSEP implementations.</t>
<t>Describe how to handle failed API calls.</t>
<t>Some cleanup on routing rules.</t>
</list>
</t>
<t>Changes in draft-18:</t>
<t>
<list style="symbols">
<t>Update demux algorithm and move it to an appendix in
preparation for merging it into BUNDLE.</t>
<t>Clarify why we can't handle an incoming offer to send
simulcast.</t>
<t>Expand IceCandidate object text.</t>
<t>Further document use of ICE candidate pool.</t>
<t>Document removeTrack.</t>
<t>Update requirements to only accept the last generated
offer/answer as an argument to setLocalDescription.</t>
<t>Allow round pixels.</t>
<t>Fix code around default timing when AVPF is not
specified.</t>
<t>Clean up terminology around m= line and m=section.</t>
<t>Provide a more realistic example for minimum decoder
capabilities.</t>
<t>Document behavior when rtcp-mux policy is require but
rtcp-mux attribute not provided.</t>
<t>Expanded discussion of RtpSender and RtpReceiver.</t>
<t>Add RtpTransceiver.currentDirection and document
setDirection.</t>
<t>Require imageattr x=0, y=0 to indicate that there are no
valid resolutions.</t>
<t>Require a privacy-preserving MID/RID construction.</t>
<t>Require support for RFC 3556 bandwidth modifiers.</t>
<t>Update maxptime description.</t>
<t>Note that endpoints may encounter extra codecs in answers
and subsequent offers from non-JSEP peers.</t>
<t>Update references.</t>
</list>
</t>
<t>Changes in draft-17:</t>
<t>
<list style="symbols">
<t>Split createOffer and createAnswer sections to clearly
indicate attributes which always appear and which only appear
when not bundled into another m= section.</t>
<t>Add descriptions of RtpTransceiver methods.</t>
<t>Describe how to process RTCP feedback attributes.</t>
<t>Clarify transceiver directions and their interaction with
3264.</t>
<t>Describe setCodecPreferences.</t>
<t>Update RTP demux algorithm. Include RTCP.</t>
<t>Update requirements for when a=rtcp is included, limiting
to cases where it is needed for backward compatibility.</t>
<t>Clarify SAR handling.</t>
<t>Updated addTrack matching algorithm.</t>
<t>Remove a=ssrc requirements.</t>
<t>Handle a=setup in reoffers.</t>
<t>Discuss how RTX/FEC should be handled.</t>
<t>Discuss how telephone-event should be handled.</t>
<t>Discuss how CN/DTX should be handled.</t>
<t>Add missing references to ABNF table.</t>
</list>
</t>
<t>Changes in draft-16:</t>
<t>
<list style="symbols">
<t>Update addIceCandidate to indicate ICE generation and
allow per-m= section end-of-candidates.</t>
<t>Update fingerprint handling to use
draft-ietf-mmusic-4572-update.</t>
<t>Update text around SDP processing of RTP header extensions
and payload formats.</t>
<t>Add sections on simulcast, addTransceiver, and
createDataChannel.</t>
<t>Clarify text to ensure that the session ID is a positive
63 bit integer.</t>
<t>Clarify SDP processing for direction indication.</t>
<t>Describe SDP processing for rtcp-mux-only.</t>
<t>Specify how SDP session version in o= line.</t>
<t>Require that when doing an re-offer, the capabilities of
the new session are mostly required to be a subset of the
previously negotiated session.</t>
<t>Clarified ICE restart interaction with bundle-only.</t>
<t>Remove support for changing SDP before calling
setLocalDescription.</t>
<t>Specify algorithm for demuxing RTP based on MID, PT, and
SSRC.</t>
<t>Clarify rules for rejecting m= lines when bundle policy is
balanced or max-bundle.</t>
</list>
</t>
<t>Changes in draft-15:</t>
<t>
<list style="symbols">
<t>Clarify text around codecs offered in subsequent
transactions to refer to what's been negotiated.</t>
<t>Rewrite LS handling text to indicate edge cases and that
we're living with them.</t>
<t>Require that answerer reject m= lines when there are no
codecs in common.</t>
<t>Enforce max-bundle on offer processing.</t>
<t>Fix TIAS formula to handle bits vs. kilobits.</t>
<t>Describe addTrack algorithm.</t>
<t>Clean up references.</t>
</list>
</t>
<t>Changes in draft-14:</t>
<t>
<list style="symbols">
<t>Added discussion of RtpTransceivers + RtpSenders +
RtpReceivers, and how they interact with
createOffer/createAnswer.</t>
<t>Removed obsolete OfferToReceiveX options.</t>
<t>Explained how addIceCandidate can be used for
end-of-candidates.</t>
</list>
</t>
<t>Changes in draft-13:</t>
<t> <!--draft-ietf-mmusic-trickle-ice-sip-18: 8840 -->
<list style="symbols"> <reference anchor="RFC8840" target="https://www.rfc-editor.org/info/rf
c8840">
<front>
<title>A Session Initiation Protocol (SIP) Usage for Incremental
Provisioning of Candidates for the Interactive Connectivity
Establishment (Trickle ICE)</title>
<t>Clarified which SDP lines can be ignored.</t> <author initials="E" surname="Ivov" fullname="Emil Ivov">
<organization/>
</author>
<author initials="T" surname="Stach" fullname="Thomas Stach">
<organization/>
</author>
<author initials="E" surname="Marocco" fullname="Enrico Marocco">
<organization/>
</author>
<author initials="C" surname="Holmberg" fullname="Christer Holmber
g">
<organization/>
</author>
<date month="July" year="2018"/>
</front>
<seriesInfo name="DOI" value="10.17487/RFC8840"/>
<seriesInfo name="RFC" value="8840"/>
</reference>
<t>Clarified how to handle various received attributes.</t> <!-- draft-ietf-avtext-rid-09; 8852 -->
<reference anchor="RFC8852" target="https://www.rfc-editor.org/info/rfc8852">
<front>
<title>RTP Stream Identifier Source Description (SDES)</title>
<author initials="A.B." surname="Roach" fullname="Adam Roach"/>
<author initials="S" surname="Nandakumar" fullname="Suhas Nandakumar"/>
<author initials="P" surname="Thatcher" fullname="Peter Thatcher"/>
<date month="July" year="2020"/>
</front>
<seriesInfo name="DOI" value="10.17487/RFC8852"/>
<seriesInfo name="RFC" value="8852"/>
</reference>
<t>Revised how attributes should be generated for bundled m= <!-- draft-ietf-ice-trickle: RFC 8838 -->
lines.</t> <reference anchor="RFC8838" target="https://www.rfc-editor.org/info/rfc8838">
<front>
<title>Trickle ICE: Incremental Provisioning of Candidates for the
Interactive Connectivity Establishment (ICE) Protocol</title>
<t>Remove unused references.</t> <author initials="E" surname="Ivov" fullname="Emil Ivov">
<organization />
</author>
<t>Remove text advocating use of unilateral PTs.</t> <author initials="J" surname="Uberti" fullname="Justin Uberti">
<organization />
</author>
<t>Trigger an ICE restart even if the ICE candidate policy is <author initials="P" surname="Saint-Andre" fullname="Peter Saint-Andre">
being made more strict.</t> <organization />
</author>
<t>Remove the 'public' ICE candidate policy.</t> <date month="July" year="2020" />
</front>
<seriesInfo name="RFC" value="8838" />
<seriesInfo name="DOI" value="10.17487/RFC8838"/>
</reference>
<t>Move open issues into GitHub issues.</t> <!-- draft-ietf-mmusic-dtls-sdp RFC-to-be 8842 -->
<reference anchor="RFC8842" target="https://www.rfc-editor.org/info/rfc8842">
<t>Split local/remote description accessors into <front>
current/pending.</t> <title>Session Description Protocol (SDP) Offer/Answer Considerations for
Datagram Transport Layer Security (DTLS) and Transport Layer Security (TLS)</tit
le>
<t>Clarify a=imageattr handling.</t> <author initials="C." surname="Holmberg" fullname="Christer Holmberg">
<organization />
</author>
<t>Add more detail on VoiceActivityDetection handling.</t> <author initials="R." surname="Shpount" fullname="Roman Shpount">
<organization />
</author>
<t>Reference draft-shieh-rtcweb-ip-handling.</t> <date month="July" year="2020" />
</front>
<seriesInfo name="RFC" value="8842" />
<seriesInfo name="DOI" value="10.17487/RFC8842"/>
<t>Make it clear when an ICE restart should occur.</t> </reference>
<t>Resolve changes needed in references.</t> <!-- draft-ietf-mmusic-ice-sip-sdp-39 RFC-to-be 8839 -->
<t>Remove MSID semantics.</t> <reference anchor='RFC8839' target="https://www.rfc-editor.org/info/rfc8839">
<front>
<title>Session Description Protocol (SDP) Offer/Answer Procedures for Interactiv
e Connectivity Establishment (ICE)</title>
<t>ice-options are now at session level.</t> <author initials='M' surname='Petit-Huguenin' fullname='Marc Petit-Huguenin'>
<organization />
</author>
<t>Default RTCP mux policy is now 'require'.</t> <author initials='S' surname='Nandakumar' fullname='Suhas Nandakumar'>
</list> <organization />
</t> </author>
<t>Changes in draft-12:</t> <author initials='C' surname='Holmberg' fullname='Christer Holmberg'>
<organization />
</author>
<t> <author initials='A' surname='Keränen' fullname='Ari Keränen'>
<list style="symbols"> <organization />
</author>
<t>Filled in sections on applying local and remote <author initials='R' surname='Shpount' fullname='Roman Shpount'>
descriptions.</t> <organization />
</author>
<t>Discussed downscaling and upscaling to fulfill imageattr <date month="July" year="2020"/>
requirements.</t>
<t>Updated what SDP can be modified by the application.</t> </front>
<seriesInfo name="RFC" value="8839"/>
<seriesInfo name="DOI" value="10.17487/RFC8839"/>
<t>Updated to latest datachannel SDP.</t> </reference>
<t>Allowed multiple fingerprint lines.</t> <!-- draft-ietf-mmusic-msid: 8830 -->
<reference anchor="RFC8830" target="https://www.rfc-editor.org/info/rfc8830">
<front>
<title>WebRTC MediaStream Identification in the Session Description Protocol
</title>
<author initials="H" surname="Alvestrand" fullname="Harald Alvestrand">
<organization />
</author>
<date month="July" year="2020" />
</front>
<seriesInfo name="RFC" value="8830" />
<seriesInfo name="DOI" value="10.17487/RFC8830"/>
</reference>
<t>Switched back to IPv4 for dummy candidates.</t> <!--draft-ietf-mmusic-mux-exclusive-12; part of C238; RFC 8858-->
<reference anchor='RFC8858' target="https://www.rfc-editor.org/info/rfc8858">
<front>
<title>Indicating Exclusive Support of RTP and RTP Control Protocol (RTCP)
Multiplexing Using the Session Description Protocol (SDP)</title>
<author initials='C.' surname='Holmberg' fullname='Christer Holmberg'>
<organization />
</author>
<t>Added additional clarity on ICE default candidates.</t> <date month="July" year='2020' />
</list> </front>
</t> <seriesInfo name='RFC' value='8858' />
<seriesInfo name="DOI" value="10.17487/RFC8858"/>
</reference>
<t>Changes in draft-11:</t> <!-- draft-ietf-mmusic-rid: 8851 -->
<reference anchor="RFC8851" target="https://www.rfc-editor.org/info/rfc8851">
<front>
<title>RTP Payload Format Restrictions</title>
<author initials="A.B." surname="Roach" fullname="Adam Roach" role="editor">
<organization/>
</author>
<date month="July" year="2020"/>
</front>
<seriesInfo name="DOI" value="10.17487/RFC8851"/>
<seriesInfo name="RFC" value="8851"/>
</reference>
<t> <!-- draft-ietf-mmusic-sctp-sdp: 8841 -->
<list style="symbols">
<t>Clarified handling of RTP CNAMEs.</t> <reference anchor="RFC8841" target="https://www.rfc-editor.org/info/rfc8841">
<t>Updated what SDP lines should be processed or ignored.</t> <front>
<title>Session Description Protocol (SDP) Offer/Answer Procedures for
Stream Control Transmission Protocol (SCTP) over Datagram Transport Layer
Security (DTLS) Transport</title>
<t>Specified how a=imageattr should be used.</t> <author initials="C." surname="Holmberg" fullname="Christer Holmberg">
</list> <organization />
</t> </author>
<t>Changes in draft-10:</t> <author initials="R." surname="Shpount" fullname="Roman Shpount">
<organization />
</author>
<t> <author initials="S." surname="Loreto" fullname="Salvatore Loreto">
<list style="symbols"> <organization />
</author>
<t>Described video size negotiation with imageattr.</t> <author initials="G." surname="Camarillo" fullname="Gonzalo Camarillo">
<organization />
</author>
<t>Clarified rejection of sections that do not have <date month="July" year="2020" />
mux-only.</t> </front>
<seriesInfo name="RFC" value="8841" />
<seriesInfo name="DOI" value="10.17487/RFC8841"/>
<t>Add handling of LS groups</t> </reference>
</list>
</t>
<t>Changes in draft-09:</t> <!-- draft-ietf-mmusic-sdp-bundle-negotiation (RFC 8843) -->
<reference anchor="RFC8843" target="https://www.rfc-editor.org/info/rfc8843"
>
<front>
<title>Negotiating Media Multiplexing Using the Session Description Prot
ocol (SDP)</title>
<author initials="C" surname="Holmberg" fullname="Christer Holmberg">
<organization/>
</author>
<author initials="H" surname="Alvestrand" fullname="Harald Alvestrand">
<organization/>
</author>
<author initials="C" surname="Jennings" fullname="Cullen Jennings">
<organization/>
</author>
<date month="July" year="2020"/>
</front>
<seriesInfo name="RFC" value="8843"/>
<seriesInfo name="DOI" value="10.17487/RFC8843"/>
</reference>
<t> <!-- draft-ietf-mmusic-sdp-mux-attributes-17 (RFC 8859) -->
<list style="symbols"> <reference anchor="RFC8859" target="https://www.rfc-editor.org/info/rfc8859">
<front>
<title>A Framework for Session Description Protocol (SDP)
Attributes When Multiplexing</title>
<author initials="S" surname="Nandakumar" fullname="Suhas Nandakumar">
<organization/>
</author>
<date month="July" year="2020"/>
</front>
<seriesInfo name="DOI" value="10.17487/RFC8859"/>
<seriesInfo name="RFC" value="8859"/>
<t>Don't return null for {local,remote}Description after </reference>
close().</t>
<t>Changed TCP/TLS to UDP/DTLS in RTP profile names.</t> <!-- draft-ietf-mmusic-sdp-simulcast: 8853 -->
<reference anchor="RFC8853" target="https://www.rfc-editor.org/info/rfc8853">
<front>
<title>Using Simulcast in Session Description Protocol (SDP) and RTP
Sessions</title>
<t>Separate out bundle and mux policy.</t> <author initials="B" surname="Burman" fullname="Bo Burman">
<organization/>
</author>
<author initials="M" surname="Westerlund" fullname="Magnus Westerlund">
<organization/>
</author>
<author initials="S" surname="Nandakumar" fullname="Suhas Nandakumar">
<organization/>
</author>
<author initials="M" surname="Zanaty" fullname="Mo Zanaty">
<organization/>
</author>
<date month="July" year="2020"/>
</front>
<seriesInfo name="DOI" value="10.17487/RFC8853"/>
<seriesInfo name="RFC" value="8853"/>
</reference>
<t>Added specific references to FEC mechanisms.</t> <!-- draft-ietf-rtcweb-fec: 8854 -->
<reference anchor="RFC8854" target="https://www.rfc-editor.org/info/rfc8854">
<front>
<title>WebRTC Forward Error Correction Requirements</title>
<author initials="J." surname="Uberti" fullname="Justin Uberti">
<organization/>
</author>
<date month="July" year="2020"/>
</front>
<seriesInfo name="RFC" value="8854"/>
<seriesInfo name="DOI" value="10.17487/RFC8854"/>
</reference>
<t>Added canTrickle mechanism.</t> <!-- draft-ietf-rtcweb-rtp-usage; RFC 8834 -->
<reference anchor="RFC8834" target="https://www.rfc-editor.org/info/rfc8834">
<front>
<title>Media Transport and Use of RTP in WebRTC</title>
<author initials="C." surname="Perkins" fullname="Colin Perkins">
<organization />
</author>
<author initials="M." surname="Westerlund" fullname="Magnus Westerlund">
<organization />
</author>
<author initials="J." surname="Ott" fullname="Jörg Ott">
<organization />
</author>
<date month="July" year="2020" />
</front>
<seriesInfo name="RFC" value="8834" />
<seriesInfo name="DOI" value="10.17487/RFC8834"/>
</reference>
<t>Added section on subsequent answers and, answer <!--draft-ietf-rtcweb-security: RFC 8826 -->
options.</t> <reference anchor="RFC8826" target="https://www.rfc-editor.org/info/rfc8826">
<front>
<title>Security Considerations for WebRTC</title>
<author initials='E.' surname='Rescorla' fullname='Eric Rescorla'>
<organization/>
</author>
<date month='July' year='2020'/>
</front>
<seriesInfo name="RFC" value="8826"/>
<seriesInfo name="DOI" value="10.17487/RFC8826"/>
</reference>
<t>Added text defining set{Local,Remote}Description <!--draft-ietf-rtcweb-security-arch: 8827 -->
behavior.</t> <reference anchor="RFC8827" target="https://www.rfc-editor.org/info/rfc8827">
</list> <front>
</t> <title>WebRTC Security Architecture</title>
<author initials='E.' surname='Rescorla' fullname='Eric Rescorla'>
<organization/>
</author>
<date month='July' year='2020'/>
</front>
<seriesInfo name="RFC" value="8827"/>
<seriesInfo name="DOI" value="10.17487/RFC8827"/>
</reference>
<t>Changes in draft-08: <xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.2119.
<list style="symbols"> xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.8174.
xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.3261.
xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.3264.
xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.3552.
xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.3605.
xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.3890.
xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.4145.
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<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.4566.
xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.4585.
xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.5124.
xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.5285.
xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.5761.
xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.5888.
xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.6236.
xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.6347.
xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.6716.
xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.6904.
xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.7160.
xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.7587.
xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.7742.
xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.7850.
xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.7874.
xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.8108.
xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.8122.
xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.8445.
xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.3711.
xml"/>
</references>
<references>
<name>Informative References</name>
<t>Added new example section and removed old examples in <!-- draft-ietf-rtcweb-ip-handling: 8828 -->
appendix.</t> <reference anchor="RFC8828" target="https://www.rfc-editor.org/info/rfc8828">
<front>
<title>WebRTC IP Address Handling Requirements</title>
<author initials="J" surname="Uberti" fullname="Justin Uberti">
<organization />
</author>
<t>Fixed &lt;proto&gt; field handling.</t> <date month="July" year="2020" />
</front>
<seriesInfo name="RFC" value="8828" />
<seriesInfo name="DOI" value="10.17487/RFC8828"/>
</reference>
<t>Added text describing a=rtcp attribute.</t> <!-- 7/6/2020: IESG eval -->
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml3/reference.I-D.ietf
-rtcweb-sdp.xml"/>
<t>Reworked handling of OfferToReceiveAudio and <xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.3389.
OfferToReceiveVideo per discussion at IETF 90.</t> xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.4568.
xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.4588.
xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.4733.
xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.5245.
xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.5506.
xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.5576.
xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.5763.
xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.5764.
xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.6120.
xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.6464.
xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.6544.
xml"/>
<t>Reworked trickle ICE handling and its impact on m= and c= <!-- [rfced] Informative References: RFC 6544 is not cited anywhere
lines per discussion at interim.</t> in this document. Please let us know where it should be cited.
<t>Added max-bundle-and-rtcp-mux policy.</t> Original:
[RFC6544] Rosenberg, J., Keranen, A., Lowekamp, B., and A. Roach,
"TCP Candidates with Interactive Connectivity
Establishment (ICE)", RFC 6544, DOI 10.17487/RFC6544,
March 2012, <https://www.rfc-editor.org/info/rfc6544>. -->
<t>Added description of maxptime handling.</t> <xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.3556.
xml"/>
<xi:include href="https://xml2rfc.ietf.org/public/rfc/bibxml/reference.RFC.3960.
xml"/>
<t>Updated ICE candidate pool default to 0.</t> <reference anchor="W3C.webrtc" target="https://www.w3.org/TR/2017/WD-web
rtc-20170515/">
<front>
<title>WebRTC 1.0: Real-time Communication Between
Browsers</title>
<author fullname="Adam Bergkvist" initials="A." surname="Bergkvist">
<organization>Ericsson</organization>
</author>
<author fullname="Daniel C. Burnett" initials="D." surname="Burnett"
>
<organization/>
</author>
<author fullname="Cullen Jennings" initials="C." surname="Jennings">
<organization>Cisco</organization>
</author>
<author fullname="Anant Narayanan" initials="A." surname="Narayanan"
>
<organization>Mozilla</organization>
</author>
<author fullname="Bernard Aboba" initials="B." surname="Aboba">
<organization>Microsoft Corporation</organization>
</author>
<author fullname="Taylor Brandstetter" initials="T." surname="Brands
tetter">
<organization>Google</organization>
</author>
<date month="May" year="2017"/>
</front>
<refcontent>World Wide Web Consortium WD WD-webrtc-20170515</refcont
ent>
</reference>
<t>Resolved open issues around AppID/receiver-ID.</t> <!-- [rfced] Informative References: The provided URL for
[W3C.webrtc] steers to a page with a red window that says
"This version is outdated!" Because the latest version
(September 2018) also discusses the RTCPeerConnection interface,
may we update this listing as suggested below?
<t>Reworked and expanded how changes to the ICE configuration Also, if you agree to this update, should Jan-Ivar Bruaroey be
are handled.</t> added to the second sentence of the Acknowledgements section
(after Adam Bergkvist)?
<t>Some reference updates.</t> Original:
[W3C.webrtc]
Bergkvist, A., Burnett, D., Jennings, C., Narayanan, A.,
Aboba, B., and T. Brandstetter, "WebRTC 1.0: Real-time
Communication Between Browsers", World Wide Web Consortium
WD WD-webrtc-20170515, May 2017,
<https://www.w3.org/TR/2017/WD-webrtc-20170515/>.
<t>Editorial clarification.</t> Suggested:
</list></t> [W3C.webrtc]
Bergkvist, A., Burnett, D., Jennings, C., Narayanan, A.,
Aboba, B., Brandstetter, T., and J-I. Bruaroey, "WebRTC
1.0: Real-time Communication Between Browsers", World
Wide Web Consortium Candidate Recommendation, September
2018, <https://www.w3.org/TR/webrtc/>. -->
<t>Changes in draft-07: <reference anchor="TS26.114" target="https://www.3gpp.org/DynaReport/261
<list style="symbols"> 14.htm">
<front>
<title>3rd Generation Partnership Project; Technical
Specification Group Services and System Aspects; IP
Multimedia Subsystem (IMS); Multimedia Telephony; Media
handling and interaction (Release 12)</title>
<seriesInfo name="3GPP TS" value="26.114 V12.8.0"/>
<author>
<organization>3GPP</organization>
</author>
<date year="2014" month="December"/>
</front>
</reference>
<t>Expanded discussion of VAD and Opus DTX.</t> <!-- [rfced] Informative References: We see on
<https://www.3gpp.org/DynaReport/26114.htm> that a newer version
dated September 2019 is available. The new version also discusses
CVO (as mentioned in Section 3.6.2 of this document). May we update
this listing as suggested below?
<t>Added a security considerations section.</t> Original:
This is required regardless of whether the receiver
supports performing receive-side rotation (e.g., through CVO
[TS26.114]), as it significantly simplifies the matching logic.
...
[TS26.114]
3GPP TS 26.114 V12.8.0, "3rd Generation Partnership
Project; Technical Specification Group Services and System
Aspects; IP Multimedia Subsystem (IMS); Multimedia
Telephony; Media handling and interaction (Release 12)",
December 2014, <http://www.3gpp.org/DynaReport/26114.htm>.
<t>Rewrote the section on modifying SDP to require Suggested:
implementations to clearly indicate whether any given [TS26.114]
modification is allowed.</t> 3GPP, "3rd Generation Partnership Project; Technical
Specification Group Services and System Aspects; IP
Multimedia Subsystem (IMS); Multimedia Telephony; Media
handling and interaction (Release 16)", 3GPP TS 26.114
V16.3.0, September 2019,
<http://www.3gpp.org/DynaReport/26114.htm>. -->
<t>Clarified impact of IceRestart on CreateOffer in local-offer </references>
state.</t> </references>
<section anchor="sec.appendix-a" numbered="true" toc="default">
<name>ABNF Definitions</name>
<t>For the syntax validation performed in
<xref target="sec.parsing-a-desc" format="default"/>, the following list o
f ABNF
definitions is used:</t>
<table anchor="sdp-abnf" align="center">
<name>SDP ABNF References</name>
<thead>
<tr>
<th align="left">Attribute</th>
<th align="left">Reference</th>
</tr>
</thead>
<tbody>
<tr>
<td align="left">ptime</td>
<td align="left">
<xref target="RFC4566" sectionFormat="of" section="9"/></td>
</tr>
<tr>
<td align="left">maxptime</td>
<td align="left">
<xref target="RFC4566" sectionFormat="of" section="9"/></td>
</tr>
<tr>
<td align="left">rtpmap</td>
<td align="left">
<xref target="RFC4566" sectionFormat="of" section="9"/></td>
</tr>
<tr>
<td align="left">recvonly</td>
<td align="left">
<xref target="RFC4566" sectionFormat="of" section="9"/></td>
</tr>
<tr>
<td align="left">sendrecv</td>
<td align="left">
<xref target="RFC4566" sectionFormat="of" section="9"/></td>
</tr>
<tr>
<td align="left">sendonly</td>
<td align="left">
<xref target="RFC4566" sectionFormat="of" section="9"/></td>
</tr>
<tr>
<td align="left">inactive</td>
<td align="left">
<xref target="RFC4566" sectionFormat="of" section="9"/></td>
</tr>
<tr>
<td align="left">framerate</td>
<td align="left">
<xref target="RFC4566" sectionFormat="of" section="9"/></td>
</tr>
<tr>
<td align="left">fmtp</td>
<td align="left">
<xref target="RFC4566" sectionFormat="of" section="9"/></td>
</tr>
<tr>
<td align="left">quality</td>
<td align="left">
<xref target="RFC4566" sectionFormat="of" section="9"/></td>
</tr>
<tr>
<td align="left">rtcp</td>
<td align="left">
<xref target="RFC3605" sectionFormat="of" section="2.1"/></td>
</tr>
<tr>
<td align="left">setup</td>
<td align="left">
Sections <xref target="RFC4145" section="3"
sectionFormat="bare"/>,
<xref target="RFC4145" section="4" sectionFormat="bare"/>, and
<xref target="RFC4145" section="5" sectionFormat="bare"/> of
<xref target="RFC4145"/></td>
</tr>
<tr>
<td align="left">connection</td>
<td align="left">
Sections <xref target="RFC4145" section="3"
sectionFormat="bare"/>,
<xref target="RFC4145" section="4" sectionFormat="bare"/>, and
<xref target="RFC4145" section="5" sectionFormat="bare"/> of
<xref target="RFC4145"/></td>
</tr>
<tr>
<td align="left">fingerprint</td>
<td align="left">
<xref target="RFC8122" sectionFormat="of" section="5"/></td>
</tr>
<tr>
<td align="left">rtcp-fb</td>
<td align="left">
<xref target="RFC4585" sectionFormat="of" section="4.2"/></td>
</tr>
<tr>
<td align="left">extmap</td>
<td align="left">
<xref target="RFC5285" sectionFormat="of" section="7"/></td>
</tr>
<tr>
<td align="left">mid</td>
<td align="left">
Sections <xref target="RFC5888" section="4"
sectionFormat="bare"/> and
<xref target="RFC5888" section="5" sectionFormat="bare"/> of
<xref target="RFC5888"/></td>
</tr>
<tr>
<td align="left">group</td>
<td align="left">
Sections <xref target="RFC5888" section="4"
sectionFormat="bare"/> and
<xref target="RFC5888" section="5" sectionFormat="bare"/> of
<xref target="RFC5888"/></td>
</tr>
<tr>
<td align="left">imageattr</td>
<td align="left">
<xref target="RFC6236" sectionFormat="of" section="3.1"/></td>
</tr>
<tr>
<td align="left">extmap (encrypt option)</td>
<td align="left">
<xref target="RFC6904" sectionFormat="of" section="4"/></td>
</tr>
<tr>
<td align="left">candidate</td>
<td align="left">
<xref target="RFC8839" sectionFormat="of" section="5.1"/></td>
</tr>
<tr>
<td align="left">remote-candidates</td>
<td align="left">
<xref target="RFC8839" sectionFormat="of" section="5.2"/></td>
</tr>
<tr>
<td align="left">ice-lite</td>
<td align="left">
<xref target="RFC8839" sectionFormat="of" section="5.3"/></td>
</tr>
<tr>
<td align="left">ice-ufrag</td>
<td align="left">
<xref target="RFC8839" sectionFormat="of" section="5.4"/></td>
</tr>
<tr>
<td align="left">ice-pwd</td>
<td align="left">
<xref target="RFC8839" sectionFormat="of" section="5.4"/></td>
</tr>
<tr>
<td align="left">ice-options</td>
<td align="left">
<xref target="RFC8839" sectionFormat="of" section="5.6"/></td>
</tr>
<tr>
<td align="left">msid</td>
<td align="left">
<xref target="RFC8830" sectionFormat="of" section="3"/></td>
</tr>
<tr>
<td align="left">rid</td>
<td align="left">
<xref target="RFC8851" sectionFormat="of" section="10"/></td>
</tr>
<tr>
<td align="left">simulcast</td>
<td align="left">
<xref target="RFC8853" sectionFormat="of" section="6.1"/></td>
</tr>
<tr>
<td align="left">tls-id</td>
<td align="left">
<xref target="RFC8842" sectionFormat="of" section="4"/></td>
</tr>
</tbody>
</table>
<t>Guidance on whether attributes should be defined at the <!-- [rfced] Appendix A: Please review the following, and let us
media level or the session level.</t> know any concerns.
<t>Renamed "default" bundle policy to "balanced".</t> a) We changed the title of Appendix A from "Appendix A" to
"ABNF Definitions." If this is not correct, please provide an
appropriately descriptive title.
<t>Removed default ICE candidate pool size and clarify how it b) Please review the citations listed in Table 1. In many cases,
works.</t> we could not see the relevant parameters listed in the cited
sections. For example, we do not see
<t>Defined a canonical order for assignment of MSTs to m= * any of the attributes (ptime, maxptime, rtpmap, recvonly, ...)
lines.</t> listed in Section 9 of [RFC4566].
* the setup attribute mentioned in Section 3 of [RFC4145].
* the connection attribute mentioned in Sections 3 or 4 of [RFC4145].
* the "mid" attribute listed in Section 5 of [RFC5888].
* the "group" attribute listed in Section 4 of [RFC5888].
<t>Removed discussion of rehydration.</t> c) We do not see the attributes "framerate," "quality," or
"connection" listed anywhere else in this document. Do they need to
be included here?
<t>Added Eric Rescorla as a draft editor.</t> d) Please note that in order to render workable hyperlinks in the
.html/.pdf files we changed the original citation format (for example,
"[RFC5285] Section 7") to this style (for example, "Section 7 of
[RFC5285]").
<t>Cleaned up references.</t> e) As noted previously, it appears that most of the listed section
numbers for RFC 8839 [I-D.ietf-mmusic-ice-sip-sdp] are "off by one" (i.e.,
that "4.1" should be "5.1," "4.2" should be "5.2," etc. We have updated the
section references accordingly; please review and let us know if any
corrections are needed.
<t>Editorial cleanup</t> f) Please confirm that Section 6.1 of RFC 8853 [I-D.ietf-mmusic-sdp-simulcast]
</list></t> is the correct section to cite here.
(We ask because we could not see a relationship, other than the
mention of "simulcast" mostly in terms of simulcast streams, and also
because we see ABNF for the "a=simulcast" attribute in Section 5.1 of
RFC 8853 [I-D.ietf-mmusic-sdp-simulcast].)
<t>Changes in draft-06: Original:
<list style="symbols"> Appendix A. Appendix A
...
| simulcast | [I-D.ietf-mmusic-sdp-simulcast] Section |
| | 6.1
<t>Reworked handling of m= line recycling.</t> Currently:
Appendix A. ABNF Definitions
...
| simulcast | Section 6.1 of [RFC8853] | -->
<t>Added handling of BUNDLE and bundle-only.</t> </section>
<section anchor="sec.acknowledgements" numbered="false" toc="default">
<name>Acknowledgements</name>
<t><contact fullname="Harald Alvestrand"/>, <contact fullname="Taylor
Brandstetter"/>, <contact fullname="Suhas Nandakumar"/>, and
<contact fullname="Peter Thatcher"/> provided significant text for this
document. <contact fullname="Bernard Aboba"/>, <contact fullname="Adam
Bergkvist"/>, <contact fullname="Dan Burnett"/>, <contact fullname="Ben
Campbell"/>, <contact fullname="Alissa Cooper"/>,
<contact fullname="Richard Ejzak"/>, <contact fullname="Stefan
Håkansson"/>, <contact fullname="Ted Hardie"/>, <contact fullname="Christe
r Holmberg"/>,
<contact fullname="Andrew Hutton"/>, <contact fullname="Randell
Jesup"/>, <contact fullname="Matthew Kaufman"/>, <contact fullname="Anant
Narayanan"/>,
<contact fullname="Adam Roach"/>, <contact fullname="Robert Sparks"/>,
<contact fullname="Neil Stratford"/>, <contact fullname="Martin
Thomson"/>, <contact fullname="Sean
Turner"/>, and <contact fullname="Magnus Westerlund"/> all provided valuab
le feedback on
this document.</t>
</section>
</back>
<t>Clarified handling of rollback.</t> <!-- [rfced] Please let us know if any changes are needed for the
following:
<t>Added text describing the ICE Candidate Pool and its a) The following terms were used inconsistently in this document.
behavior.</t> We chose to use the latter forms. Please let us know any objections.
<t>Allowed OfferToReceiveX to create multiple recvonly m= a m= / an "m=" (per published RFCs and per author feedback for other
sections.</t> documents in this cluster
</list></t>
<t>Changes in draft-05: q= value / "q=" value (We also changed 'x= and y= values' to
<list style="symbols"> "x=" and "y=" values')
<t>Fixed several issues identified in the createOffer/Answer a RTP / an RTP
sections during document review.</t>
<t>Updated references.</t> to true / to "true" (per the rest of the cluster)
</list></t>
<t>Changes in draft-04: to false / to "false" (per the rest of the cluster)
<list style="symbols">
<t>Filled in sections on createOffer and createAnswer.</t> bundle-tag / BUNDLE-tag (per other documents in this cluster)
<t>Added SDP examples.</t> offer-answer exchange / offer/answer exchange
<t>Fixed references.</t> clockrate / clock rate (per RFCs 3389 and 7160)
</list></t>
<t>Changes in draft-03: other than IDENTICAL and TRANSPORT / other than IDENTICAL or TRANSPORT
<list style="symbols">
<t>Added text describing relationship to W3C specification</t> b) The following terms appear to be used inconsistently in this
</list></t> document. Please let us know which form is preferred.
<t>Changes in draft-02: createOffer API / createOffer() API (We see "createAnswer() API" but
<list style="symbols"> not "createAnswer API"; however, we also see "getCapabilities API"
<!-- A --> and "W3C RTCPeerConnection API." In RFC 8826 (draft-ietf-rtcweb-security),
we see "XMLHttpRequest() API.")
<t>Converted from nroff</t> setLocalDescription API / setLocalDescription() API
<!-- B --> / setRemoteDescription() API
(We also see "whether setLocalDescription or
setRemoteDescription is called" and "before
setRemoteDescription() is called")
<t>Removed comparisons to old approaches abandoned by the mid identifiers / MID identifiers
working group</t> Because RFC 8843 (draft-ietf-mmusic-sdp-bundle-negotiation) and
<!-- C --> RFC 8852 (draft-ietf-avtext-rid) define "MID" as "Media Identification,"
would it be appropriate to use "mid values" (or "MID values"*)
and "MIDs" instead of "mid identifiers" and "MID identifiers"?
<t>Removed stuff that has moved to W3C specification</t> * We see inconsistent capitalization in this cluster of documents;
<!-- D --> a separate question regarding which form to use will be sent
separately.
<t>Align SDP handling with W3C draft</t> "RID identifiers" and "rid-identifier": RFC 8851 (draft-ietf-mmusic-rid)
<!-- E --> defines "RID" as "restriction identifier." Should "RID
identifiers" and "rid-identifier" be "rid-ids" and "rid-id"
per RFC 8851 (draft-ietf-mmusic-rid), to avoid "identifier identifier(s)"?
<t>Clarified section on forking.</t> the stable state (1 instance) / the "stable" state (3 instances)
<!-- F --> (Also, should 'the previous stable state' be 'a previous stable
<!-- G --> state' (per Section 5.7) or 'the previous "stable" state'?)
<!-- H -->
<!-- I -->
<!-- J -->
<!-- K -->
<!-- L -->
</list></t>
<t>Changes in draft-01: sess-id / <sess-id> (Section 5.2.1, second bullet item)
<list style="symbols">
<t>Added diagrams for architecture and state machine.</t> set to zero / set to 0 (May we use the latter, because of
'"dummy" port value of 9' in Sections 5.2.1 and 5.3.1?)
<t>Added sections on forking and rehydration.</t> dummy value / "dummy" value
<t>Clarified meaning of "pranswer" and "answer".</t> fmt value / "fmt" value
<t>Reworked how ICE restarts and media directions are mid value / "mid" value
controlled.</t>
<t>Added list of parameters that can be changed in a tls-id value / "a=tls-id" value
description.</t>
<t>Updated suggested API and examples to match latest "IceRestart" option / IceRestart option
thinking.</t>
<t>Suggested API and examples have been moved to an dissociate (Section 5.10) / disassociate (Sections 3.4.1 and 5.7)
appendix.</t> Example: "A rollback disassociates any RtpTransceivers that were
</list></t> associated with m= sections by the application of the rolled-back
session description (see Section 5.10 and Section 5.9)."
<t>Changes in draft -00: c) Are "canTrickleIceCandidates" (Section 4.1.15) and "canTrickle
<list style="symbols"> property" (Section 5.10) distinct terms? If so, should this
distinction be clarified in the text? -->
<t>Migrated from draft-uberti-rtcweb-jsep-02.</t>
</list></t>
</section>
</back>
</rfc> </rfc>
 End of changes. 910 change blocks. 
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