Please check the errata for any errors or issues reported since publication.
See also translations.
Initial Author of this Specification was Ian Hickson, Google Inc., with
the following copyright statement:
© Copyright 2004-2011 Apple Computer, Inc., Mozilla Foundation, and Opera
Software ASA. You are granted a license to use, reproduce and create
derivative works of this document. All subsequent changes since 26 July
2011 done by the W3C WebRTC Working Group are under the following
Copyright:
© 2011-2021 W3C® (MIT, ERCIM,
Keio, Beihang). W3C liability,
trademark and permissive document license rules
apply.
This document defines a set of ECMAScript APIs in WebIDL to allow media and generic application data to be sent to and received from another browser or device implementing the appropriate set of real-time protocols. This specification is being developed in conjunction with a protocol specification developed by the IETF RTCWEB group and an API specification to get access to local media devices.
This section describes the status of this document at the time of its publication. Other documents may supersede this document. A list of current W3C publications and the latest revision of this technical report can be found in the W3C technical reports index at https://www.w3.org/TR/.
The API is based on preliminary work done in the WHATWG.
Its associated test suite is used to build an implementation report of the API.
This document was published by the Web Real-Time Communications Working Group as a Recommendation.
GitHub Issues are preferred for discussion of this specification. Alternatively, you can send comments to our mailing list. Please send them to public-webrtc@w3.org (archives).
A W3C Recommendation is a specification that, after extensive consensus-building, has received the endorsement of the W3C and its Members. W3C recommends the wide deployment of this specification as a standard for the Web. Future updates to this Recommendation may incorporate new features.
This document was produced by a group operating under the W3C Patent Policy. W3C maintains a public list of any patent disclosures made in connection with the deliverables of the group; that page also includes instructions for disclosing a patent. An individual who has actual knowledge of a patent which the individual believes contains Essential Claim(s) must disclose the information in accordance with section 6 of the W3C Patent Policy.
This document is governed by the 15 September 2020 W3C Process Document.
This section is non-normative.
There are a number of facets to peer-to-peer communications and video-conferencing in HTML covered by this specification:
This document defines the APIs used for these features. This specification is being developed in conjunction with a protocol specification developed by the IETF RTCWEB group and an API specification to get access to local media devices [GETUSERMEDIA] developed by the WebRTC Working Group. An overview of the system can be found in [RFC8825] and [RFC8826].
As well as sections marked as non-normative, all authoring guidelines, diagrams, examples, and notes in this specification are non-normative. Everything else in this specification is normative.
The key words MAY, MUST, MUST NOT, and SHOULD in this document are to be interpreted as described in BCP 14 [RFC2119] [RFC8174] when, and only when, they appear in all capitals, as shown here.
This specification defines conformance criteria that apply to a single product: the user agent that implements the interfaces that it contains.
Conformance requirements phrased as algorithms or specific steps may be implemented in any manner, so long as the end result is equivalent. (In particular, the algorithms defined in this specification are intended to be easy to follow, and not intended to be performant.)
Implementations that use ECMAScript to implement the APIs defined in this specification MUST implement them in a manner consistent with the ECMAScript Bindings defined in the Web IDL specification [WEBIDL], as this specification uses that specification and terminology.
The EventHandler
interface, representing a callback used for event
handlers, is defined in [HTML].
The concepts queue a task and networking task source are defined in [HTML].
The concept fire an event is defined in [DOM].
The terms event, event handlers and event handler event types are defined in [HTML].
Performance
.timeOrigin
and Performance
.now
()
are defined in
[hr-time].
The terms serializable objects, serialization steps, and deserialization steps are defined in [HTML].
The terms MediaStream
, MediaStreamTrack
, and
MediaStreamConstraints
are defined in [GETUSERMEDIA]. Note that
MediaStream
is extended in § 9.2
MediaStream
in this document while MediaStreamTrack
is extended in § 9.3
MediaStreamTrack
in this document.
The term Blob
is defined in [FILEAPI].
The term media description is defined in [RFC4566].
The term media transport is defined in [RFC7656].
The term generation is defined in [RFC8838] Section 2.
The terms stats object and monitored object are defined in [WEBRTC-STATS].
When referring to exceptions, the terms throw and created are defined in [WEBIDL].
The callback VoidFunction
is defined in [WEBIDL].
The term "throw" is used as specified in [INFRA]: it terminates the current processing steps.
The terms fulfilled, rejected, resolved, pending and settled used in the context of Promises are defined in [ECMASCRIPT-6.0].
The terms bundle, bundle-only and bundle-policy are defined in [RFC8829].
The AlgorithmIdentifier is defined in [WebCryptoAPI].
The general principles for Javascript APIs apply, including the
principle of run-to-completion
and no-data-races as defined in [API-DESIGN-PRINCIPLES]. That is,
while a task is running, external events do not influence what's
visible to the Javascript application. For example, the amount of data
buffered on a data channel will increase due to "send" calls while
Javascript is executing, and the decrease due to packets being sent
will be visible after a task checkpoint.
It is the responsibility of the user agent to make sure the set of
values presented to the application is consistent - for instance that
getContributingSources() (which is synchronous) returns values for all
sources measured at the same time.
An
instance allows an application to establish
peer-to-peer communications with another RTCPeerConnection
instance in another browser, or to another endpoint implementing the
required protocols. Communications are coordinated by the exchange of
control messages (called a signaling protocol) over a signaling
channel which is provided by unspecified means, but generally by a
script in the page via the server, e.g. using Web
Sockets or RTCPeerConnection
XMLHttpRequest
[xhr].
RTCConfiguration
Dictionary
The
defines a set of parameters to configure
how the peer-to-peer communication established via
RTCConfiguration
is established or re-established.
RTCPeerConnection
WebIDLdictionaryRTCConfiguration
{ sequence<RTCIceServer
>iceServers
;RTCIceTransportPolicy
iceTransportPolicy
;RTCBundlePolicy
bundlePolicy
;RTCRtcpMuxPolicy
rtcpMuxPolicy
; sequence<RTCCertificate
>certificates
; [EnforceRange] octeticeCandidatePoolSize
= 0; };
RTCConfiguration
Members
iceServers
of type sequence<RTCIceServer
>
An array of objects describing servers available to be used by ICE, such as STUN and TURN servers.
iceTransportPolicy
of type
RTCIceTransportPolicy
.
Indicates which candidates the ICE Agent is allowed to use.
bundlePolicy
of type RTCBundlePolicy
.
Indicates which media-bundling policy to use when gathering ICE candidates.
rtcpMuxPolicy
of type RTCRtcpMuxPolicy
.
Indicates which rtcp-mux policy to use when gathering ICE candidates.
certificates
of type sequence<RTCCertificate
>
A set of certificates that the
uses
to authenticate.
RTCPeerConnection
Valid values for this parameter are created through calls
to the generateCertificate
()
function.
Although any given DTLS connection will use only one
certificate, this attribute allows the caller to provide
multiple certificates that support different algorithms.
The final certificate will be selected based on the DTLS
handshake, which establishes which certificates are
allowed. The
implementation selects
which of the certificates is used for a given connection;
how certificates are selected is outside the scope of this
specification.
RTCPeerConnection
Existing implementations only utilize the first certificate provided; the others are ignored.
If this value is absent, then a default set of certificates
is generated for each
instance.
RTCPeerConnection
This option allows applications to establish key
continuity. An
can be persisted in
[INDEXEDDB] and reused. Persistence and reuse also
avoids the cost of key generation.
RTCCertificate
The value for this configuration option cannot change after its value is initially selected.
iceCandidatePoolSize
of type
octet, defaulting to
0
Size of the prefetched ICE pool as defined in [RFC8829] (section 3.5.4. and section 4.1.1.).
RTCIceCredentialType
Enum
WebIDLenumRTCIceCredentialType
{ "password
" };
Enumeration description | |
---|---|
password
|
The credential is a long-term authentication username and password, as described in [RFC5389], Section 10.2. |
RTCIceServer
Dictionary
The
dictionary is used to describe the STUN and
TURN servers that can be used by the ICE Agent to establish a
connection with a peer.
RTCIceServer
WebIDLdictionaryRTCIceServer
{ required (DOMString or sequence<DOMString>)urls
; DOMStringusername
; DOMStringcredential
;RTCIceCredentialType
credentialType
= "password"; };
RTCIceServer
Members
urls
of type (DOMString or
sequence<DOMString>), required
STUN or TURN URI(s) as defined in [RFC7064] and [RFC7065] or other URI types.
username
of type DOMString
If this
object represents a TURN server,
and RTCIceServer
is
"credentialType
", then this attribute
specifies the username to use with that TURN server.
password
credential
of type DOMString
If this
object represents a TURN server,
then this attribute specifies the credential to use with
that TURN server.
RTCIceServer
If
is
"credentialType
", password
represents a long-term authentication password, as
described in [RFC5389], Section 10.2.
credential
To support additional values of
,
credentialType
may evolve in future as a union.
credential
credentialType
of type RTCIceCredentialType
, defaulting
to "password
"
If this
object represents a TURN server,
then this attribute specifies how credential
should be used when that TURN server requests
authorization.
RTCIceServer
An example array of
objects is:
RTCIceServer
[
{urls: 'stun:stun1.example.net'},
{urls: ['turns:turn.example.org', 'turn:turn.example.net'],
username: 'user',
credential: 'myPassword',
credentialType: 'password'},
];
RTCIceTransportPolicy
Enum
As described in [RFC8829] (section 4.1.1.), if
the
member of the
iceTransportPolicy
is specified, it defines the ICE candidate policy [RFC8829] (section 3.5.3.) the
browser uses to surface the permitted candidates to the
application; only these candidates will be used for connectivity
checks.
RTCConfiguration
WebIDLenumRTCIceTransportPolicy
{ "relay
", "all
" };
Enumeration description (non-normative) | |
---|---|
relay
|
The ICE Agent uses only media relay candidates such as candidates passing through a TURN server. Note
This can be used to prevent the remote endpoint from
learning the user's IP addresses, which may be desired in
certain use cases. For example, in a "call"-based
application, the application may want to prevent an
unknown caller from learning the callee's IP addresses
until the callee has consented in some way.
|
all
|
The ICE Agent can use any type of candidate when this value is specified. Note
The implementation can still use its own candidate
filtering policy in order to limit the IP addresses
exposed to the application, as noted in the description
of . .
|
RTCBundlePolicy
Enum
As described in [RFC8829] (section 4.1.1.), bundle policy affects which media tracks are negotiated if the remote endpoint is not bundle-aware, and what ICE candidates are gathered. If the remote endpoint is bundle-aware, all media tracks and data channels are bundled onto the same transport.
WebIDLenumRTCBundlePolicy
{ "balanced
", "max-compat
", "max-bundle
" };
Enumeration description (non-normative) | |
---|---|
balanced
|
Gather ICE candidates for each media type in use (audio, video, and data). If the remote endpoint is not bundle-aware, negotiate only one audio and video track on separate transports. |
max-compat
|
Gather ICE candidates for each track. If the remote endpoint is not bundle-aware, negotiate all media tracks on separate transports. |
max-bundle
|
Gather ICE candidates for only one track. If the remote endpoint is not bundle-aware, negotiate only one media track. |
RTCRtcpMuxPolicy
Enum
As described in [RFC8829] (section 4.1.1.), the
affects what ICE candidates are gathered to
support non-multiplexed RTCP. The only value defined in this spec
is "RTCRtcpMuxPolicy
".
require
WebIDL enumRTCRtcpMuxPolicy
{ "require
" };
Enumeration description (non-normative) | |
---|---|
require
|
Gather ICE candidates only for RTP and multiplex RTCP on the RTP candidates. If the remote endpoint is not capable of rtcp-mux, session negotiation will fail. |
These dictionaries describe the options that can be used to control the offer/answer creation process.
WebIDLdictionary RTCOfferAnswerOptions
{};
RTCOfferAnswerOptions
Members
WebIDL dictionaryRTCOfferOptions
:RTCOfferAnswerOptions
{ booleaniceRestart
= false; };
RTCOfferOptions
Members
iceRestart
of type boolean, defaulting to
false
When the value of this dictionary member is
true
, or the relevant
object's [[LocalIceCredentialsToReplace]] slot is
not empty, then the generated description will have ICE
credentials that are different from the current credentials
(as visible in the
RTCPeerConnection
attribute's
SDP). Applying the generated description will restart ICE,
as described in section 9.1.1.1 of [RFC5245].
currentLocalDescription
When the value of this dictionary member is
false
, and the relevant
object's [[LocalIceCredentialsToReplace]] slot is
empty, and the
RTCPeerConnection
attribute has
valid ICE credentials, then the generated description will
have the same ICE credentials as the current value from the
currentLocalDescription
attribute.
currentLocalDescription
Performing an ICE restart is recommended when
transitions to
"iceConnectionState
". An application may
additionally choose to listen for the
failed
transition to
"iceConnectionState
" and then use other
sources of information (such as using
disconnected
to measure if the number of
bytes sent or received over the next couple of seconds
increases) to determine whether an ICE restart is
advisable.
getStats
The RTCAnswerOptions
dictionary describe options
specific to session description of type "
"
(none in this version of the specification).
answer
WebIDLdictionaryRTCAnswerOptions
:RTCOfferAnswerOptions
{};
RTCSignalingState
Enum
WebIDLenumRTCSignalingState
{ "stable
", "have-local-offer
", "have-remote-offer
", "have-local-pranswer
", "have-remote-pranswer
", "closed
" };
Enumeration description | |
---|---|
stable
|
There is no offer/answer exchange in progress. This is also the initial state, in which case the local and remote descriptions are empty. |
have-local-offer
|
A local description, of type " ", has
been successfully applied.
|
have-remote-offer
|
A remote description, of type " ", has
been successfully applied.
|
have-local-pranswer
|
A remote description of type " " has
been successfully applied and a local description of type
" " has been successfully applied.
|
have-remote-pranswer
|
A local description of type " " has been
successfully applied and a remote description of type
" " has been successfully applied.
|
closed
|
The has been closed; its
[[IsClosed]] slot is true .
|
An example set of transitions might be:
stable
"
have-local-offer
"
have-remote-pranswer
"
stable
"
stable
"
have-remote-offer
"
have-local-pranswer
"
stable
"
RTCIceGatheringState
Enum
WebIDLenumRTCIceGatheringState
{ "new
", "gathering
", "complete
" };
Enumeration description | |
---|---|
new
|
Any of the s are in the
" " gathering state and none of
the transports are in the
" " state, or there are no
transports.
|
gathering
|
Any of the s are in the
" " state.
|
complete
|
At least one exists, and all
s are in the
" " gathering state.
|
The set of transports considered is the set of transports presently referenced by the PeerConnection's set of transceivers.
RTCPeerConnectionState
Enum
WebIDLenumRTCPeerConnectionState
{ "closed
", "failed
", "disconnected
", "new
", "connecting
", "connected
" };
Enumeration description | |
---|---|
closed
|
The object's [[IsClosed]]
slot is true .
|
failed
|
The previous state doesn't apply and any
s are in the
" " state or any
s are in the
" " state.
|
disconnected
|
None of the previous states apply and any
s are in the
" " state.
|
new
|
None of the previous states apply and all
s are in the
" " or
" " state, and all
s are in the
" " or
" " state, or there are no
transports.
|
connecting
|
None of the previous states apply and any
is in the
" " state or any
is in the
" " state.
|
connected
|
None of the previous states apply and all
s are in the
" ",
" " or
" " state, and all
s are in the
" " or
" " state.
|
The set of transports considered is the set of transports presently referenced by the PeerConnection's set of transceivers.
RTCIceConnectionState
Enum
WebIDLenumRTCIceConnectionState
{ "closed
", "failed
", "disconnected
", "new
", "checking
", "completed
", "connected
" };
Enumeration description | |
---|---|
closed
|
The object's [[IsClosed]]
slot is true .
|
failed
|
The previous state doesn't apply and any
s are in the
" " state.
|
disconnected
|
None of the previous states apply and any
s are in the
" " state.
|
new
|
None of the previous states apply and all
s are in the
" " or
" " state, or there are no
transports.
|
checking
|
None of the previous states apply and any
s are in the
" " or
" " state.
|
completed
|
None of the previous states apply and all
s are in the
" " or
" " state.
|
connected
|
None of the previous states apply and all
s are in the
" ",
" " or
" " state.
|
The set of transports considered is the set of transports presently referenced by the PeerConnection's set of transceivers.
Note that if an
is discarded as a result of
signaling (e.g. RTCP mux or bundling), or created as a result of
signaling (e.g. adding a new media description), the state
may advance directly from one state to another.
RTCIceTransport
The [RFC8829] specification, as a whole, describes the details of how
the
operates. References to specific
subsections of [RFC8829] are provided as appropriate.
RTCPeerConnection
Calling new
creates an
(configuration)RTCPeerConnection
object.
RTCPeerConnection
configuration.
contains
information used to find and access the servers used by ICE. The
application can supply multiple servers of each type, and any TURN
server MAY also be used as a STUN server for the purposes of
gathering server reflexive candidates.
iceServers
An
object has a signaling state, a
connection state, an ICE gathering state, and
an ICE connection state. These are initialized when the
object is created.
RTCPeerConnection
The ICE protocol implementation of an
is
represented by an ICE agent [RFC5245]. Certain
RTCPeerConnection
methods involve interactions with the ICE
Agent, namely RTCPeerConnection
, addIceCandidate
,
setConfiguration
, setLocalDescription
and setRemoteDescription
.
These interactions are described in the relevant sections in this
document and in [RFC8829]. The ICE Agent also provides
indications to the user agent when the state of its internal
representation of an close
changes, as described in
§ 5.6
RTCIceTransport
RTCIceTransport
Interface
.
The task source for the tasks listed in this section is the networking task source.
The state of the SDP negotiation is represented by the signaling
state and the internal variables
[[CurrentLocalDescription]],
[[CurrentRemoteDescription]],
[[PendingLocalDescription]] and
[[PendingRemoteDescription]]. These are only set inside the
and setLocalDescription
operations,
and modified by the setRemoteDescription
operation and the surface a candidate procedure. In each case, all the
modifications to all the five variables are completed before the
procedures fire any events or invoke any callbacks, so the
modifications are made visible at a single point in time.
addIceCandidate
As one of the unloading document cleanup steps, run the following steps:
Let window be document's relevant global object.
For each
object connection
whose relevant
global object is window, close the connection with connection and the value RTCPeerConnection
true
.
When the RTCPeerConnection.constructor()
is
invoked, the user agent MUST run the following steps:
If any of the steps enumerated below fails for a reason not
specified here, throw an UnknownError
with the
attribute set to an
appropriate description.
message
Let connection be a newly created
object.
RTCPeerConnection
Let connection have a [[DocumentOrigin]] internal slot, initialized to the relevant settings object's origin.
If the
value in
configuration is non-empty, run the following
steps for each certificate in certificates:
certificates
If the value of
certificate.
is less
than the current time, throw an
expires
InvalidAccessError
.
If certificate.[[Origin]] is not
same origin with
connection.[[DocumentOrigin]], throw an InvalidAccessError
.
Store certificate.
Else, generate one or more new
instances
with this RTCCertificate
instance and store them. This
MAY happen asynchronously and the value of
RTCPeerConnection
remains
certificates
undefined
for the subsequent steps. As noted in
Section 4.3.2.3 of [RFC8826], WebRTC utilizes
self-signed rather than Public Key Infrastructure (PKI)
certificates, so that the expiration check is to ensure that
keys are not used indefinitely and additional certificate
checks are unnecessary.
Initialize connection's ICE Agent.
If the value of
configuration.
is iceTransportPolicy
undefined
, set it to
"
".
all
If the value of
configuration.
is
bundlePolicy
undefined
, set it to
"
".
balanced
If the value of
configuration.
is rtcpMuxPolicy
undefined
, set it to
"
".
require
Let connection have a [[Configuration]] internal slot. Set the configuration specified by configuration.
Let connection have an [[IsClosed]]
internal slot, initialized to false
.
Let connection have a
[[NegotiationNeeded]] internal slot, initialized
to false
.
Let connection have an
[[SctpTransport]] internal slot, initialized to
null
.
Let connection have an [[Operations]] internal slot, representing an operations chain, initialized to an empty list.
Let connection have a
[[UpdateNegotiationNeededFlagOnEmptyChain]]
internal slot, initialized to false
.
Let connection have an
[[LastCreatedOffer]] internal slot, initialized
to ""
.
Let connection have an
[[LastCreatedAnswer]] internal slot, initialized
to ""
.
Let connection have an [[EarlyCandidates]] internal slot, initialized to an empty list.
Set connection's signaling state to
"
".
stable
Set connection's ICE connection state to
"
".
new
Set connection's ICE gathering state to
"
".
new
Set connection's connection state to
"
".
new
Let connection have a
[[PendingLocalDescription]] internal slot,
initialized to null
.
Let connection have a
[[CurrentLocalDescription]] internal slot,
initialized to null
.
Let connection have a
[[PendingRemoteDescription]] internal slot,
initialized to null
.
Let connection have a
[[CurrentRemoteDescription]] internal slot,
initialized to null
.
Let connection have a [[LocalIceCredentialsToReplace]] internal slot, initialized to an empty set.
Return connection.
An
object has an operations
chain, [[Operations]], which ensures that only one
asynchronous operation in the chain executes concurrently. If
subsequent calls are made while the returned promise of a
previous call is still not settled, they are added to the
chain and executed when all the previous calls have finished
executing and their promises have settled.
RTCPeerConnection
To chain an operation to an
object's operations chain, run the
following steps:
RTCPeerConnection
Let connection be the
object.
RTCPeerConnection
If connection.[[IsClosed]] is
true
, return a promise rejected with a
newly created InvalidStateError
.
Let operation be the operation to be chained.
Let p be a new promise.
Append operation to [[Operations]].
If the length of [[Operations]] is exactly 1, execute operation.
Upon fulfillment or rejection of the promise returned by the operation, run the following steps:
If connection.[[IsClosed]] is
true
, abort these steps.
If the promise returned by operation was fulfilled with a value, fulfill p with that value.
If the promise returned by operation was rejected with a value, reject p with that value.
Upon fulfillment or rejection of p, execute the following steps:
If connection.[[IsClosed]] is
true
, abort these steps.
Remove the first element of [[Operations]].
If [[Operations]] is non-empty, execute the operation represented by the first element of [[Operations]], and abort these steps.
If
connection.[[UpdateNegotiationNeededFlagOnEmptyChain]]
is false
, abort these steps.
Set
connection.[[UpdateNegotiationNeededFlagOnEmptyChain]]
to false
.
Update the negotiation-needed flag for connection.
Return p.
An
object has an aggregated connection
state. Whenever the state of an RTCPeerConnection
changes
or when the [[IsClosed]] slot turns RTCDtlsTransport
true
,
the user agent MUST update the connection state by queueing a
task that runs the following steps:
Let connection be this
object.
RTCPeerConnection
Let newState be the value of deriving a new state
value as described by the
enum.
RTCPeerConnectionState
If connection's connection state is equal to newState, abort these steps.
Let connection's connection state be newState.
Fire an event named
at
connection.
connectionstatechange
To update the ICE
gathering state of an
instance
connection, the user agent MUST queue a task that runs
the following steps:
RTCPeerConnection
If connection.[[IsClosed]] is
true
, abort these steps.
Let newState be the value of deriving a new state
value as described by the
enum.
RTCIceGatheringState
If connection's ICE gathering state is equal to newState, abort these steps.
Set connection's ICE gathering state to newState.
Fire an event named
at
connection.
icegatheringstatechange
If newState is
"
", fire an event
named complete
using the
icecandidate
interface with the candidate
attribute set to RTCPeerConnectionIceEvent
null
at connection.
RTCIceTransport
and/or RTCPeerConnection
.
To
set a local session description description on
an
object connection, set the session description
description on connection with the additional
value RTCPeerConnection
false
.
To
set a remote session description description
on an
object connection, set the session description
description on connection with the additional
value RTCPeerConnection
true
.
To set
a session description description on an
object connection, given a
remote boolean, run the following steps:
RTCPeerConnection
Let p be a new promise.
If description.
is
"type
" and connection's signaling state is either "rollback
",
"stable
", or
"have-local-pranswer
", then reject p with a newly created
have-remote-pranswer
InvalidStateError
and abort these steps.
Let jsepSetOfTransceivers be a shallow copy of connection's set of transceivers.
In parallel, start the process to apply description as described in [RFC8829] (section 5.5. and section 5.6.), with these additional restrictions:
Use jsepSetOfTransceivers as the source of truth with regard to what "RtpTransceivers" exist, and their [[JsepMid]] internal slot as their "mid property".
If remote is true
, validate
back-to-back offers as if answers were applied in
between, by running the check for subsequent offers as if
it were in stable state.
If applying description leads to modifying a transceiver transceiver, and transceiver.[[Sender]].[[SendEncodings]] is non-empty, and not equal to the encodings that would result from processing description, the process of applying description fails. This specification does not allow remotely initiated RID renegotiation.
If the process to apply description fails for any reason, then the user agent MUST queue a task that runs the following steps:
If connection.[[IsClosed]] is
true
, then abort these steps.
If
description.
is invalid for the current signaling state of
connection as described in
[RFC8829] (section 5.5. and section 5.6.), then reject p with
a newly created type
InvalidStateError
and abort these steps.
If the content of description is not valid
SDP syntax, then reject p with an
(with RTCError
set to
"errorDetail
" and the
sdp-syntax-error
attribute set to the line
number in the SDP where the syntax error was
detected) and abort these steps.
sdpLineNumber
If remote is true
, the
connection's
is
RTCRtcpMuxPolicy
and the description does
not use RTCP mux, then reject p with
a newly created
require
InvalidAccessError
and abort these steps.
If the description attempted to renegotiate RIDs, as
described above, then reject p with
a newly created
InvalidAccessError
and abort these steps.
If the content of description is invalid,
then reject p with a newly created InvalidAccessError
and abort
these steps.
For all other errors, reject p with
a newly created OperationError
.
If description is applied successfully, the user agent MUST queue a task that runs the following steps:
If connection.[[IsClosed]] is
true
, then abort these steps.
If remote is true
and
description is of type
"
", then if any
offer
addTrack
()
methods succeeded
during the process to apply description,
abort these steps and start the process over as if
they had succeeded prior, to include the extra
transceiver(s) in the process.
If description is of type
"
" and the signaling state
of connection is
"offer
" then for each
transceiver in connection's set of transceivers, run the following steps:
stable
Set transceiver.[[Sender]].[[LastStableStateSenderTransport]] to transceiver.[[Sender]].[[SenderTransport]].
Set transceiver.[[Receiver]].[[LastStableStateReceiverTransport]] to transceiver.[[Receiver]].[[ReceiverTransport]].
Set transceiver.[[Receiver]].[[LastStableStateAssociatedRemoteMediaStreams]] to transceiver.[[Receiver]].[[AssociatedRemoteMediaStreams]].
Set transceiver.[[Receiver]].[[LastStableStateReceiveCodecs]] to transceiver.[[Receiver]].[[ReceiveCodecs]].
If remote is false
, then run
one of the following steps:
If description is of type
"
", set
connection.[[PendingLocalDescription]]
to a new offer
object
constructed from description, set
connection's signaling state to
"RTCSessionDescription
", and release early candidates.
have-local-offer
If description is of type
"
", then this completes an
offer answer negotiation. Set
connection.[[CurrentLocalDescription]]
to a new answer
object
constructed from description, and set
connection.[[CurrentRemoteDescription]]
to
connection.[[PendingRemoteDescription]].
Set both
connection.[[PendingRemoteDescription]]
and
connection.[[PendingLocalDescription]]
to RTCSessionDescription
null
. Set both
connection.[[LastCreatedOffer]]
and
connection.[[LastCreatedAnswer]]
to ""
, set connection's
signaling state to
"
", and release
early candidates. Finally, if none of the ICE
credentials in
connection.[[LocalIceCredentialsToReplace]]
are present in description, then set
connection.[[LocalIceCredentialsToReplace]]
to an empty set.
stable
If description is of type
"
", then set
connection.[[PendingLocalDescription]]
to a new pranswer
object
constructed from description, set
connection's signaling state to
"RTCSessionDescription
", and
release early candidates.
have-local-pranswer
Otherwise, (if remote is
true
) run one of the following steps:
If description is of type
"
", set
connection.[[PendingRemoteDescription]]
attribute to a new offer
object constructed from description,
and set connection's signaling
state to
"RTCSessionDescription
".
have-remote-offer
If description is of type
"
", then this completes an
offer answer negotiation. Set
connection.[[CurrentRemoteDescription]]
to a new answer
object
constructed from description, and set
connection.[[CurrentLocalDescription]]
to
connection.[[PendingLocalDescription]].
Set both
connection.[[PendingRemoteDescription]]
and
connection.[[PendingLocalDescription]]
to RTCSessionDescription
null
. Set both
connection.[[LastCreatedOffer]]
and
connection.[[LastCreatedAnswer]]
to ""
, and set
connection's signaling state to
"
". Finally, if none
of the ICE credentials in
connection.[[LocalIceCredentialsToReplace]]
are present in the newly set
connection.[[CurrentLocalDescription]],
then set
connection.[[LocalIceCredentialsToReplace]]
to an empty set.
stable
If description is of type
"
", then set
connection.[[PendingRemoteDescription]]
to a new pranswer
object
constructed from description and set
connection's signaling state to
"RTCSessionDescription
".
have-remote-pranswer
If description is of type
"
", and it initiates the closure
of an existing SCTP association, as defined in
[RFC8841], Sections 10.3 and 10.4, set the value
of connection.[[SctpTransport]] to
answer
null
.
Let trackEventInits, muteTracks, addList, removeList and errorList be empty lists.
If description is of type
"
" or "answer
",
then run the following steps:
pranswer
If description initiates the
establishment of a new SCTP association, as
defined in [RFC8841], Sections 10.3 and 10.4,
create an RTCSctpTransport with an initial
state of "
"
and assign the result to the
[[SctpTransport]] slot. Otherwise, if an
SCTP association is established, but the
connecting
max-message-size
SDP
attribute is updated, update the data max
message size of
connection.[[SctpTransport]].
If description negotiates the DTLS
role of the SCTP transport, then for each
, channel, with a
RTCDataChannel
null
, run the
following step:
id
closed
", and add
channnel to errorList.
If description is not of type
"
", then run the following
steps:
rollback
If remote is false
, then
run the following steps for each media
description in description:
If the media description was not yet associated with an
object then run the following steps:
RTCRtpTransceiver
Let transceiver be the
used to create the
media description.
RTCRtpTransceiver
Set transceiver.[[Mid]] to transceiver.[[JsepMid]].
If
transceiver.[[Stopped]]
is true
, abort these sub
steps.
If the media description is
indicated as using an existing media
transport according to [RFC8843],
let transport be the
object representing
the RTP/RTCP component of that transport.
RTCDtlsTransport
Otherwise, let transport be a
newly created
object
with a new underlying
RTCDtlsTransport
.
RTCIceTransport
Set transceiver.[[Sender]].[[SenderTransport]] to transport.
Set transceiver.[[Receiver]].[[ReceiverTransport]] to transport.
Let transceiver be the
associated with
the media description.
RTCRtpTransceiver
If transceiver.[[Stopped]]
is true
, abort these sub steps.
Let direction be an
value
representing the direction from the media
description.
RTCRtpTransceiverDirection
If direction is
"
" or
"sendrecv
",
set
transceiver.[[Receptive]]
to recvonly
true
, otherwise set it to
false
.
Set transceiver.[[Receiver]].[[ReceiveCodecs]] to the codecs that description negotiates for receiving and which the user agent is currently prepared to receive.
If description is of type
"
" or
"answer
", then run the
following steps:
pranswer
Set
transceiver.[[Sender]].[[SendCodecs]]
to the codecs that description
negotiates for sending and which the user
agent is currently capable of sending,
and set
transceiver.[[Sender]].[[LastReturnedParameters]]
to null
.
If direction is
"
"
or
"sendonly
",
and
transceiver.[[FiredDirection]]
is either
"inactive
"
or
"sendrecv
",
then run the following steps:
recvonly
Set the associated remote streams given transceiver.[[Receiver]], an empty list, another empty list, and removeList.
process the removal of a remote track for the media description, given transceiver and muteTracks.
Set transceiver.[[CurrentDirection]] and transceiver.[[FiredDirection]] to direction.
Otherwise, (if remote is
true
) run the following steps for
each media description in
description:
If the description is of type
"
" and contains a request
to receive simulcast, use the order of the
rid values specified in the simulcast
attribute to create an
offer
dictionary for
each of the simulcast layers, populating the
RTCRtpEncodingParameters
member
according to the corresponding rid value, and
let sendEncodings be the list
containing the created dictionaries.
Otherwise, let sendEncodings be an
empty list.
rid
scaleResolutionDownBy
to 2^(length of sendEncodings -
encoding index - 1)
.
As described by [RFC8829] (section 5.10.),
attempt to find an existing
object,
transceiver, to represent the media description.
RTCRtpTransceiver
If a suitable transceiver was found
(transceiver is set) and
sendEncodings is non-empty, set
transceiver.[[Sender]].[[SendEncodings]]
to sendEncodings, and set
transceiver.[[Sender]].[[LastReturnedParameters]]
to null
.
If no suitable transceiver was found (transceiver is unset), run the following steps:
Create an RTCRtpSender, sender, from the media description using sendEncodings.
Create an RTCRtpReceiver, receiver, from the media description.
Create an RTCRtpTransceiver with
sender, receiver
and an
value of
"RTCRtpTransceiverDirection
",
and let transceiver be the
result.
recvonly
Add transceiver to the connection's set of transceivers.
If description is of type
"
" or
"answer
", and
transceiver.
[[Sender]].[[SendEncodings]]
.length is greater than pranswer
1
, then
run the following steps:
If description indicates that simulcast is not supported or desired, then remove all dictionaries in transceiver.[[Sender]].[[SendEncodings]] except the first one and abort these sub steps.
If description rejects any of the offered layers, then remove the dictionaries that correspond to rejected layers from transceiver.[[Sender]].[[SendEncodings]].
Update the paused status as indicated by
[RFC8853] of each simulcast
layer by setting the
member on the corresponding dictionaries
in
transceiver.[[Sender]].[[SendEncodings]]
to active
true
for unpaused or to
false
for paused.
Set transceiver.[[Mid]] to transceiver.[[JsepMid]].
Let direction be an
value
representing the direction from the media
description, but with the send and receive
directions reversed to represent this peer's
point of view. If the media description
is rejected, set direction to
"RTCRtpTransceiverDirection
".
inactive
If direction is
"
" or
"sendrecv
",
let msids be a list of the MSIDs
that the media description indicates
transceiver.[[Receiver]].[[ReceiverTrack]]
is to be associated with. Otherwise, let
msids be an empty list.
recvonly
Process remote tracks with transceiver, direction, msids, addList, removeList, and trackEventInits.
Set transceiver.[[Receiver]].[[ReceiveCodecs]] to the codecs that description negotiates for receiving and which the user agent is currently prepared to receive.
If description is of type
"
" or
"answer
", then run the
following steps:
pranswer
Set transceiver.[[Sender]].[[SendCodecs]] to the codecs that description negotiates for sending and which the user agent is currently capable of sending.
Set transceiver.[[CurrentDirection]] and transceiver.[[Direction]]s to direction.
Let transport be the
object representing
the RTP/RTCP component of the media
transport used by
transceiver's associated
media description, according to
[RFC8843].
RTCDtlsTransport
Set transceiver.[[Sender]].[[SenderTransport]] to transport.
Set transceiver.[[Receiver]].[[ReceiverTransport]] to transport.
Set the [[IceRole]] of transport according to the rules of [RFC8445].
unknown
, do not modify
[[IceRole]].
controlling
.
a=ice-lite
,
set [[IceRole]] to
controlling
.
a=ice-lite
,
set [[IceRole]] to
controlled
.
If the media description is rejected,
and
transceiver.[[Stopped]] is
false
, then stop the
RTCRtpTransceiver transceiver.
Otherwise, (if description is of type
"
") run the following steps:
rollback
Let pendingDescription be either
connection.[[PendingLocalDescription]]
or
connection.[[PendingRemoteDescription]],
whichever one is not null
.
For each transceiver in the connection's set of transceivers run the following steps:
If transceiver was not associated with a media description
prior to pendingDescription being set,
disassociate it and set both
transceiver.[[JsepMid]]
and transceiver.[[Mid]] to
null
.
Set transceiver.[[Sender]].[[SenderTransport]] to transceiver.[[Sender]].[[LastStableStateSenderTransport]].
Set transceiver.[[Receiver]].[[ReceiverTransport]] to transceiver.[[Receiver]].[[LastStableStateReceiverTransport]].
Set transceiver.[[Receiver]].[[ReceiveCodecs]] to transceiver.[[Receiver]].[[LastStableStateReceiveCodecs]].
If the signaling state of
connection is
"
",
run the following sub steps:
have-remote-offer
Let msids be a list of the
id
s of all
MediaStream
objects in
transceiver.[[Receiver]].[[LastStableStateAssociatedRemoteMediaStreams]],
or an empty list if there are none.
Process remote tracks with transceiver, transceiver.[[CurrentDirection]], msids, addList, removeList, and trackEventInits.
If transceiver was created when
pendingDescription was set, and a
track has never been attached to it via
addTrack
()
, then stop the RTCRtpTransceiver
transceiver, and remove it from
connection's set of
transceivers.
Set
connection.[[PendingLocalDescription]]
and
connection.[[PendingRemoteDescription]]
to null
, and set
connection's signaling state to
"
".
stable
If description is of type
"
", then run the following
steps:
answer
For each transceiver in the connection's set of transceivers run the following steps:
If transceiver is
stopped
, associated with an m= section and the associated m=
section is rejected in
connection.[[CurrentLocalDescription]]
or
connection.[[CurrentRemoteDescription]],
remove the transceiver from the
connection's set of
transceivers.
If connection's signaling state is
now "
", run the following
steps:
stable
For any transceiver that was removed
from the set of transceivers in a previous
step, if any of its transports
(transceiver.[[Sender]].[[SenderTransport]]
or
transceiver.[[Receiver]].[[ReceiverTransport]])
are still not closed and they're no longer
referenced by a non-stopped transceiver, close
the
s and their associated
RTCDtlsTransport
s. This results in events
firing on these objects in a queued task.
RTCIceTransport
Clear the negotiation-needed flag and update the negotiation-needed flag.
If connection's signaling state
changed above, fire an event named
at connection.
signalingstatechange
For each channel in errorList,
fire an event named
using the error
interface with the
RTCErrorEvent
attribute set to
"errorDetail
" at
channel.
data-channel-failure
For each track in muteTracks,
set the muted state of track to the
value true
.
For each stream and track pair in removeList, remove the track track from stream.
For each stream and track pair in addList, add the track track to stream.
For each entry entry in
trackEventInits, fire an event named
using the track
interface with
its RTCTrackEvent
attribute initialized
to entry.receiver
,
its receiver
attribute initialized to
entry.track
, its
track
attribute initialized to
entry.streams
and
its streams
attribute
initialized to
entry.transceiver
at
the connection object.
transceiver
Resolve p with
undefined
.
Return p.
To set a configuration, run the following steps:
Let configuration be the
dictionary to be processed.
RTCConfiguration
Let connection be the target
object.
RTCPeerConnection
If configuration.
is set, run the following steps:
certificates
If the length of
configuration.
is different from the length of
connection.[[Configuration]].certificates
,
throw an certificates
InvalidModificationError
.
Let index be initialized to 0.
Let size be initialized to the length of
configuration.
.
certificates
While index is less than size, run the following steps:
If the ECMAScript object represented by the value of
configuration.
at index is not the same as the ECMAScript
object represented by the value of
connection.[[Configuration]].certificates
at index, throw an
certificates
InvalidModificationError
.
Increment index by 1.
If the value of
configuration.
is
set and its value differs from the connection's
bundle policy, throw an
bundlePolicy
InvalidModificationError
.
If the value of
configuration.
is set and its value differs from the connection's
rtcpMux policy, throw an
rtcpMuxPolicy
InvalidModificationError
.
If the value of
configuration.
is set and its value differs from the connection's
previously set iceCandidatePoolSize
, and
iceCandidatePoolSize
has already been
called, throw an
setLocalDescription
InvalidModificationError
.
Set the ICE Agent's ICE transports setting to the
value of
configuration.
.
As defined in [RFC8829] (section 4.1.18.), if the new ICE
transports setting changes the existing setting, no action
will be taken until the next gathering phase. If a script
wants this to happen immediately, it should do an ICE
restart.
iceTransportPolicy
Set the ICE Agent's prefetched ICE candidate pool
size as defined in [RFC8829] (section 3.5.4. and section 4.1.1.) to the
value of
configuration.
.
If the new ICE candidate pool size changes the existing
setting, this may result in immediate gathering of new pooled
candidates, or discarding of existing pooled candidates, as
defined in [RFC8829] (section 4.1.18.).
iceCandidatePoolSize
Let validatedServers be an empty list.
If configuration.
is defined, then run the following steps for each element:
iceServers
Let server be the current list element.
Let urls be
server.
.
urls
If urls is a string, set urls to a list consisting of just that string.
If urls is empty, throw a
SyntaxError
.
For each url in urls run the following steps:
Parse the url using the generic URI syntax
defined in [RFC3986] and obtain the scheme
name. If the parsing based on the syntax
defined in [RFC3986] fails, throw
a SyntaxError
. If the scheme name is
not implemented by the browser throw
a NotSupportedError
. If scheme name is
turn
or turns
, and parsing the url
using the syntax defined in [RFC7065] fails, throw a SyntaxError
. If scheme
name is stun
or
stuns
, and parsing the
url using the syntax defined in
[RFC7064] fails, throw a
SyntaxError
.
If scheme name is turn
or turns
, and either of
server.
or
server.username
are
omitted, then throw an
credential
InvalidAccessError
.
If scheme name is turn
or turns
, and
server.
is
"credentialType
", and
server.password
is not
a DOMString, then
throw an credential
InvalidAccessError
.
Append server to validatedServers.
Set the ICE Agent's ICE servers list to validatedServers.
As defined in [RFC8829] (section 4.1.18.), if a new list of servers replaces the ICE Agent's existing ICE servers list, no action will be taken until the next gathering phase. If a script wants this to happen immediately, it should do an ICE restart. However, if the ICE candidate pool has a nonzero size, any existing pooled candidates will be discarded, and new candidates will be gathered from the new servers.
Store configuration in the [[Configuration]] internal slot.
The RTCPeerConnection
interface presented in this
section is extended by several partial interfaces throughout this
specification. Notably, the RTP Media API section, which adds
the APIs to send and receive MediaStreamTrack
objects.
WebIDL[Exposed=Window] interfaceRTCPeerConnection
: EventTarget {constructor
(optionalRTCConfiguration
configuration = {}); Promise<RTCSessionDescriptionInit
>createOffer
(optionalRTCOfferOptions
options = {}); Promise<RTCSessionDescriptionInit
>createAnswer
(optionalRTCAnswerOptions
options = {}); Promise<undefined>setLocalDescription
(optionalRTCLocalSessionDescriptionInit
description = {}); readonly attributeRTCSessionDescription
?localDescription
; readonly attributeRTCSessionDescription
?currentLocalDescription
; readonly attributeRTCSessionDescription
?pendingLocalDescription
; Promise<undefined>setRemoteDescription
(RTCSessionDescriptionInit
description); readonly attributeRTCSessionDescription
?remoteDescription
; readonly attributeRTCSessionDescription
?currentRemoteDescription
; readonly attributeRTCSessionDescription
?pendingRemoteDescription
; Promise<undefined>addIceCandidate
(optionalRTCIceCandidateInit
candidate = {}); readonly attributeRTCSignalingState
signalingState
; readonly attributeRTCIceGatheringState
iceGatheringState
; readonly attributeRTCIceConnectionState
iceConnectionState
; readonly attributeRTCPeerConnectionState
connectionState
; readonly attribute boolean?canTrickleIceCandidates
; undefinedrestartIce
();RTCConfiguration
getConfiguration
(); undefinedsetConfiguration
(optionalRTCConfiguration
configuration = {}); undefinedclose
(); attribute EventHandleronnegotiationneeded
; attribute EventHandleronicecandidate
; attribute EventHandleronicecandidateerror
; attribute EventHandleronsignalingstatechange
; attribute EventHandleroniceconnectionstatechange
; attribute EventHandleronicegatheringstatechange
; attribute EventHandleronconnectionstatechange
; // Legacy Interface Extensions // Supporting the methods in this section is optional. // If these methods are supported // they must be implemented as defined // in section "Legacy Interface Extensions" Promise<undefined>createOffer
(RTCSessionDescriptionCallback
successCallback,RTCPeerConnectionErrorCallback
failureCallback, optionalRTCOfferOptions
options = {}); Promise<undefined>setLocalDescription
(RTCLocalSessionDescriptionInit
description, VoidFunction successCallback,RTCPeerConnectionErrorCallback
failureCallback); Promise<undefined>createAnswer
(RTCSessionDescriptionCallback
successCallback,RTCPeerConnectionErrorCallback
failureCallback); Promise<undefined>setRemoteDescription
(RTCSessionDescriptionInit
description, VoidFunction successCallback,RTCPeerConnectionErrorCallback
failureCallback); Promise<undefined>addIceCandidate
(RTCIceCandidateInit
candidate, VoidFunction successCallback,RTCPeerConnectionErrorCallback
failureCallback); };
localDescription
of type RTCSessionDescription
, readonly,
nullable
The
attribute MUST return
[[PendingLocalDescription]] if it is not
localDescription
null
and otherwise it MUST return
[[CurrentLocalDescription]].
Note that
[[CurrentLocalDescription]].
and
[[PendingLocalDescription]].sdp
need not be string-wise identical to the
sdp
value passed to the
corresponding sdp
call (i.e. SDP may be
parsed and reformatted, and ICE candidates may be added).
setLocalDescription
currentLocalDescription
of type RTCSessionDescription
, readonly,
nullable
The
attribute MUST return
[[CurrentLocalDescription]].
currentLocalDescription
It represents the local description that was successfully
negotiated the last time the
transitioned into the stable state plus any local
candidates that have been generated by the ICE Agent
since the offer or answer was created.
RTCPeerConnection
pendingLocalDescription
of type RTCSessionDescription
, readonly,
nullable
The
attribute MUST return
[[PendingLocalDescription]].
pendingLocalDescription
It represents a local description that is in the process of
being negotiated plus any local candidates that have been
generated by the ICE Agent since the offer or answer
was created. If the
is in the stable
state, the value is RTCPeerConnection
null
.
remoteDescription
of type RTCSessionDescription
, readonly,
nullable
The
attribute MUST return
[[PendingRemoteDescription]] if it is not
remoteDescription
null
and otherwise it MUST return
[[CurrentRemoteDescription]].
Note that
[[CurrentRemoteDescription]].
and
[[PendingRemoteDescription]].sdp
need not be string-wise identical to the
sdp
value passed to the
corresponding sdp
call (i.e. SDP may be
parsed and reformatted, and ICE candidates may be added).
setRemoteDescription
currentRemoteDescription
of type RTCSessionDescription
, readonly,
nullable
The
attribute MUST return
[[CurrentRemoteDescription]].
currentRemoteDescription
It represents the last remote description that was
successfully negotiated the last time the
transitioned into the stable state
plus any remote candidates that have been supplied via
RTCPeerConnection
addIceCandidate
()
since the offer or
answer was created.
pendingRemoteDescription
of type RTCSessionDescription
, readonly,
nullable
The
attribute MUST return
[[PendingRemoteDescription]].
pendingRemoteDescription
It represents a remote description that is in the process
of being negotiated, complete with any remote candidates
that have been supplied via
addIceCandidate
()
since the offer or
answer was created. If the
is in the
stable state, the value is RTCPeerConnection
null
.
signalingState
of
type RTCSignalingState
,
readonly
The
attribute MUST return the
signalingState
object's signaling state.
RTCPeerConnection
iceGatheringState
of type RTCIceGatheringState
, readonly
The
attribute MUST return the ICE
gathering state of the iceGatheringState
instance.
RTCPeerConnection
iceConnectionState
of type RTCIceConnectionState
, readonly
The
attribute MUST return the ICE
connection state of the iceConnectionState
instance.
RTCPeerConnection
connectionState
of type RTCPeerConnectionState
, readonly
The
attribute MUST return the connection state of the connectionState
instance.
RTCPeerConnection
canTrickleIceCandidates
of type
boolean, readonly, nullable
The
attribute indicates whether
the remote peer is able to accept trickled ICE candidates
[RFC8838]. The value is determined based on whether a
remote description indicates support for trickle ICE, as
defined in [RFC8829] (section 4.1.17.).
Prior to the completion of
canTrickleIceCandidates
, this value is
setRemoteDescription
null
.
onnegotiationneeded
of type
EventHandler
negotiationneeded
.
onicecandidate
of type EventHandler
icecandidate
.
onicecandidateerror
of type
EventHandler
icecandidateerror
.
onsignalingstatechange
of type
EventHandler
signalingstatechange
.
oniceconnectionstatechange
of type
EventHandler
iceconnectionstatechange
onicegatheringstatechange
of type
EventHandler
icegatheringstatechange
.
onconnectionstatechange
of type
EventHandler
connectionstatechange
.
createOffer
The
method generates a blob of SDP that
contains an RFC 3264 offer with the supported
configurations for the session, including descriptions of
the local createOffer
MediaStreamTrack
s attached to this
, the codec/RTP/RTCP capabilities
supported by this implementation, and parameters of the ICE agent and the DTLS connection. The
options parameter may be supplied to provide
additional control over the offer generated.
RTCPeerConnection
If a system has limited resources (e.g. a finite number of
decoders),
needs to return an offer that
reflects the current state of the system, so that
createOffer
will succeed when it attempts to
acquire those resources. The session descriptions MUST
remain usable by setLocalDescription
without causing an
error until at least the end of the fulfillment
callback of the returned promise.
setLocalDescription
Creating the SDP MUST follow the appropriate process for
generating an offer described in [RFC8829], except the user
agent MUST treat a stopping
transceiver as stopped
for the
purposes of RFC8829 in this case.
As an offer, the generated SDP will contain the full set of
codec/RTP/RTCP capabilities supported or preferred by the
session (as opposed to an answer, which will include only a
specific negotiated subset to use). In the event
is called after the session is established,
createOffer
will generate an offer that is compatible
with the current session, incorporating any changes that
have been made to the session since the last complete
offer-answer exchange, such as addition or removal of
tracks. If no changes have been made, the offer will
include the capabilities of the current local description
as well as any additional capabilities that could be
negotiated in an updated offer.
createOffer
The generated SDP will also contain the ICE agent's
,
usernameFragment
and ICE options (as defined
in [RFC5245], Section 14) and may also contain any local
candidates that have been gathered by the agent.
password
The
value in
configuration for the certificates
provides the certificates configured by the application for
the RTCPeerConnection
. These certificates, along with
any default certificates are used to produce a set of
certificate fingerprints. These certificate fingerprints
are used in the construction of SDP.
RTCPeerConnection
The process of generating an SDP exposes a subset of the
media capabilities of the underlying system, which provides
generally persistent cross-origin information on the
device. It thus increases the fingerprinting surface of the
application. In privacy-sensitive contexts, browsers can
consider mitigations such as generating SDP matching only a
common subset of the capabilities.
When the method is called, the user agent MUST run the following steps:
Let connection be the
object on which the method was invoked.
RTCPeerConnection
If connection.[[IsClosed]] is
true
, return a promise rejected with
a newly created InvalidStateError
.
Return the result of chaining the result of creating an offer with connection to connection's operations chain.
To create an offer given connection run the following steps:
If connection's signaling state is
neither "
" nor
"stable
", return a
promise rejected with a newly created have-local-offer
InvalidStateError
.
Let p be a new promise.
In parallel, begin the in-parallel steps to create an offer given connection and p.
Return p.
The in-parallel steps to create an offer given connection and a promise p are as follows:
If connection was not constructed with a set of certificates, and one has not yet been generated, wait for it to be generated.
Inspect the offerer's system state to determine the currently available resources as necessary for generating the offer, as described in [RFC8829] (section 4.1.8.).
If this inspection failed for any reason, reject
p with a newly created
OperationError
and abort these steps.
Queue a task that runs the final steps to create an offer given connection and p.
The final steps to create an offer given connection and a promise p are as follows:
If connection.[[IsClosed]] is
true
, then abort these steps.
If connection was modified in such a way that additional inspection of the offerer's system state is necessary, then in parallel begin the in-parallel steps to create an offer again, given connection and p, and abort these steps.
createOffer
was called when only an audio RTCRtpTransceiver
was
added to connection, but while performing
the in-parallel steps to create an offer, a video
RTCRtpTransceiver
was added, requiring additional
inspection of video system resources.
Given the information that was obtained from previous
inspection, the current state of connection
and its
s, generate an SDP offer,
sdpString, as described in [RFC8829] (section 5.2.).
RTCRtpTransceiver
As described in [RFC8843] (Section 7), if
bundling is used (see
) an
offerer tagged m= section must be selected in order
to negotiate a BUNDLE group. The user agent MUST
choose the m= section that corresponds to the first
non-stopped transceiver in the set of
transceivers as the offerer tagged m= section.
This allows the remote endpoint to predict which
transceiver is the offerer tagged m= section
without having to parse the SDP.
RTCBundlePolicy
The codec preferences of a media
description's associated transceiver is
said to be the value of the
.[[PreferredCodecs]]
with the following filtering applied (or said not
to be set if [[PreferredCodecs]] is empty):
RTCRtpTransceiver
If the
is
"direction
",
exclude any codecs not included in the
intersection of
sendrecv
.RTCRtpSender
(kind).getCapabilities
and
codecs
.RTCRtpReceiver
(kind).getCapabilities
.
codecs
If the
is
"direction
",
exclude any codecs not included in
sendonly
.RTCRtpSender
(kind).getCapabilities
.
codecs
If the
is
"direction
",
exclude any codecs not included in
recvonly
.RTCRtpReceiver
(kind).getCapabilities
.
codecs
The filtering MUST NOT change the order of the codec preferences.
If the length of the [[SendEncodings]] slot
of the
is larger than 1, then for
each encoding given in [[SendEncodings]] of
the RTCRtpSender
, add an RTCRtpSender
a=rid send
line to the corresponding
media section, and add an a=simulcast:send
line giving the RIDs
in the same order as given in the
field. No RID
restrictions are set.
encodings
[RFC8853] section 5.2 specifies that the order of RIDs in the a=simulcast line suggests a proposed order of preference. If the browser decides not to transmit all encodings, one should expect it to stop sending the last encoding in the list first.
Let offer be a newly created
dictionary with its
RTCSessionDescriptionInit
member initialized
to the string "type
" and its
offer
member initialized to
sdpString.
sdp
Set the [[LastCreatedOffer]] internal slot to sdpString.
Resolve p with offer.
createAnswer
The
method generates an [SDP] answer
with the supported configuration for the session that is
compatible with the parameters in the remote configuration.
Like createAnswer
, the returned blob of SDP contains
descriptions of the local createOffer
MediaStreamTrack
s attached to
this
, the codec/RTP/RTCP options
negotiated for this session, and any candidates that have
been gathered by the ICE Agent. The
options parameter may be supplied to provide
additional control over the generated answer.
RTCPeerConnection
Like
, the returned description SHOULD
reflect the current state of the system. The session
descriptions MUST remain usable by createOffer
without causing an error until at least the end of the fulfillment callback of the returned promise.
setLocalDescription
As an answer, the generated SDP will contain a specific codec/RTP/RTCP configuration that, along with the corresponding offer, specifies how the media plane should be established. The generation of the SDP MUST follow the appropriate process for generating an answer described in [RFC8829].
The generated SDP will also contain the ICE agent's
,
usernameFragment
and ICE options (as defined
in [RFC5245], Section 14) and may also contain any local
candidates that have been gathered by the agent.
password
The
value in
configuration for the certificates
provides the certificates configured by the application for
the RTCPeerConnection
. These certificates, along with
any default certificates are used to produce a set of
certificate fingerprints. These certificate fingerprints
are used in the construction of SDP.
RTCPeerConnection
An answer can be marked as provisional, as described in
[RFC8829] (section 4.1.10.1.), by setting
the
to
"type
".
pranswer
When the method is called, the user agent MUST run the following steps:
Let connection be the
object on which the method was invoked.
RTCPeerConnection
If connection.[[IsClosed]] is
true
, return a promise rejected with
a newly created InvalidStateError
.
Return the result of chaining the result of creating an answer with connection to connection's operations chain.
To create an answer given connection run the following steps:
If connection's signaling state is
neither "
" nor
"have-remote-offer
", return a
promise rejected with a newly created have-local-pranswer
InvalidStateError
.
Let p be a new promise.
In parallel, begin the in-parallel steps to create an answer given connection and p.
Return p.
The in-parallel steps to create an answer given connection and a promise p are as follows:
If connection was not constructed with a set of certificates, and one has not yet been generated, wait for it to be generated.
Inspect the answerer's system state to determine the currently available resources as necessary for generating the answer, as described in [RFC8829] (section 4.1.9.).
If this inspection failed for any reason, reject
p with a newly created
OperationError
and abort these steps.
Queue a task that runs the final steps to create an answer given p.
The final steps to create an answer given a promise p are as follows:
If connection.[[IsClosed]] is
true
, then abort these steps.
If connection was modified in such a way that additional inspection of the answerer's system state is necessary, then in parallel begin the in-parallel steps to create an answer again given connection and p, and abort these steps.
createAnswer
was called when an RTCRtpTransceiver
's direction
was "recvonly
", but
while performing the in-parallel steps to create an
answer, the direction was changed to
"sendrecv
", requiring
additional inspection of video encoding resources.
Given the information that was obtained from previous
inspection and the current state of
connection and its
s,
generate an SDP answer, sdpString, as
described in [RFC8829] (section 5.3.).
RTCRtpTransceiver
The codec preferences of an m= section's
associated transceiver is said to be the value of
the
.[[PreferredCodecs]]
with the following filtering applied (or said not
to be set if [[PreferredCodecs]] is empty):
RTCRtpTransceiver
If the
is
"direction
",
exclude any codecs not included in the
intersection of
sendrecv
.RTCRtpSender
(kind).getCapabilities
and
codecs
.RTCRtpReceiver
(kind).getCapabilities
.
codecs
If the
is
"direction
",
exclude any codecs not included in
sendonly
.RTCRtpSender
(kind).getCapabilities
.
codecs
If the
is
"direction
",
exclude any codecs not included in
recvonly
.RTCRtpReceiver
(kind).getCapabilities
.
codecs
The filtering MUST NOT change the order of the codec preferences.
If the length of the [[SendEncodings]] slot
of the
is larger than 1, then for
each encoding given in [[SendEncodings]] of
the RTCRtpSender
, add an RTCRtpSender
a=rid send
line to the corresponding
media section, and add an a=simulcast:send
line giving the RIDs
in the same order as given in the
field. No RID
restrictions are set.
encodings
Let answer be a newly created
dictionary with its
RTCSessionDescriptionInit
member initialized
to the string "type
" and its
answer
member initialized to
sdpString.
sdp
Set the [[LastCreatedAnswer]] internal slot to sdpString.
Resolve p with answer.
setLocalDescription
The
method instructs the
setLocalDescription
to apply the supplied
RTCPeerConnection
as the local
description.
RTCLocalSessionDescriptionInit
This API changes the local media state. In order to
successfully handle scenarios where the application wants
to offer to change from one media format to a different,
incompatible format, the
MUST be able
to simultaneously support use of both the current and
pending local descriptions (e.g. support codecs that exist
in both descriptions) until a final answer is received, at
which point the RTCPeerConnection
can fully adopt the
pending local description, or rollback to the current
description if the remote side rejected the change.
RTCPeerConnection
Passing in a description is optional. If left out, then
will implicitly create an offer or create an answer, as needed. As noted in
[RFC8829] (section 5.4.), if a
description with SDP is passed in, that SDP is not allowed
to have changed from when it was returned from either
setLocalDescription
or createOffer
.
createAnswer
When the method is invoked, the user agent MUST run the following steps:
Let description be the method's first argument.
Let connection be the
object on which the method was invoked.
RTCPeerConnection
Let sdp be
description.
.
sdp
Return the result of chaining the following steps to connection's operations chain:
Let type be
description.
if present, or "type
" if not
present and connection's signaling
state is either "offer
",
"stable
", or
"have-local-offer
";
otherwise "have-remote-pranswer
".
answer
If type is "
", and
sdp is not the empty string and not
equal to
connection.[[LastCreatedOffer]],
then return a promise rejected with a newly
created
offer
InvalidModificationError
and abort these steps.
If type is "
" or
"answer
", and sdp is
not the empty string and not equal to
connection.[[LastCreatedAnswer]],
then return a promise rejected with a newly
created
pranswer
InvalidModificationError
and abort these steps.
If sdp is the empty string, and
type is "
", then run
the following sub steps:
offer
Set sdp to the value of connection.[[LastCreatedOffer]].
If sdp is the empty string, or if it no longer accurately represents the offerer's system state of connection, then let p be the result of creating an offer with connection, and return the result of reacting to p with a fulfillment step that sets the local session description indicated by its first argument.
If sdp is the empty string, and
type is "
" or
"answer
", then run the following
sub steps:
pranswer
Set sdp to the value of connection.[[LastCreatedAnswer]].
If sdp is the empty string, or if it no longer accurately represents the answerer's system state of connection, then let p be the result of creating an answer with connection, and return the result of reacting to p with the following fulfillment steps:
Let answer be the first argument to these fulfillment steps.
Return the result of setting the local
session description indicated by
{type,
answer.
.
}sdp
Return the result of setting the local
session description indicated by {type, sdp}
.
As noted in [RFC8829] (section 5.9.), calling this method may trigger the ICE candidate gathering process by the ICE Agent.
setRemoteDescription
The
method instructs the
setRemoteDescription
to apply the supplied
RTCPeerConnection
as the remote offer or
answer. This API changes the local media state.
RTCSessionDescriptionInit
When the method is invoked, the user agent MUST run the following steps:
Let description be the method's first argument.
Let connection be the
object on which the method was invoked.
RTCPeerConnection
Return the result of chaining the following steps to connection's operations chain:
If
description.
is "type
" and is invalid for the
current signaling state of
connection as described in
[RFC8829] (section 5.5. and section 5.6.),
then run the following sub steps:
offer
Let p be the result of setting
the local session description indicated by
{type:
"
.
"}rollback
Return the result of reacting to p with a fulfillment step that sets the remote session description description, and abort these steps.
Return the result of setting the remote session description description.
addIceCandidate
The
method provides a remote candidate
to the ICE Agent. This method can also be used to
indicate the end of remote candidates when called with an
empty string for the addIceCandidate
member.
The only members of the argument used by this method are
candidate
, candidate
,
sdpMid
, and
sdpMLineIndex
; the rest are ignored.
When the method is invoked, the user agent MUST run the
following steps:
usernameFragment
Let candidate be the method's argument.
Let connection be the
object on which the method was invoked.
RTCPeerConnection
If candidate.
is not an empty string and both
candidate.candidate
and
candidate.sdpMid
are sdpMLineIndex
null
, return a promise rejected
with a newly created TypeError
.
Return the result of chaining the following steps to connection's operations chain:
If
is
remoteDescription
null
return a promise rejected
with a newly created
InvalidStateError
.
If candidate.
is not sdpMid
null
, run the following steps:
If
candidate.
is not equal to the mid of any media
description in
sdpMid
, return
a promise rejected with a newly created remoteDescription
OperationError
.
Else, if
candidate.
is not sdpMLineIndex
null
, run the following steps:
If
candidate.
is equal to or larger than the number of media
descriptions in
sdpMLineIndex
, return
a promise rejected with a newly created remoteDescription
OperationError
.
If either
candidate.
or
candidate.sdpMid
indicate a media description in
sdpMLineIndex
whose
associated transceiver is remoteDescription
stopped
, return a promise resolved with
undefined
.
If
candidate.
is not usernameFragment
null
, and is not equal to any
username fragment present in the corresponding media description of an applied remote
description, return a promise rejected with a
newly created OperationError
.
Let p be a new promise.
In parallel, if the candidate is not administratively prohibited, add the ICE
candidate candidate as described in
[RFC8829] (section 4.1.19.).
Use
candidate.
to identify the ICE generation; if
usernameFragment
is
usernameFragment
null
, process the candidate
for the most recent ICE generation.
If
candidate.
is an empty string, process candidate as
an end-of-candidates indication for the
corresponding media description and ICE
candidate generation. If both
candidate.candidate
and
candidate.sdpMid
are sdpMLineIndex
null
, then this end-of-candidates
indication applies to all media descriptions.
If candidate could not be successfully added the user agent MUST queue a task that runs the following steps:
If
connection.[[IsClosed]]
is true
, then abort these
steps.
Reject p with a newly created OperationError
and
abort these steps.
If candidate is applied successfully, or if the candidate was administratively prohibited the user agent MUST queue a task that runs the following steps:
If
connection.[[IsClosed]]
is true
, then abort these
steps.
If
connection.[[PendingRemoteDescription]]
is not null
, and represents
the ICE generation for which
candidate was processed, add
candidate to
connection.[[PendingRemoteDescription]].sdp.
If
connection.[[CurrentRemoteDescription]]
is not null
, and represents
the ICE generation for which
candidate was processed, add
candidate to
connection.[[CurrentRemoteDescription]].sdp.
Resolve p with
undefined
.
Return p.
A candidate is administratively prohibited if the UA has decided not to allow connection attempts to this address.
For privacy reasons, there is no indication to the developer about whether or not an address/port is blocked; it behaves exactly as if there was no response from the address.
The UA MUST prohibit connections to addresses on the [Fetch] block bad port list, and MAY choose to prohibit connections to other addresses.
If the
member of
the iceTransportPolicy
is
RTCConfiguration
, candidates requiring
external resolution, such as mDNS candidates and DNS
candidates, MUST be prohibited.
relay
Due to WebIDL processing,
(addIceCandidate
null
) is
interpreted as a call with the default dictionary present,
which, in the above algorithm, indicates end-of-candidates
for all media descriptions and ICE candidate generation.
This is by design for legacy reasons.
restartIce
The
method tells the restartIce
that ICE should be restarted. Subsequent calls to
RTCPeerConnection
will create descriptions that will restart
ICE, as described in section 9.1.1.1 of [RFC5245].
createOffer
When this method is invoked, the user agent MUST run the following steps:
Let connection be the
on which the method was invoked.
RTCPeerConnection
Empty connection.[[LocalIceCredentialsToReplace]], and populate it with all ICE credentials (ice-ufrag and ice-pwd as defined in section 15.4 of [RFC5245]) found in connection.[[CurrentLocalDescription]], as well as all ICE credentials found in connection.[[PendingLocalDescription]].
Update the negotiation-needed flag for connection.
getConfiguration
Returns an
object representing the
current configuration of this RTCConfiguration
object.
RTCPeerConnection
When this method is called, the user agent MUST return the
object stored in the
[[Configuration]] internal slot.
RTCConfiguration
setConfiguration
The
method updates the configuration
of this setConfiguration
object. This includes
changing the configuration of the ICE Agent. As noted
in [RFC8829] (section 3.5.1.),
when the ICE configuration changes in a way that requires a
new gathering phase, an ICE restart is required.
RTCPeerConnection
When the
method is invoked, the user
agent MUST run the following steps:
setConfiguration
Let connection be the
on which the method was invoked.
RTCPeerConnection
If connection.[[IsClosed]] is
true
, throw an
InvalidStateError
.
Set the configuration specified by configuration.
close
When the
method is invoked, the user agent MUST
run the following steps:
close
Let connection be the
object on which the method was invoked.
RTCPeerConnection
false
.
The close the connection algorithm given a connection and a disappear boolean, is as follows:
If connection.[[IsClosed]] is
true
, abort these steps.
Set connection.[[IsClosed]] to
true
.
Set connection's signaling state to
"
". This does not fire any
event.
closed
Let transceivers be the result of executing
the CollectTransceivers
algorithm. For every
transceiver in
transceivers, run the following steps:
RTCRtpTransceiver
If transceiver.[[Stopped]] is
true
, abort these sub steps.
Stop the RTCRtpTransceiver with transceiver and disappear.
Set the [[ReadyState]] slot of each of
connection's
s to
"RTCDataChannel
".
closed
RTCDataChannel
s will be closed abruptly and the
closing procedure will not be invoked.
If connection.[[SctpTransport]] is
not null
, tear down the underlying SCTP
association by sending an SCTP ABORT chunk and set the
[[SctpTransportState]] to
"
".
closed
Set the [[DtlsTransportState]] slot of each of
connection's
s to
"RTCDtlsTransport
".
closed
Destroy connection's ICE Agent, abruptly ending any active ICE processing and releasing any relevant resources (e.g. TURN permissions).
Set the [[IceTransportState]] slot of each of
connection's
s to
"RTCIceTransport
".
closed
Set connection's ICE connection state
to "
". This does not
fire any event.
closed
Set connection's connection state to
"
". This does not fire
any event.
closed
RTCPeerConnection
interface since overloaded
functions are not allowed to be defined in partial interfaces.
Supporting the methods in this section is optional. However, if these methods are supported it is mandatory to implement according to what is specified here.
addStream
method that used to exist on
RTCPeerConnection
is easy to polyfill as:
RTCPeerConnection.prototype.addStream = function(stream) {
stream.getTracks().forEach((track) => this.addTrack(track, stream));
};
createOffer
When the createOffer
method
is called, the user agent MUST run the following steps:
Let successCallback be the method's first argument.
Let failureCallback be the callback indicated by the method's second argument.
Let options be the callback indicated by the method's third argument.
Run the steps specified by
's
RTCPeerConnection
createOffer
()
method with
options as the sole argument, and let
p be the resulting promise.
Upon fulfillment of p with value offer, invoke successCallback with offer as the argument.
Upon rejection of p with reason r, invoke failureCallback with r as the argument.
Return a promise resolved with
undefined
.
setLocalDescription
When the setLocalDescription
method is called,
the user agent MUST run the following steps:
Let description be the method's first argument.
Let successCallback be the callback indicated by the method's second argument.
Let failureCallback be the callback indicated by the method's third argument.
Run the steps specified by
's
RTCPeerConnection
method with
description as the sole argument, and let
p be the resulting promise.
setLocalDescription
Upon fulfillment of p, invoke
successCallback with
undefined
as the argument.
Upon rejection of p with reason r, invoke failureCallback with r as the argument.
Return a promise resolved with
undefined
.
createAnswer
createAnswer
method does not take an RTCAnswerOptions
parameter,
since no known legacy createAnswer
implementation ever
supported it.
When the createAnswer
method is called, the user agent MUST run the following
steps:
Let successCallback be the method's first argument.
Let failureCallback be the callback indicated by the method's second argument.
Run the steps specified by
's
RTCPeerConnection
createAnswer
()
method with no
arguments, and let p be the resulting
promise.
Upon fulfillment of p with value answer, invoke successCallback with answer as the argument.
Upon rejection of p with reason r, invoke failureCallback with r as the argument.
Return a promise resolved with
undefined
.
setRemoteDescription
When the setRemoteDescription
method is called,
the user agent MUST run the following steps:
Let description be the method's first argument.
Let successCallback be the callback indicated by the method's second argument.
Let failureCallback be the callback indicated by the method's third argument.
Run the steps specified by
's
RTCPeerConnection
method
with description as the sole argument, and
let p be the resulting promise.
setRemoteDescription
Upon fulfillment of p, invoke
successCallback with
undefined
as the argument.
Upon rejection of p with reason r, invoke failureCallback with r as the argument.
Return a promise resolved with
undefined
.
addIceCandidate
When the addIceCandidate
method is called, the user agent MUST run the following
steps:
Let candidate be the method's first argument.
Let successCallback be the callback indicated by the method's second argument.
Let failureCallback be the callback indicated by the method's third argument.
Run the steps specified by
's
RTCPeerConnection
addIceCandidate
()
method with
candidate as the sole argument, and let
p be the resulting promise.
Upon fulfillment of p, invoke
successCallback with
undefined
as the argument.
Upon rejection of p with reason r, invoke failureCallback with r as the argument.
Return a promise resolved with
undefined
.
These callbacks are only used on the legacy APIs.
RTCPeerConnectionErrorCallback
WebIDLcallback RTCPeerConnectionErrorCallback
= undefined (DOMException error);
RTCPeerConnectionErrorCallback
Parameters
error
of type
DOMException
RTCSessionDescriptionCallback
WebIDLcallbackRTCSessionDescriptionCallback
= undefined (RTCSessionDescriptionInit
description);
RTCSessionDescriptionCallback
Parameters
RTCSessionDescriptionInit
This section describes a set of legacy extensions that may be
used to influence how an offer is created, in addition to the
media added to the
. Developers are
encouraged to use the RTCPeerConnection
API instead.
RTCRtpTransceiver
When
is called with any of the
legacy options specified in this section, run the followings
steps instead of the regular createOffer
steps:
createOffer
Let options be the methods first argument.
Let connection be the current
object.
RTCPeerConnection
For each offerToReceive<Kind>
member in options with kind, kind, run
the following steps:
For each non-stopped
"
" transceiver
of transceiver kind kind, set
transceiver.[[Direction]] to
"sendrecv
".
sendonly
For each non-stopped
"
" transceiver
of transceiver kind kind, set
transceiver.[[Direction]] to
"recvonly
".
inactive
Continue with the next option, if any.
If connection has any non-stopped
"
" or
"sendrecv
" transceivers of
transceiver kind kind, continue with the
next option, if any.
recvonly
Let transceiver be the result of invoking the
equivalent of
connection.
(kind),
except that this operation MUST NOT update the
negotiation-needed flag.
addTransceiver
If transceiver is unset because the previous operation threw an error, abort these steps.
Set transceiver.[[Direction]] to
"
".
recvonly
Run the steps specified by
to create the offer.
createOffer
WebIDLpartial dictionaryRTCOfferOptions
{ booleanofferToReceiveAudio
; booleanofferToReceiveVideo
; };
offerToReceiveAudio
of type boolean
This setting provides additional control over the directionality of audio. For example, it can be used to ensure that audio can be received, regardless if audio is sent or not.
offerToReceiveVideo
of type boolean
This setting provides additional control over the directionality of video. For example, it can be used to ensure that video can be received, regardless if video is sent or not.
An
object MUST not be garbage collected as
long as any event can cause an event handler to be triggered on the
object. When the object's [[IsClosed]] internal slot is
RTCPeerConnection
true
, no such event handler can be triggered and it is
therefore safe to garbage collect the object.
All
and RTCDataChannel
MediaStreamTrack
objects that are
connected to an
have a strong reference to
the RTCPeerConnection
object.
RTCPeerConnection
All methods that return promises are governed by the standard error handling rules of promises. Methods that do not return promises may throw exceptions to indicate errors.
RTCSdpType
The
enum describes the type of an
RTCSdpType
, RTCSessionDescriptionInit
,
or RTCLocalSessionDescriptionInit
instance.
RTCSessionDescription
WebIDLenumRTCSdpType
{ "offer
", "pranswer
", "answer
", "rollback
" };
Enumeration description | |
---|---|
offer
|
An |
pranswer
|
An |
answer
|
An |
rollback
|
An |
RTCSessionDescription
Class
The
class is used by
RTCSessionDescription
to expose local and remote session
descriptions.
RTCPeerConnection
WebIDL[Exposed=Window] interfaceRTCSessionDescription
{constructor
(RTCSessionDescriptionInit
descriptionInitDict); readonly attributeRTCSdpType
type
; readonly attribute DOMStringsdp
; [Default] objecttoJSON
(); };
constructor()
The RTCSessionDescription()
constructor takes a dictionary argument,
description, whose content is used to initialize
the new
object. This constructor
is deprecated; it exists for legacy compatibility reasons
only.
RTCSessionDescription
type
of type RTCSdpType
, readonly
sdp
of type DOMString, readonly, defaulting to
""
toJSON()
WebIDLdictionaryRTCSessionDescriptionInit
{ requiredRTCSdpType
type
; DOMStringsdp
= ""; };
RTCSessionDescriptionInit
Members
type
of type RTCSdpType
, required
sdp
of type DOMString
type
is "rollback
",
this member is unused.
WebIDLdictionaryRTCLocalSessionDescriptionInit
{RTCSdpType
type
; DOMStringsdp
= ""; };
RTCLocalSessionDescriptionInit
Members
type
of type RTCSdpType
setLocalDescription
will infer the type
based on the RTCPeerConnection
's signaling state.
sdp
of type DOMString
type
is
"rollback
", this member is unused.
Many changes to state of an
will require
communication with the remote side via the signaling channel, in
order to have the desired effect. The app can be kept informed as to
when it needs to do signaling, by listening to the RTCPeerConnection
negotiationneeded
event. This event is fired according
to the state of the connection's negotiation-needed flag,
represented by a [[NegotiationNeeded]] internal slot.
This section is non-normative.
If an operation is performed on an
that
requires signaling, the connection will be marked as needing
negotiation. Examples of such operations include adding or stopping
an RTCPeerConnection
, or adding the first RTCRtpTransceiver
.
RTCDataChannel
Internal changes within the implementation can also result in the connection being marked as needing negotiation.
Note that the exact procedures for updating the negotiation-needed flag are specified below.
This section is non-normative.
The negotiation-needed flag is cleared when a session description
of type "
" is
set successfully, and the supplied description
matches the state of the answer
s and
RTCRtpTransceiver
s that currently exist on the
RTCDataChannel
. Specifically, this means that all
non-RTCPeerConnection
stopped
transceivers have an associated section in the local description with matching
properties, and, if any data channels have been created, a data
section exists in the local description.
Note that the exact procedures for updating the negotiation-needed flag are specified below.
The process below occurs where referenced elsewhere in this document. It also may occur as a result of internal changes within the implementation that affect negotiation. If such changes occur, the user agent MUST update the negotiation-needed flag.
To update the negotiation-needed flag for connection, run the following steps:
If the length of connection.[[Operations]]
is not 0
, then set
connection.[[UpdateNegotiationNeededFlagOnEmptyChain]]
to true
, and abort these steps.
Queue a task to run the following steps:
If connection.[[IsClosed]] is
true
, abort these steps.
If the length of
connection.[[Operations]] is not
0
, then set
connection.[[UpdateNegotiationNeededFlagOnEmptyChain]]
to true
, and abort these steps.
If connection's signaling state is not
"
", abort these steps.
stable
The negotiation-needed flag will be updated once the state
transitions to "
", as part of
the steps for setting a session description.
stable
If the result of checking if negotiation is needed is false
,
clear the negotiation-needed flag by setting
connection.[[NegotiationNeeded]] to
false
, and abort these steps.
If connection.[[NegotiationNeeded]] is
already true
, abort these steps.
Set connection.[[NegotiationNeeded]] to
true
.
Fire an event named
at
connection.
negotiationneeded
The task queueing prevents
from firing
prematurely, in the common situation where multiple
modifications to connection are being made at
once.
negotiationneeded
Additionally, we avoid racing with negotiation methods by
only firing
when the operations
chain is empty.
negotiationneeded
To check if negotiation is needed for connection, perform the following checks:
If any implementation-specific negotiation is required, as
described at the start of this section, return
true
.
If
connection.[[LocalIceCredentialsToReplace]]
is not empty, return true
.
Let description be connection.[[CurrentLocalDescription]].
If connection has created any
s,
and no m= section in description has been negotiated
yet for data, return RTCDataChannel
true
.
For each transceiver in connection's set of transceivers, perform the following checks:
If transceiver.[[Stopping]] is
true
and
transceiver.[[Stopped]] is
false
, return true
.
If transceiver isn't stopped
and isn't yet associated with an m= section
in description, return true
.
If transceiver isn't stopped
and is associated with an m= section in
description then perform the following checks:
If transceiver.[[Direction]] is
"
" or
"sendrecv
", and the associated m= section in description
either doesn't contain a single sendonly
a=msid
line, or the number of MSIDs from
the a=msid
lines in this
m=
section, or the MSID values
themselves, differ from what is in
transceiver.sender.[[AssociatedMediaStreamIds]],
return true
.
If description is of type
"
", and the direction of the associated m= section in neither
connection.[[CurrentLocalDescription]]
nor
connection.[[CurrentRemoteDescription]]
matches transceiver.[[Direction]],
return offer
true
. In this step, when the
direction is compared with a direction found in
[[CurrentRemoteDescription]], the description's
direction must be reversed to represent the peer's
point of view.
If description is of type
"
", and the direction of the associated m= section in the description
does not match
transceiver.[[Direction]]
intersected with the offered direction (as described in
[RFC8829] (section 5.3.1.)),
return answer
true
.
If transceiver is stopped
and is associated with an m= section, but the
associated m= section is not yet rejected in
connection.[[CurrentLocalDescription]]
or
connection.[[CurrentRemoteDescription]],
return true
.
If all the preceding checks were performed and
true
was not returned, nothing remains to be
negotiated; return false
.
RTCIceCandidate
Interface
This interface describes an ICE candidate, described in [RFC5245]
Section 2. Other than
,
candidate
,
sdpMid
, and
sdpMLineIndex
, the remaining attributes
are derived from parsing the usernameFragment
member in candidateInitDict, if it is well formed.
candidate
WebIDL[Exposed=Window] interfaceRTCIceCandidate
{constructor
(optionalRTCIceCandidateInit
candidateInitDict = {}); readonly attribute DOMStringcandidate
; readonly attribute DOMString?sdpMid
; readonly attribute unsigned short?sdpMLineIndex
; readonly attribute DOMString?foundation
; readonly attributeRTCIceComponent
?component
; readonly attribute unsigned long?priority
; readonly attribute DOMString?address
; readonly attributeRTCIceProtocol
?protocol
; readonly attribute unsigned short?port
; readonly attributeRTCIceCandidateType
?type
; readonly attributeRTCIceTcpCandidateType
?tcpType
; readonly attribute DOMString?usernameFragment
;RTCIceCandidateInit
toJSON
(); };
constructor()
The RTCIceCandidate()
constructor
takes a dictionary argument, candidateInitDict,
whose content is used to initialize the new
object.
RTCIceCandidate
When invoked, run the following steps:
sdpMid
and
sdpMLineIndex
members of
candidateInitDict are null
, throw a TypeError
.
Return the result of creating an RTCIceCandidate with candidateInitDict.
To create an RTCIceCandidate with a candidateInitDict dictionary, run the following steps:
RTCIceCandidate
object.
null
:
foundation
, component
, priority
, address
,
protocol
, port
, type
, tcpType
,
relatedAddress
, and relatedPort
.
candidate
, sdpMid
,
sdpMLineIndex
, usernameFragment
.
candidate
dictionary member of
candidateInitDict. If candidate is
not an empty string, run the following steps:
candidate-attribute
grammar.
candidate-attribute
has failed,
abort these steps.
The constructor for
only does basic
parsing and type checking for the dictionary members in
candidateInitDict. Detailed validation on the
well-formedness of RTCIceCandidate
,
candidate
,
sdpMid
,
sdpMLineIndex
with the
corresponding session description is done when passing
the usernameFragment
object to
RTCIceCandidate
addIceCandidate
()
.
To maintain backward compatibility, any error on parsing
the candidate attribute is ignored. In such
case, the
attribute holds the raw
candidate
string given in
candidateInitDict, but derivative attributes
such as candidate
, foundation
, etc are set to
priority
null
.
Most attributes below are defined in section 15.1 of [RFC5245].
candidate
of type DOMString, readonly
candidate-attribute
as defined in
section 15.1 of [RFC5245]. If this RTCIceCandidate
represents an end-of-candidates indication or a peer
reflexive remote candidate, candidate
is an empty string.
sdpMid
of type DOMString, readonly, nullable
null
, this contains the media stream
"identification-tag" defined in [RFC5888] for the
media component this candidate is associated with.
sdpMLineIndex
of type unsigned short, readonly, nullable
null
, this indicates the index (starting
at zero) of the media description in the SDP this
candidate is associated with.
foundation
of type DOMString, readonly, nullable
RTCIceTransport
s.
component
of type RTCIceComponent
, readonly, nullable
rtp
" or "rtcp
").
This corresponds to the component-id
field in candidate-attribute
, decoded to the string
representation as defined in RTCIceComponent
.
priority
of type unsigned long, readonly, nullable
address
of type DOMString, readonly, nullable
The address of the candidate, allowing for IPv4 addresses,
IPv6 addresses, and fully qualified domain names (FQDNs).
This corresponds to the connection-address
field in candidate-attribute
.
Remote candidates may be exposed, for instance via
[[SelectedCandidatePair]].
.
By default, the user agent MUST leave the
remote
attribute as address
null
for any exposed remote candidate. Once a
instance learns on an address by the
web application using
RTCPeerConnection
, the user agent can
expose the addIceCandidate
attribute value in any
address
of the RTCIceCandidate
instance
representing a remote candidate with that newly learnt
address.
RTCPeerConnection
The addresses exposed in candidates gathered via ICE and
made visibile to the application in
instances can reveal more information about the device
and the user (e.g. location, local network topology) than
the user might have expected in a non-WebRTC enabled
browser.
RTCIceCandidate
These addresses are always exposed to the application, and potentially exposed to the communicating party, and can be exposed without any specific user consent (e.g. for peer connections used with data channels, or to receive media only).
These addresses can also be used as temporary or
persistent cross-origin states, and thus contribute to
the fingerprinting surface of the device.
Applications can avoid exposing addresses to the
communicating party, either temporarily or permanently,
by forcing the ICE Agent to report only relay
candidates via the
member of
iceTransportPolicy
.
RTCConfiguration
To limit the addresses exposed to the application itself, browsers can offer their users different policies regarding sharing local addresses, as defined in [RFC8828].
protocol
of type RTCIceProtocol
, readonly, nullable
udp
"/"tcp
"). This
corresponds to the transport
field
in candidate-attribute
.
port
of type unsigned short, readonly, nullable
type
of type RTCIceCandidateType
, readonly,
nullable
candidate-types
field in candidate-attribute
.
tcpType
of type RTCIceTcpCandidateType
, readonly,
nullable
protocol
is "tcp
", tcpType
represents the type of TCP candidate. Otherwise, tcpType
is null
. This corresponds to the tcp-type
field in candidate-attribute
.
relatedAddress
of type DOMString, readonly, nullable
relatedAddress
is the IP
address of the candidate that it is derived from. For host
candidates, the relatedAddress
is null
. This
corresponds to the rel-address
field
in candidate-attribute
.
relatedPort
of type unsigned short, readonly, nullable
relatedPort
is the port of
the candidate that it is derived from. For host candidates,
the relatedPort
is null
. This corresponds to
the rel-port
field in candidate-attribute
.
usernameFragment
of type DOMString, readonly, nullable
ufrag
as defined in
section 15.4 of [RFC5245].
toJSON()
toJSON
()
operation of the
RTCIceCandidate
interface, run the following steps:
RTCIceCandidateInit
dictionary.
candidate
, sdpMid
,
sdpMLineIndex
, usernameFragment
»:
RTCIceCandidate
object.
json[attr]
to value.
WebIDLdictionaryRTCIceCandidateInit
{ DOMStringcandidate
= ""; DOMString?sdpMid
= null; unsigned short?sdpMLineIndex
= null; DOMString?usernameFragment
= null; };
RTCIceCandidateInit
Members
candidate
of type DOMString, defaulting to
""
candidate-attribute
as defined in
section 15.1 of [RFC5245]. If this represents an
end-of-candidates indication, candidate
is an empty
string.
sdpMid
of type DOMString, nullable, defaulting to
null
null
, this contains the media stream
"identification-tag" defined in [RFC5888] for the media
component this candidate is associated with.
sdpMLineIndex
of type unsigned short, nullable, defaulting
to null
null
, this indicates the index (starting
at zero) of the media description in the SDP this
candidate is associated with.
usernameFragment
of type DOMString, nullable, defaulting to
null
null
, this carries the ufrag
as defined in section 15.4 of [RFC5245].
candidate-attribute
Grammar
The candidate-attribute
grammar is used to parse the
member of
candidateInitDict in the candidate
RTCIceCandidate
()
constructor.
The primary grammar for candidate-attribute
is defined in
section 15.1 of [RFC5245]. In addition, the browser MUST support
the grammar extension for ICE TCP as defined in section 4.5 of
[RFC6544].
The browser MAY support other grammar extensions for candidate-attribute
as defined in other RFCs.
RTCIceProtocol
Enum
The
represents the protocol of the ICE
candidate.
RTCIceProtocol
RTCIceTcpCandidateType
Enum
The
represents the type of the ICE TCP
candidate, as defined in [RFC6544].
RTCIceTcpCandidateType
WebIDLenumRTCIceTcpCandidateType
{ "active
", "passive
", "so
" };
Enumeration description | |
---|---|
active
|
An " " TCP candidate is
one for which the transport will attempt to open an
outbound connection but will not receive incoming
connection requests.
|
passive
|
A " " TCP candidate is
one for which the transport will receive incoming
connection attempts but not attempt a connection.
|
so
|
An " " candidate is one for
which the transport will attempt to open a connection
simultaneously with its peer.
|
The user agent will typically only gather
ICE TCP candidates.
active
RTCIceCandidateType
Enum
The
represents the type of the ICE
candidate, as defined in [RFC5245] section 15.1.
RTCIceCandidateType
WebIDLenumRTCIceCandidateType
{ "host
", "srflx
", "prflx
", "relay
" };
Enumeration description | |
---|---|
host
|
A host candidate, as defined in Section 4.1.1.1 of [RFC5245]. |
srflx
|
A server reflexive candidate, as defined in Section 4.1.1.2 of [RFC5245]. |
prflx
|
A peer reflexive candidate, as defined in Section 4.1.1.2 of [RFC5245]. |
relay
|
A relay candidate, as defined in Section 7.1.3.2.1 of [RFC5245]. |
RTCPeerConnectionIceEvent
The icecandidate
event of the
uses the RTCPeerConnection
interface.
RTCPeerConnectionIceEvent
When firing an
event that contains an
RTCPeerConnectionIceEvent
object, it MUST include values for both
RTCIceCandidate
and sdpMid
.
If the sdpMLineIndex
is of type
"RTCIceCandidate
" or type
"srflx
", the
relay
property of the event MUST be set
to the URL of the ICE server from which the candidate was obtained.
url
icecandidate
event is used for three different types of
indications:
A candidate has been gathered. The
member of the event
will be populated normally. It should be signaled to the
remote peer and passed into
candidate
.
addIceCandidate
An
has finished gathering a generation of candidates, and is providing an end-of-candidates
indication as defined by Section 8.2 of [RFC8838]. This
is indicated by
RTCIceTransport
.candidate
being set to an empty string. The
candidate
object should be
signaled to the remote peer and passed into
candidate
like a typical ICE
candidate, in order to provide the end-of-candidates
indication to the remote peer.
addIceCandidate
All
s have finished gathering candidates,
and the RTCIceTransport
's RTCPeerConnection
has
transitioned to "RTCIceGatheringState
". This is
indicated by the complete
member of the event being set to candidate
null
. This only
exists for backwards compatibility, and this event does not
need to be signaled to the remote peer. It's equivalent to an
event with the
"icegatheringstatechange
" state.
complete
WebIDL[Exposed=Window] interfaceRTCPeerConnectionIceEvent
: Event {constructor
(DOMString type, optionalRTCPeerConnectionIceEventInit
eventInitDict = {}); readonly attributeRTCIceCandidate
?candidate
; readonly attribute DOMString?url
; };
RTCPeerConnectionIceEvent.constructor()
candidate
of type RTCIceCandidate
, readonly, nullable
The
attribute is the candidate
object with the new ICE candidate that caused the event.
RTCIceCandidate
This attribute is set to null
when an event is
generated to indicate the end of candidate gathering.
Even where there are multiple media components, only one
event containing a null
candidate is fired.
url
of type DOMString, readonly, nullable
The
attribute is the STUN or TURN URL that
identifies the STUN or TURN server used to gather this
candidate. If the candidate was not gathered from a STUN or
TURN server, this parameter will be set to
url
null
.
WebIDL dictionaryRTCPeerConnectionIceEventInit
: EventInit {RTCIceCandidate
?candidate
; DOMString?url
; };
RTCPeerConnectionIceEventInit
Members
candidate
of type RTCIceCandidate
, nullable
See the
attribute
of the candidate
interface.
RTCPeerConnectionIceEvent
url
of type DOMString, nullable
url
attribute is the STUN or TURN URL that identifies
the STUN or TURN server used to gather this candidate.
RTCPeerConnectionIceErrorEvent
The icecandidateerror
event of the
uses the RTCPeerConnection
interface.
RTCPeerConnectionIceErrorEvent
WebIDL[Exposed=Window] interfaceRTCPeerConnectionIceErrorEvent
: Event {constructor
(DOMString type,RTCPeerConnectionIceErrorEventInit
eventInitDict); readonly attribute DOMString?address
; readonly attribute unsigned short?port
; readonly attribute DOMStringurl
; readonly attribute unsigned shorterrorCode
; readonly attribute USVStringerrorText
; };
RTCPeerConnectionIceErrorEvent.constructor()
address
of type DOMString, readonly, nullable
The
attribute is the local IP address used to
communicate with the STUN or TURN server.
address
On a multihomed system, multiple interfaces may be used to contact the server, and this attribute allows the application to figure out on which one the failure occurred.
If the local IP address value is not already exposed as
part of a local candidate, the
attribute will
be set to address
null
.
port
of type unsigned short, readonly, nullable
The
attribute is the port used to communicate with
the STUN or TURN server.
port
If the
attribute is address
null
, the
attribute is also set to port
null
.
url
of type DOMString, readonly
The
attribute is the STUN or TURN URL that
identifies the STUN or TURN server for which the failure
occurred.
url
errorCode
of type unsigned short, readonly
The
attribute is the numeric STUN error code
returned by the STUN or TURN server [STUN-PARAMETERS].
errorCode
If no host candidate can reach the server,
will be set to the value 701 which is outside the STUN
error code range. This error is only fired once per server
URL while in the errorCode
of
"RTCIceGatheringState
".
gathering
errorText
of type USVString, readonly
The
attribute is the STUN reason text
returned by the STUN or TURN server [STUN-PARAMETERS].
errorText
If the server could not be reached,
will be
set to an implementation-specific value providing details
about the error.
errorText
WebIDL dictionaryRTCPeerConnectionIceErrorEventInit
: EventInit { DOMString?address
; unsigned short?port
; DOMStringurl
; required unsigned shorterrorCode
; USVStringerrorText
; };
RTCPeerConnectionIceErrorEventInit
Members
address
of type DOMString, nullable
The local address used to communicate with the STUN or TURN
server, or null
.
port
of type unsigned short, nullable
The local port used to communicate with the STUN or TURN
server, or null
.
url
of type DOMString
The STUN or TURN URL that identifies the STUN or TURN server for which the failure occurred.
errorCode
of type unsigned short, required
The numeric STUN error code returned by the STUN or TURN server.
errorText
of type USVString
The STUN reason text returned by the STUN or TURN server.
The certificates that
instances use to
authenticate with peers use the RTCPeerConnection
interface. These
objects can be explicitly generated by applications using the
RTCCertificate
method and can be provided
in the generateCertificate
when constructing a new
RTCConfiguration
instance.
RTCPeerConnection
The explicit certificate management functions provided here are
optional. If an application does not provide the
configuration option when
constructing an certificates
a new set of certificates MUST
be generated by the user agent. That set MUST include an ECDSA
certificate with a private key on the P-256 curve and a signature
with a SHA-256 hash.
RTCPeerConnection
WebIDLpartial interfaceRTCPeerConnection
{ static Promise<RTCCertificate
>generateCertificate
(AlgorithmIdentifier keygenAlgorithm); };
generateCertificate
, static
The
function causes the user
agent to create an X.509 certificate [X509V3] and
corresponding private key. A handle to information is
provided in the form of the generateCertificate
interface. The
returned RTCCertificate
can be used to control the
certificate that is offered in the DTLS sessions established
by RTCCertificate
.
RTCPeerConnection
The keygenAlgorithm argument is used to control how the private key associated with the certificate is generated. The keygenAlgorithm argument uses the WebCrypto [WebCryptoAPI] AlgorithmIdentifier type.
The following values MUST be supported by a user
agent: { name: "RSASSA-PKCS1-v1_5",
modulusLength: 2048, publicExponent: new Uint8Array([1, 0,
1]), hash: "SHA-256" }
, and { name:
"ECDSA", namedCurve:
"P-256"
}
.
It is expected that a user agent will have a small or even fixed set of values that it will accept.
The certificate produced by this process also contains a
signature. The validity of this signature is only relevant
for compatibility reasons. Only the public key and the
resulting certificate fingerprint are used by
, but it is more likely that a
certificate will be accepted if the certificate is well
formed. The browser selects the algorithm used to sign the
certificate; a browser SHOULD select SHA-256 [FIPS-180-4]
if a hash algorithm is needed.
RTCPeerConnection
The resulting certificate MUST NOT include information that can be linked to a user or user agent. Randomized values for distinguished name and serial number SHOULD be used.
When the method is called, the user agent MUST run the following steps:
Let keygenAlgorithm be the first argument to
.
generateCertificate
Let expires be a DOMTimeStamp
value of
2592000000.
This means the certificate will by default expire in 30
days from the time of the
call.
generateCertificate
If keygenAlgorithm is an object, run the following steps:
Let certificateExpiration be the result of
converting
the ECMAScript object represented by
keygenAlgorithm to an
dictionary.
RTCCertificateExpiration
If the conversion fails with an error, return a promise that is rejected with error.
If
certificateExpiration.
is not expires
undefined
, set expires
to
certificateExpiration.
.
expires
If expires is greater than 31536000000, set expires to 31536000000.
This means the certificate cannot be valid for
longer than 365 days from the time of the
call.
generateCertificate
A user agent MAY further cap the value of expires.
Let normalizedKeygenAlgorithm be the result of
normalizing an
algorithm with an operation name of generateKey
and a supportedAlgorithms
value specific to production of certificates for
.
RTCPeerConnection
If the above normalization step fails with an error, return a promise that is rejected with error.
If the normalizedKeygenAlgorithm parameter
identifies an algorithm that the user agent cannot
or will not use to generate a certificate for
, return a promise that is rejected with a RTCPeerConnection
DOMException
of type
NotSupportedError
. In particular,
normalizedKeygenAlgorithm MUST be an
asymmetric algorithm that can be used to produce a
signature used to authenticate DTLS connections.
Let p be a new promise.
Run the following steps in parallel:
Perform the generate key operation specified by normalizedKeygenAlgorithm using keygenAlgorithm.
Let generatedKeyingMaterial and generatedKeyCertificate be the private keying material and certificate generated by the above step.
Let certificate be a new
object.
RTCCertificate
Set certificate.[[Expires]] to the current time plus expires value.
Set certificate.[[Origin]] to the relevant settings object's origin.
Store the generatedKeyingMaterial in a secure module, and let handle be a reference identifier to it.
Set certificate.[[KeyingMaterialHandle]] to handle.
Set certificate.[[Certificate]] to generatedCertificate.
Resolve p with certificate.
Return p.
RTCCertificateExpiration
Dictionary
is used to set an expiration date on
certificates generated by
RTCCertificateExpiration
.
generateCertificate
WebIDLdictionaryRTCCertificateExpiration
{ [EnforceRange] DOMTimeStampexpires
; };
expires
, of type DOMTimeStamp
An optional
attribute MAY be added to the
definition of the algorithm that is passed to
expires
. If this parameter is
present it indicates the maximum time that the
generateCertificate
is valid for relative to the current time.
RTCCertificate
RTCCertificate
Interface
The
interface represents a certificate used to
authenticate WebRTC communications. In addition to the visible
properties, internal slots contain a handle to the generated
private keying materal ([[KeyingMaterialHandle]]), a
certificate ([[Certificate]]) that
RTCCertificate
uses to authenticate with a peer, and the
origin ([[Origin]]) that created the object.
RTCPeerConnection
WebIDL[Exposed=Window, Serializable] interfaceRTCCertificate
{ readonly attribute DOMTimeStampexpires
; sequence<RTCDtlsFingerprint
>getFingerprints
(); };
expires
of type DOMTimeStamp, readonly
The expires attribute indicates the date and
time in milliseconds relative to 1970-01-01T00:00:00Z after
which the certificate will be considered invalid by the
browser. After this time, attempts to construct an
using this certificate fail.
RTCPeerConnection
Note that this value might not be reflected in a
notAfter
parameter in the
certificate itself.
getFingerprints
Returns the list of certificate fingerprints, one of which is computed with the digest algorithm used in the certificate signature.
For the purposes of this API, the [[Certificate]] slot
contains unstructured binary data. No mechanism is provided for
applications to access the [[KeyingMaterialHandle]]
internal slot or the keying material it references. Implementations
MUST support applications storing and retrieving
objects from persistent storage, in a manner that also preserves
the keying material referenced by [[KeyingMaterialHandle]].
Implementations SHOULD store the sensitive keying material in a
secure module safe from same-process memory attacks. This allows
the private key to be stored and used, but not easily read using a
memory attack.
RTCCertificate
objects are serializable objects
[HTML]. Their serialization steps, given value
and serialized, are:
RTCCertificate
expires
attribute.
Their deserialization steps, given serialized and value, are:
expires
attribute to contain serialized.[[Expires]].
Supporting structured cloning in this manner allows
instances to be persisted to stores. It also
allows instances to be passed to other origins using APIs like
RTCCertificate
postMessage
(message, options)
[html]. However, the object cannot
be used by any other origin than the one that originally created
it.
The RTP media API lets a web application send and receive
MediaStreamTrack
s over a peer-to-peer connection. Tracks, when
added to an
, result in signaling; when this
signaling is forwarded to a remote peer, it causes corresponding tracks
to be created on the remote side.
RTCPeerConnection
There is not an exact 1:1 correspondence between tracks sent by one
and received by the other. For one, IDs of tracks
sent have no mapping to the IDs of tracks received. Also,
RTCPeerConnection
changes the track sent by an
replaceTrack
without creating a new track on the receiver side; the
corresponding RTCRtpSender
will only have a single track,
potentially representing multiple sources of media stitched together.
Both RTCRtpReceiver
and
addTransceiver
can be used to cause the same track to be
sent multiple times, which will be observed on the receiver side as
multiple receivers each with its own separate track. Thus it's more
accurate to think of a 1:1 relationship between an replaceTrack
on
one side and an RTCRtpSender
's track on the other side, matching
senders and receivers using the RTCRtpReceiver
's
RTCRtpTransceiver
if necessary.
mid
When sending media, the sender may need to rescale or resample the media to meet various requirements including the envelope negotiated by SDP.
Following the rules in [RFC8829] (section 3.6.), the video MAY be downscaled in order to fit the SDP constraints. The media MUST NOT be upscaled to create fake data that did not occur in the input source, the media MUST NOT be cropped except as needed to satisfy constraints on pixel counts, and the aspect ratio MUST NOT be changed.
The WebRTC Working Group is seeking implementation feedback on the need and timeline for a more complex handling of this situation. Some possible designs have been discussed in GitHub issue 1283.
When video is rescaled, for example for certain combinations of width
or height and
values, situations when the resulting width or height is not an integer
may occur. In such situations the user agent MUST use the integer part of the
result. What to transmit if the integer part of the scaled width or
height is zero is implementation-specific.
scaleResolutionDownBy
The actual encoding and transmission of MediaStreamTrack
s is
managed through objects called
s. Similarly, the
reception and decoding of RTCRtpSender
MediaStreamTrack
s is managed through
objects called
s. Each RTCRtpReceiver
is associated
with at most one track, and each track to be received is associated
with exactly one RTCRtpSender
.
RTCRtpReceiver
The encoding and transmission of each MediaStreamTrack
SHOULD be
made such that its characteristics (width
,
height
and frameRate
for video tracks; sampleSize
, sampleRate
and
channelCount
for audio tracks) are to a
reasonable degree retained by the track created on the remote side.
There are situations when this does not apply, there may for example be
resource constraints at either endpoint or in the network or there may
be
settings applied that instruct the implementation
to act differently.
RTCRtpSender
An
object contains a set of RTCPeerConnection
s,
representing the paired senders and receivers with some shared state.
This set is
initialized to the empty set when the RTCRtpTransceiver
object is
created. RTCPeerConnection
s and RTCRtpSender
s are always
created at the same time as an RTCRtpReceiver
, which they will
remain attached to for their lifetime. RTCRtpTransceiver
s are
created implicitly when the application attaches a RTCRtpTransceiver
MediaStreamTrack
to an
via the RTCPeerConnection
addTrack
()
method, or explicitly when the application uses the
method. They are also created when
a remote description is applied that includes a new media description.
Additionally, when a remote description is applied that indicates the
remote endpoint has media to send, the relevant addTransceiver
MediaStreamTrack
and
are surfaced to the application via the
RTCRtpReceiver
event.
track
In order for an
to send and/or receive media with
another endpoint this must be negotiated with SDP such that both
endpoints have an RTCRtpTransceiver
object that is associated
with the same media description.
RTCRtpTransceiver
When creating an offer, enough media descriptions will be generated to cover all transceivers on that end. When this offer is set as the local description, any disassociated transceivers get associated with media descriptions in the offer.
When an offer is set as the remote description, any media descriptions
in it not yet associated with a transceiver get associated with a new
or existing transceiver. In this case, only disassociated transceivers
that were created via the addTrack
()
method may
be associated. Disassociated transceivers created via the
addTransceiver
()
method, however, won't get
associated even if media descriptions are available in the remote
offer. Instead, new transceivers will be created and associated if
there aren't enough addTrack
()
-created
transceivers. This sets addTrack
()
-created and
addTransceiver
()
-created transceivers apart in a
critical way that is not observable from inspecting their attributes.
When creating an answer, only media media descriptions that were
present in the offer may be listed in the answer. As a consequence, any
transceivers that were not associated when setting the remote offer
remain disassociated after setting the local answer. This can be
remedied by the answerer creating a follow-up offer, initiating another
offer/answer exchange, or in the case of using
addTrack
()
-created transceivers, making sure that
enough media descriptions are offered in the initial exchange.
The RTP media API extends the
interface as
described below.
RTCPeerConnection
WebIDL partial interfaceRTCPeerConnection
{ sequence<RTCRtpSender
>getSenders
(); sequence<RTCRtpReceiver
>getReceivers
(); sequence<RTCRtpTransceiver
>getTransceivers
();RTCRtpSender
addTrack
(MediaStreamTrack track, MediaStream... streams); undefinedremoveTrack
(RTCRtpSender
sender);RTCRtpTransceiver
addTransceiver
((MediaStreamTrack or DOMString) trackOrKind, optionalRTCRtpTransceiverInit
init = {}); attribute EventHandlerontrack
; };
ontrack
of type EventHandler
The event type of this event handler is
.
track
getSenders
Returns a sequence of
objects representing
the RTP senders that belong to non-stopped
RTCRtpSender
objects currently attached to this
RTCRtpTransceiver
object.
RTCPeerConnection
When the
method is invoked, the user agent
MUST return the result of executing the getSenders
CollectSenders
algorithm.
We define the CollectSenders algorithm as follows:
CollectTransceivers
algorithm.
false
, add
transceiver.[[Sender]] to
senders.
getReceivers
Returns a sequence of
objects representing
the RTP receivers that belong to non-stopped
RTCRtpReceiver
objects currently attached to this
RTCRtpTransceiver
object.
RTCPeerConnection
When the
method is invoked, the user agent
MUST run the following steps:
getReceivers
CollectTransceivers
algorithm.
false
, add
transceiver.[[Receiver]] to
receivers.
getTransceivers
Returns a sequence of
objects
representing the RTP transceivers that are currently attached
to this RTCRtpTransceiver
object.
RTCPeerConnection
The
method MUST return the result of
executing the getTransceivers
CollectTransceivers
algorithm.
We define the CollectTransceivers algorithm as follows:
RTCRtpTransceiver
objects in this RTCPeerConnection
object's set of transceivers, in insertion order.
addTrack
Adds a new track to the
, and indicates
that it is contained in the specified RTCPeerConnection
MediaStream
s.
When the
method is invoked, the user agent MUST
run the following steps:
addTrack
Let connection be the
object on which this method was invoked.
RTCPeerConnection
Let track be the MediaStreamTrack
object
indicated by the method's first argument.
Let kind be track.kind.
Let streams be a list of MediaStream
objects constructed from the method's remaining
arguments, or an empty list if the method was called with
a single argument.
If connection.[[IsClosed]] is
true
, throw an
InvalidStateError
.
Let senders be the result of executing the
CollectSenders
algorithm. If an
for
track already exists in senders, throw an RTCRtpSender
InvalidAccessError
.
The steps below describe how to determine if an existing
sender can be reused. Doing so will cause future calls to
and
createOffer
to mark the
corresponding media description as createAnswer
sendrecv
or sendonly
and add the MSID of the sender's
streams, as defined in [RFC8829] (section 5.2.2. and section 5.3.2.).
If any
object in senders
matches all the following criteria, let sender
be that object, or RTCRtpSender
null
otherwise:
The sender's track is null.
The transceiver kind of the
, associated with the sender,
matches kind.
RTCRtpTransceiver
The [[Stopping]] slot of the
associated with the sender is
RTCRtpTransceiver
false
.
The sender has never been used to send. More
precisely, the [[CurrentDirection]] slot of
the
associated with the sender
has never had a value of
"RTCRtpTransceiver
" or
"sendrecv
".
sendonly
If sender is not null
, run the
following steps to use that sender:
Set sender.[[SenderTrack]] to track.
Set sender.[[AssociatedMediaStreamIds]] to an empty set.
For each stream in streams, add stream.id to [[AssociatedMediaStreamIds]] if it's not already there.
Let transceiver be the
associated with
sender.
RTCRtpTransceiver
If transceiver.[[Direction]] is
"
", set
transceiver.[[Direction]] to
"recvonly
".
sendrecv
If transceiver.[[Direction]] is
"
", set
transceiver.[[Direction]] to
"inactive
".
sendonly
If sender is null
, run the
following steps:
Create an RTCRtpSender with track, kind and streams, and let sender be the result.
Create an RTCRtpReceiver with kind, and let receiver be the result.
Create an RTCRtpTransceiver with
sender, receiver and an
value of
"RTCRtpTransceiverDirection
", and let
transceiver be the result.
sendrecv
Add transceiver to connection's set of transceivers.
A track could have contents that are inaccessible to the
application. This can be due to anything that would make
a track CORS
cross-origin. These tracks can be supplied to the
addTrack
()
method, and have an
created for them, but content MUST NOT
be transmitted. Silence (audio), black frames (video) or
equivalently absent content is sent in place of track
content.
RTCRtpSender
Note that this property can change over time.
Update the negotiation-needed flag for connection.
Return sender.
removeTrack
Stops sending media from sender. The
will still appear in RTCRtpSender
. Doing
so will cause future calls to getSenders
to mark the media description for the corresponding transceiver as
"createOffer
" or
"recvonly
", as defined in
[RFC8829] (section 5.2.2.).
inactive
When the other peer stops sending a track in this manner, the
track is removed from any remote MediaStream
s that were
initially revealed in the track
event, and if the MediaStreamTrack
is not already muted,
a mute
event is fired at the
track.
removeTrack
()
can be achieved by
setting the
RTCRtpTransceiver
.direction
attribute of the corresponding transceiver and invoking
RTCRtpSender
.replaceTrack
(null) on the
sender. One minor difference is that
replaceTrack
()
is asynchronous and
removeTrack
()
is synchronous.
When the
method is invoked, the user agent
MUST run the following steps:
removeTrack
Let sender be the argument to
.
removeTrack
Let connection be the
object on which the method was invoked.
RTCPeerConnection
If connection.[[IsClosed]] is
true
, throw an
InvalidStateError
.
If sender was not created by
connection, throw an
InvalidAccessError
.
Let senders be the result of executing the
CollectSenders
algorithm.
If sender is not in senders (which
indicates its transceiver was stopped or removed due to
setting a session description of
"type
"), then abort these steps.
rollback
If sender.[[SenderTrack]] is null, abort these steps.
Set sender.[[SenderTrack]] to null.
Let transceiver be the
object corresponding to sender.
RTCRtpTransceiver
If transceiver.[[Direction]] is
"
", set
transceiver.[[Direction]] to
"sendrecv
".
recvonly
If transceiver.[[Direction]] is
"
", set
transceiver.[[Direction]] to
"sendonly
".
inactive
Update the negotiation-needed flag for connection.
addTransceiver
Create a new
and add it to the set
of transceivers.
RTCRtpTransceiver
Adding a transceiver will cause future calls to
to add a media description for the
corresponding transceiver, as defined in [RFC8829] (section 5.2.2.).
createOffer
The initial value of
is null.
Setting a session description may later change it to a
non-null value.
mid
The
argument can be
used to specify the number of offered simulcast encodings,
and optionally their RIDs and encoding parameters.
sendEncodings
When this method is invoked, the user agent MUST run the following steps:
Let init be the second argument.
Let streams be
init.
.
streams
Let sendEncodings be
init.
.
sendEncodings
Let direction be
init.
.
direction
If the first argument is a string, let it be kind and run the following steps:
If kind is not a legal
MediaStreamTrack
kind
,
throw a TypeError
.
Let track be null
.
If the first argument is a MediaStreamTrack
, let it
be track and let kind be
track.kind.
If connection.[[IsClosed]] is
true
, throw an
InvalidStateError
.
Verify that each
value
in sendEncodings conforms to the grammar
specified in Section 10 of [RFC8851]. If one of
the RIDs does not meet these requirements, throw a rid
TypeError
.
If any
dictionary in
sendEncodings contains a read-only
parameter other than
RTCRtpEncodingParameters
, throw
an rid
InvalidAccessError
.
Verify that the value of each
member in sendEncodings that is defined
is greater than or equal to 1.0. If one of the
scaleResolutionDownBy
values does not meet this requirement, throw a scaleResolutionDownBy
RangeError
.
Let maxN be the maximum number of total
simultaneous encodings the user agent may support for
this kind, at minimum 1
.This
should be an optimistic number since the codec to be
used is not known yet.
If sendEncodings contains any encoding
whose
attribute is defined, set any undefined
scaleResolutionDownBy
of
the other encodings to 1.0.
scaleResolutionDownBy
If the number of
stored
in sendEncodings exceeds maxN,
then trim sendEncodings from the tail
until its length is maxN.
RTCRtpEncodingParameters
scaleResolutionDownBy
attribues of sendEncodings are still
undefined, initialize each encoding's
scaleResolutionDownBy
to
2^(length of sendEncodings - encoding
index - 1)
. This results in smaller-to-larger
resolutions where the last encoding has no scaling
applied to it, e.g. 4:2:1 if the length is 3.
If the number of
now
stored in sendEncodings is RTCRtpEncodingParameters
1
,
then remove any
member
from the lone entry.
rid
RTCRtpEncodingParameters
in
sendEncodings allows the application to
subsequently set encoding parameters using
setParameters
, even when simulcast
isn't used.
Create an RTCRtpSender with track, kind, streams and sendEncodings and let sender be the result.
If sendEncodings is set, then subsequent calls
to
will be configured to send multiple
RTP encodings as defined in [RFC8829] (section 5.2.2. and section 5.2.1.). When
createOffer
is called with
a corresponding remote description that is able to
receive multiple RTP encodings as defined in
[RFC8829] (section 3.7.), the
setRemoteDescription
may send multiple RTP encodings and the
parameters retrieved via the transceiver's
RTCRtpSender
.sender
getParameters
()
will reflect the encodings negotiated.
Create an RTCRtpReceiver with kind and let receiver be the result.
Create an RTCRtpTransceiver with sender, receiver and direction, and let transceiver be the result.
Add transceiver to connection's set of transceivers.
Update the negotiation-needed flag for connection.
Return transceiver.
WebIDLdictionaryRTCRtpTransceiverInit
{RTCRtpTransceiverDirection
direction
= "sendrecv"; sequence<MediaStream>streams
= []; sequence<RTCRtpEncodingParameters
>sendEncodings
= []; };
RTCRtpTransceiverInit
Members
direction
of type RTCRtpTransceiverDirection
,
defaulting to "sendrecv
"
RTCRtpTransceiver
.
streams
of type sequence<MediaStream
>
When the remote PeerConnection's track event fires
corresponding to the
being added, these
are the streams that will be put in the event.
RTCRtpReceiver
sendEncodings
of type sequence<RTCRtpEncodingParameters
>
A sequence containing parameters for sending RTP encodings of media.
WebIDLenumRTCRtpTransceiverDirection
{ "sendrecv
", "sendonly
", "recvonly
", "inactive
", "stopped
" };
RTCRtpTransceiverDirection Enumeration description
|
|
---|---|
sendrecv
|
The 's
sender will offer to send RTP, and will send RTP
if the remote peer accepts and
sender.
() . [i].
is true for any value of i. The
's will offer to
receive RTP, and will receive RTP if the remote peer accepts.
|
sendonly
|
The 's
sender will offer to send RTP, and will send RTP
if the remote peer accepts and
sender.
() . [i].
is true for any value of i. The
's will not offer to
receive RTP, and will not receive RTP.
|
recvonly
|
The 's will not offer
to send RTP, and will not send RTP. The
's will offer to
receive RTP, and will receive RTP if the remote peer accepts.
|
inactive
|
The 's will not offer
to send RTP, and will not send RTP. The
's will not offer to
receive RTP, and will not receive RTP.
|
stopped
|
The will neither send nor receive RTP.
It will generate a zero port in the offer. In answers, its
will not offer to send RTP, and its
will not offer to receive RTP. This is a
terminal state.
|
An application can reject incoming media descriptions by setting
the transceiver's direction to either
"
" to turn off both
directions temporarily, or to
"inactive
" to reject only the
incoming side. To permanently reject an m-line in a manner that
makes it available for reuse, the application would need to call
sendonly
.RTCRtpTransceiver
stop
()
and subsequently
initiate negotiation from its end.
To process remote tracks
given an
transceiver,
direction, msids, addList,
removeList, and trackEventInits, run the
following steps:
RTCRtpTransceiver
Set the associated remote streams with transceiver.[[Receiver]], msids, addList, and removeList.
If direction is
"
" or
"sendrecv
" and
transceiver.[[FiredDirection]] is neither
"recvonly
" nor
"sendrecv
", or the previous step
increased the length of addList, process the
addition of a remote track with transceiver and
trackEventInits.
recvonly
If direction is
"
" or
"sendonly
", set
transceiver.[[Receptive]] to
inactive
false
.
If direction is
"
" or
"sendonly
", and
transceiver.[[FiredDirection]] is either
"inactive
" or
"sendrecv
", process the
removal of a remote track for the media description,
with transceiver and muteTracks.
recvonly
Set transceiver.[[FiredDirection]] to direction.
To process the addition of
a remote track given an
transceiver and trackEventInits, run the
following steps:
RTCRtpTransceiver
Let receiver be transceiver.[[Receiver]].
Let track be receiver.[[ReceiverTrack]].
Let streams be receiver.[[AssociatedRemoteMediaStreams]].
Create a new
dictionary with
receiver, track, streams and
transceiver as members and add it to
trackEventInits.
RTCTrackEventInit
To process the removal of a
remote track with an
transceiver and muteTracks, run the following
steps:
RTCRtpTransceiver
Let receiver be transceiver.[[Receiver]].
Let track be receiver.[[ReceiverTrack]].
If track.muted is false
, add
track to muteTracks.
To set the associated
remote streams given
receiver,
msids, addList, and removeList,
run the following steps:
RTCRtpReceiver
Let connection be the
object
associated with receiver.
RTCPeerConnection
For each MSID in msids, unless a MediaStream
object has previously been created with that id
for this connection, create a
MediaStream
object with that id
.
Let streams be a list of the MediaStream
objects
created for this connection with the id
s corresponding to msids.
Let track be receiver.[[ReceiverTrack]].
For each stream in receiver.[[AssociatedRemoteMediaStreams]] that is not present in streams, add stream and track as a pair to removeList.
For each stream in streams that is not present in receiver.[[AssociatedRemoteMediaStreams]], add stream and track as a pair to addList.
Set receiver.[[AssociatedRemoteMediaStreams]] to streams.
RTCRtpSender
Interface
The
interface allows an application to control how a
given RTCRtpSender
MediaStreamTrack
is encoded and transmitted to a remote
peer. When
is called on an
setParameters
object, the encoding is changed appropriately.
RTCRtpSender
To create an RTCRtpSender with a MediaStreamTrack
,
track, a string, kind, a list of
MediaStream
objects, streams, and optionally a list of
objects, sendEncodings, run
the following steps:
RTCRtpEncodingParameters
Let sender be a new
object.
RTCRtpSender
Let sender have a [[SenderTrack]] internal slot initialized to track.
Let sender have a [[SenderTransport]]
internal slot initialized to null
.
Let sender have a
[[LastStableStateSenderTransport]] internal slot
initialized to null
.
Let sender have a [[Dtmf]] internal slot
initialized to null
.
If kind is "audio"
then create an
RTCDTMFSender dtmf and set the [[Dtmf]]
internal slot to dtmf.
Let sender have an
[[AssociatedMediaStreamIds]] internal slot,
representing a list of Ids of MediaStream
objects that this
sender is to be associated with. The
[[AssociatedMediaStreamIds]] slot is used when
sender is represented in SDP as described in
[RFC8829] (section 5.2.1.).
Set sender.[[AssociatedMediaStreamIds]] to an empty set.
For each stream in streams, add stream.id to [[AssociatedMediaStreamIds]] if it's not already there.
Let sender have a [[SendEncodings]]
internal slot, representing a list of
dictionaries.
RTCRtpEncodingParameters
If sendEncodings is given as input to this algorithm,
and is non-empty, set the [[SendEncodings]] slot to
sendEncodings. Otherwise, set it to a list containing
a single
with
RTCRtpEncodingParameters
set to active
true
.
Let sender have a [[SendCodecs]] internal
slot, representing a list of
dictionaries, and initialized to an empty list.
RTCRtpCodecParameters
Let sender have a
[[LastReturnedParameters]] internal slot, which will
be used to match
and
getParameters
transactions.
setParameters
Return sender.
WebIDL[Exposed=Window] interfaceRTCRtpSender
{ readonly attribute MediaStreamTrack?track
; readonly attributeRTCDtlsTransport
?transport
; staticRTCRtpCapabilities
?getCapabilities
(DOMString kind); Promise<undefined>setParameters
(RTCRtpSendParameters
parameters);RTCRtpSendParameters
getParameters
(); Promise<undefined>replaceTrack
(MediaStreamTrack? withTrack); undefinedsetStreams
(MediaStream... streams); Promise<RTCStatsReport
>getStats
(); };
track
of type MediaStreamTrack
, readonly, nullable
The
attribute is the track that is associated with
this track
object. If RTCRtpSender
is ended, or if
the track's output is disabled, i.e. the track is disabled
and/or muted, the track
MUST send black frames
(video) and MUST NOT send (audio). In the case of video, the
RTCRtpSender
SHOULD send one black frame per second. If
RTCRtpSender
is track
null
then the
does
not send. On getting, the attribute MUST return the value of
the [[SenderTrack]] slot.
RTCRtpSender
transport
of type RTCDtlsTransport
, readonly, nullable
The
attribute is the transport over which media
from transport
is sent in the form of RTP packets. Prior to
construction of the track
object, the
RTCDtlsTransport
attribute will be null. When bundling is used,
multiple transport
objects will share one
RTCRtpSender
and will all send RTP and RTCP over the same
transport.
transport
On getting, the attribute MUST return the value of the [[SenderTransport]] slot.
getCapabilities
, static
The getCapabilities
()
method returns the most optimistic
view of the capabilities of the system for sending media of
the given kind. It does not reserve any resources, ports, or
other state but is meant to provide a way to discover the
types of capabilities of the browser including which codecs
may be supported. User agents MUST support kind
values of "audio"
and "video"
. If
the system has no capabilities corresponding to the value of
the kind argument,
returns
getCapabilities
null
.
These capabilities provide generally persistent cross-origin
information on the device and thus increases the
fingerprinting surface of the application. In
privacy-sensitive contexts, browsers can consider mitigations
such as reporting only a common subset of the capabilities.
The codec capabilities returned affect the
setCodecPreferences
()
algorithm and
what inputs it throws InvalidModificationError
on,
and should also be consistent with information revealed by
createOffer
()
and createAnswer
()
about codecs
negotiated for sending, to ensure any
privacy mitigations are effective.
setParameters
The
method updates how setParameters
is encoded
and transmitted to a remote peer.
track
When the
method is called, the user agent
MUST run the following steps:
setParameters
RTCRtpSender
object on which setParameters
is
invoked.
RTCRtpTransceiver
object associated with
sender (i.e. sender is
transceiver.[[Sender]]).
true
, return a promise rejected with a
newly created InvalidStateError
.
null
, return a promise rejected with a
newly created InvalidStateError
.
encodings
.
codecs
.
RTCRtpEncodingParameters
stored in
sender.[[SendEncodings]].
InvalidModificationError
:
encodings.length
is
different from N.
Verify that each encoding in encodings has
a
member whose value is greater than or equal to 1.0. If one of the
scaleResolutionDownBy
values does not meet this requirement, return a
promise rejected with a newly created scaleResolutionDownBy
RangeError
.
null
.
encodings
.
undefined
.
RTCError
whose
errorDetail
is set to
"hardware-encoder-not-available
"
and abort these steps.
RTCError
whose
errorDetail
is set to
"hardware-encoder-error
" and
abort these steps.
OperationError
.
does not cause SDP renegotiation and can
only be used to change what the media stack is sending or
receiving within the envelope negotiated by Offer/Answer. The
attributes in the setParameters
dictionary are
designed to not enable this, so attributes like
RTCRtpSendParameters
that cannot be changed are
read-only. Other things, like bitrate, are controlled using
limits such as cname
, where
the user agent needs to ensure it does not exceed the maximum
bitrate specified by maxBitrate
,
while at the same time making sure it satisfies constraints
on bitrate specified in other places such as the SDP.
maxBitrate
getParameters
The getParameters
()
method returns the
object's current parameters for how RTCRtpSender
is encoded and
transmitted to a remote track
.
RTCRtpReceiver
When
is called, the user agent MUST run the
following steps:
getParameters
Let sender be the
object on
which the getter was invoked.
RTCRtpSender
If sender.[[LastReturnedParameters]]
is not null
, return
sender.[[LastReturnedParameters]], and
abort these steps.
Let result be a new
dictionary constructed as follows:
RTCRtpSendParameters
transactionId
is set to a new
unique identifier.
encodings
is set to the value of
the [[SendEncodings]] internal slot.
headerExtensions
sequence is
populated based on the header extensions that have been
negotiated for sending.
codecs
is set to the value of the
[[SendCodecs]] internal slot.
rtcp
.cname
is
set to the CNAME of the associated RTCPeerConnection
.
rtcp
.reducedSize
is set to true
if reduced-size RTCP has been
negotiated for sending, and false
otherwise.
Set sender.[[LastReturnedParameters]] to result.
Queue a task that sets
sender.[[LastReturnedParameters]] to
null
.
Return result.
may be used with getParameters
to
change the parameters in the following way:
setParameters
async function updateParameters() {
try {
const params = sender.getParameters();
// ... make changes to parameters
params.encodings[0].active = false;
await sender.setParameters(params);
} catch (err) {
console.error(err);
}
}
After a completed call to
, subsequent calls
to setParameters
will return the modified set of
parameters.
getParameters
replaceTrack
Attempts to replace the
's current RTCRtpSender
with another track provided (or with a track
null
track), without renegotiation.
When the
method is invoked, the user agent
MUST run the following steps:
replaceTrack
Let sender be the
object on
which RTCRtpSender
is invoked.
replaceTrack
Let transceiver be the
object associated with sender.
RTCRtpTransceiver
Let connection be the
object associated with sender.
RTCPeerConnection
Let withTrack be the argument to this method.
If withTrack is non-null and
withTrack.kind
differs from the
transceiver kind of transceiver, return
a promise rejected with a newly created TypeError
.
Return the result of chaining the following steps to connection's operations chain:
If transceiver.[[Stopped]] is
true
, return a promise rejected
with a newly created
InvalidStateError
.
Let p be a new promise.
Let sending be true
if
transceiver.[[CurrentDirection]]
is "
" or
"sendrecv
", and
sendonly
false
otherwise.
Run the following steps in parallel:
If sending is true
, and
withTrack is null
, have
the sender stop sending.
If sending is true
, and
withTrack is not null
,
determine if withTrack can be sent
immediately by the sender without violating the
sender's already-negotiated envelope, and if it
cannot, then reject p with a
newly created
InvalidModificationError
, and abort these
steps.
If sending is true
, and
withTrack is not null
,
have the sender switch seamlessly to transmitting
withTrack instead of the sender's
existing track.
Queue a task that runs the following steps:
If connection.[[IsClosed]]
is true
, abort these steps.
Set sender.[[SenderTrack]] to withTrack.
Resolve p with
undefined
.
Return p.
Changing dimensions and/or frame rates might not require negotiation. Cases that may require negotiation include:
setStreams
Sets the MediaStream
s to be associated with this sender's
track.
When the
method is invoked, the user agent
MUST run the following steps:
setStreams
Let sender be the
object on
which this method was invoked.
RTCRtpSender
Let connection be the
object on which this method was invoked.
RTCPeerConnection
If connection.[[IsClosed]] is
true
, throw an
InvalidStateError
.
Let streams be a list of MediaStream
objects constructed from the method's arguments, or an
empty list if the method was called without arguments.
Set sender.[[AssociatedMediaStreamIds]] to an empty set.
For each stream in streams, add stream.id to [[AssociatedMediaStreamIds]] if it's not already there.
Update the negotiation-needed flag for connection.
getStats
Gathers stats for this sender only and reports the result asynchronously.
When the getStats
()
method is invoked, the user agent
MUST run the following steps:
Let selector be the
object on
which the method was invoked.
RTCRtpSender
Let p be a new promise, and run the following steps in parallel:
Gather the stats indicated by selector according to the stats selection algorithm.
Resolve p with the resulting
object, containing the gathered
stats.
RTCStatsReport
Return p.
RTCRtpParameters
Dictionary
WebIDLdictionaryRTCRtpParameters
{ required sequence<RTCRtpHeaderExtensionParameters
>headerExtensions
; requiredRTCRtcpParameters
rtcp
; required sequence<RTCRtpCodecParameters
>codecs
; };
RTCRtpParameters
Members
headerExtensions
of type sequence<RTCRtpHeaderExtensionParameters
>,
required
A sequence containing parameters for RTP header extensions. Read-only parameter.
rtcp
of type RTCRtcpParameters
, required
Parameters used for RTCP. Read-only parameter.
codecs
of type sequence<RTCRtpCodecParameters
>,
required
A sequence containing the media codecs that an
will choose from, as well as entries for
RTX, RED and FEC mechanisms. Corresponding to each media
codec where retransmission via RTX is enabled, there will
be an entry in RTCRtpSender
with a
codecs
attribute indicating
retransmission via mimeType
audio/rtx
or
video/rtx
, and an
attribute (providing
the "apt" and "rtx-time" parameters). Read-only
parameter.
sdpFmtpLine
RTCRtpSendParameters
Dictionary
WebIDL dictionaryRTCRtpSendParameters
:RTCRtpParameters
{ required DOMStringtransactionId
; required sequence<RTCRtpEncodingParameters
>encodings
; };
RTCRtpSendParameters
Members
transactionId
of type DOMString, required
A unique identifier for the last set of parameters applied.
Ensures that
can only be
called based on a previous setParameters
,
and that there are no intervening changes. Read-only
parameter.
getParameters
encodings
of type sequence<RTCRtpEncodingParameters
>,
required
A sequence containing parameters for RTP encodings of media.
RTCRtpReceiveParameters
Dictionary
WebIDL dictionaryRTCRtpReceiveParameters
:RTCRtpParameters
{ };
RTCRtpCodingParameters
Dictionary
WebIDLdictionaryRTCRtpCodingParameters
{ DOMStringrid
; };
RTCRtpCodingParameters
Members
rid
of type DOMString
If set, this RTP encoding will be sent with the RID header
extension as defined by [RFC8829] (section 5.2.1.). The RID is not
modifiable via
. It can only
be set or modified in setParameters
on the sending side. Read-only parameter.
addTransceiver
RTCRtpDecodingParameters
Dictionary
WebIDLdictionaryRTCRtpDecodingParameters
:RTCRtpCodingParameters
{};
RTCRtpEncodingParameters
Dictionary
WebIDL dictionaryRTCRtpEncodingParameters
:RTCRtpCodingParameters
{ booleanactive
= true; unsigned longmaxBitrate
; doublescaleResolutionDownBy
; };
RTCRtpEncodingParameters
Members
active
of type boolean, defaulting to
true
Indicates that this encoding is actively being sent.
Setting it to false
causes this encoding to no
longer be sent. Setting it to true
causes this
encoding to be sent. Since setting the value to
false
does not cause the SSRC to be removed,
an RTCP BYE is not sent.
maxBitrate
of type unsigned long
When present, indicates the maximum bitrate that can be
used to send this encoding. The user agent is free to
allocate bandwidth between the encodings, as long as the
value is not exceeded. The encoding may also
be further constrained by other limits (such as
per-transport or per-session bandwidth limits) below the
maximum specified here. maxBitrate
is computed the same
way as the Transport Independent Application Specific
Maximum (TIAS) bandwidth defined in [RFC3890] Section
6.2.2, which is the maximum bandwidth needed without
counting IP or other transport layers like TCP or UDP. The
unit of maxBitrate
is bits per second.
maxBitrate
How the bitrate is achieved is media and encoding dependent. For video, a frame will always be sent as fast as possible, but frames may be dropped until bitrate is low enough. Thus, even a bitrate of zero will allow sending one frame. For audio, it might be necessary to stop playing if the bitrate does not allow the chosen encoding enough bandwidth to be sent.
scaleResolutionDownBy
of type
double
This member is only present if the sender's kind
is "video"
. The video's
resolution will be scaled down in each dimension by the
given value before sending. For example, if the value is
2.0, the video will be scaled down by a factor of 2 in each
dimension, resulting in sending a video of one quarter the
size. If the value is 1.0, the video will not be affected.
The value must be greater than or equal to 1.0. By default,
scaling is applied by a factor of two to the power of the
layer's number, in order of smaller to higher resolutions,
e.g. 4:2:1. If there is only one layer, the sender will by
default not apply any scaling, (i.e.
will be
1.0).
scaleResolutionDownBy
RTCRtcpParameters
Dictionary
WebIDLdictionaryRTCRtcpParameters
{ DOMStringcname
; booleanreducedSize
; };
RTCRtcpParameters
Members
cname
of type DOMString
The Canonical Name (CNAME) used by RTCP (e.g. in SDES messages). Read-only parameter.
reducedSize
of type boolean
Whether reduced size RTCP [RFC5506] is configured (if true) or compound RTCP as specified in [RFC3550] (if false). Read-only parameter.
RTCRtpHeaderExtensionParameters
Dictionary
WebIDLdictionaryRTCRtpHeaderExtensionParameters
{ required DOMStringuri
; required unsigned shortid
; booleanencrypted
= false; };
RTCRtpHeaderExtensionParameters
Members
uri
of type DOMString, required
The URI of the RTP header extension, as defined in [RFC5285]. Read-only parameter.
id
of type unsigned short, required
The value put in the RTP packet to identify the header extension. Read-only parameter.
encrypted
of type boolean
Whether the header extension is encrypted or not. Read-only parameter.
The
dictionary enables an
application to determine whether a header extension is configured
for use within an RTCRtpHeaderExtensionParameters
or RTCRtpSender
. For an
RTCRtpReceiver
transceiver, an application can
determine the "direction" parameter (defined in Section 5 of
[RFC5285]) of a header extension as follows without having to
parse SDP:
RTCRtpTransceiver
sender
.getParameters
()
.headerExtensions
.
receiver
.getParameters
()
.headerExtensions
.
sender
.getParameters
()
.headerExtensions
and
transceiver.receiver
.getParameters
()
.headerExtensions
.
sender
.getParameters
()
.headerExtensions
nor
transceiver.receiver
.getParameters
()
.headerExtensions
.
RTCRtpCodecParameters
Dictionary
WebIDLdictionaryRTCRtpCodecParameters
{ required octetpayloadType
; required DOMStringmimeType
; required unsigned longclockRate
; unsigned shortchannels
; DOMStringsdpFmtpLine
; };
RTCRtpCodecParameters
Members
payloadType
of type octet, required
The RTP payload type used to identify this codec. Read-only parameter.
mimeType
of type DOMString, required
The codec MIME media type/subtype. Valid media types and subtypes are listed in [IANA-RTP-2]. Read-only parameter.
clockRate
of type unsigned long, required
The codec clock rate expressed in Hertz. Read-only parameter.
channels
of type unsigned short
When present, indicates the number of channels (mono=1, stereo=2). Read-only parameter.
sdpFmtpLine
of type DOMString
The "format specific parameters" field from the
a=fmtp
line in the SDP
corresponding to the codec, if one exists, as defined by
[RFC8829] (section 5.8.). For an
, these parameters come from the remote
description, and for an RTCRtpSender
, they come from
the local description. Read-only parameter.
RTCRtpReceiver
RTCRtpCapabilities
Dictionary
WebIDLdictionaryRTCRtpCapabilities
{ required sequence<RTCRtpCodecCapability
>codecs
; required sequence<RTCRtpHeaderExtensionCapability
>headerExtensions
; };
RTCRtpCapabilities
Members
codecs
of type sequence<RTCRtpCodecCapability
>,
required
Supported media codecs as well as entries for RTX, RED and
FEC mechanisms. There will only be a single entry in
for retransmission via RTX, with
codecs
not present.
sdpFmtpLine
headerExtensions
of type sequence<RTCRtpHeaderExtensionCapability
>,
required
Supported RTP header extensions.
RTCRtpCodecCapability
Dictionary
WebIDLdictionaryRTCRtpCodecCapability
{ required DOMStringmimeType
; required unsigned longclockRate
; unsigned shortchannels
; DOMStringsdpFmtpLine
; };
RTCRtpCodecCapability
Members
The
dictionary provides information
about codec capabilities. Only capability combinations that
would utilize distinct payload types in a generated SDP offer
are provided. For example:
RTCRtpCodecCapability
mimeType
of type DOMString, required
The codec MIME media type/subtype. Valid media types and subtypes are listed in [IANA-RTP-2].
clockRate
of type unsigned long, required
The codec clock rate expressed in Hertz.
channels
of type unsigned short
If present, indicates the maximum number of channels (mono=1, stereo=2).
sdpFmtpLine
of type DOMString
The "format specific parameters" field from the
a=fmtp
line in the SDP
corresponding to the codec, if one exists.
RTCRtpHeaderExtensionCapability
Dictionary
WebIDLdictionaryRTCRtpHeaderExtensionCapability
{ DOMStringuri
; };
RTCRtpHeaderExtensionCapability
Members
uri
of type DOMString
The URI of the RTP header extension, as defined in [RFC5285].
RTCRtpReceiver
Interface
The
interface allows an application to inspect the
receipt of a RTCRtpReceiver
MediaStreamTrack
.
To create an RTCRtpReceiver with a string, kind, run the following steps:
Let receiver be a new
object.
RTCRtpReceiver
Let track be a new MediaStreamTrack
object
[GETUSERMEDIA]. The source of track is a
remote source provided by receiver. Note
that the track.id
is
generated by the user agent and does not map to any track
IDs on the remote side.
Initialize track.kind to kind.
Initialize track.label to the result of concatenating
the string "remote "
with kind.
Initialize track.readyState to live
.
Initialize track.muted to true
. See the
MediaStreamTrack
section about how the muted
attribute
reflects if a MediaStreamTrack
is receiving media data or
not.
Let receiver have a [[ReceiverTrack]] internal slot initialized to track.
Let receiver have a [[ReceiverTransport]]
internal slot initialized to null
.
Let receiver have a
[[LastStableStateReceiverTransport]] internal slot
initialized to null
.
Let receiver have an
[[AssociatedRemoteMediaStreams]] internal slot,
representing a list of MediaStream
objects that the
MediaStreamTrack
object of this receiver is associated with,
and initialized to an empty list.
Let receiver have a [[LastStableStateAssociatedRemoteMediaStreams]] internal slot and initialize it to an empty list.
Let receiver have a [[ReceiveCodecs]]
internal slot, representing a list of
dictionaries, and initialized to an empty list.
RTCRtpCodecParameters
Let receiver have a [[LastStableStateReceiveCodecs]] internal slot and initialize it to an empty list.
Return receiver.
WebIDL[Exposed=Window] interfaceRTCRtpReceiver
{ readonly attribute MediaStreamTracktrack
; readonly attributeRTCDtlsTransport
?transport
; staticRTCRtpCapabilities
?getCapabilities
(DOMString kind);RTCRtpReceiveParameters
getParameters
(); sequence<RTCRtpContributingSource
>getContributingSources
(); sequence<RTCRtpSynchronizationSource
>getSynchronizationSources
(); Promise<RTCStatsReport
>getStats
(); };
track
of type
MediaStreamTrack
, readonly
The
attribute is the track that is associated with
this track
object receiver.
RTCRtpReceiver
Note that
.track
stop()
is final,
although clones are not affected. Since
receiver.
.track
stop()
does not implicitly stop receiver, Receiver
Reports continue to be sent. On getting, the attribute MUST
return the value of the [[ReceiverTrack]] slot.
transport
of type RTCDtlsTransport
, readonly, nullable
The
attribute is the transport over which media
for the receiver's transport
is received in
the form of RTP packets. Prior to construction of the
track
object, the RTCDtlsTransport
attribute will
be transport
null
. When bundling is used, multiple
objects will share one RTCRtpReceiver
and
will all receive RTP and RTCP over the same transport.
transport
On getting, the attribute MUST return the value of the [[ReceiverTransport]] slot.
getCapabilities
, static
The getCapabilities
()
method returns the most optimistic
view of the capabilities of the system for receiving media of
the given kind. It does not reserve any resources, ports, or
other state but is meant to provide a way to discover the
types of capabilities of the browser including which codecs
may be supported. User agents MUST support kind
values of "audio"
and "video"
. If
the system has no capabilities corresponding to the value of
the kind argument,
returns
getCapabilities
null
.
These capabilities provide generally persistent cross-origin
information on the device and thus increases the
fingerprinting surface of the application. In
privacy-sensitive contexts, browsers can consider mitigations
such as reporting only a common subset of the capabilities.
The codec capabilities returned affect the
setCodecPreferences
()
algorithm and
what inputs it throws InvalidModificationError
on,
and should also be consistent with information revealed by
createOffer
()
and createAnswer
()
about codecs
negotiated for reception, to ensure any
privacy mitigations are effective.
getParameters
The getParameters
()
method returns the
object's current parameters for how RTCRtpReceiver
is decoded.
track
When
is called, the
getParameters
dictionary is constructed as
follows:
RTCRtpReceiveParameters
headerExtensions
sequence is populated
based on the header extensions that the receiver is currently
prepared to receive.
is set to the value of the
[[ReceiveCodecs]] internal slot.
codecs
getParameters
. But if the
remote endpoint only answers with two, the absent codec
will no longer be returned by getParameters
as the
receiver no longer needs to be prepared to receive it.
rtcp
.reducedSize
is set to true
if the receiver is currently
prepared to receive reduced-size RTCP packets, and
false
otherwise.
rtcp
.cname
is left
out.
getContributingSources
Returns an
for each unique CSRC
identifier received by this RTCRtpContributingSource
in the last 10
seconds, in descending RTCRtpReceiver
order.
timestamp
getSynchronizationSources
Returns an
for each unique
SSRC identifier received by this RTCRtpSynchronizationSource
in the
last 10 seconds, in descending
RTCRtpReceiver
order.
timestamp
getStats
Gathers stats for this receiver only and reports the result asynchronously.
When the getStats
()
method is invoked, the user agent
MUST run the following steps:
Let selector be the
object
on which the method was invoked.
RTCRtpReceiver
Let p be a new promise, and run the following steps in parallel:
Gather the stats indicated by selector according to the stats selection algorithm.
Resolve p with the resulting
object, containing the gathered
stats.
RTCStatsReport
Return p.
The RTCRtpContributingSource
and
RTCRtpSynchronizationSource
dictionaries contain
information about a given contributing source (CSRC) or
synchronization source (SSRC) respectively. When an audio or video
frame from one or more RTP packets is delivered to the
's RTCRtpReceiver
MediaStreamTrack
, the user agent MUST queue
a task to update the relevant information for the
and RTCRtpContributingSource
dictionaries based on the content of those packets. The information
relevant to the RTCRtpSynchronizationSource
dictionary
corresponding to the SSRC identifier, is updated each time, and if an
RTP packet contains CSRC identifiers, then the information relevant
to the RTCRtpSynchronizationSource
dictionaries corresponding to
those CSRC identifiers is also updated. The user agent MUST process
RTP packets in order of ascending RTP timestamps. The user agent MUST
keep information from RTP packets delivered to the
RTCRtpContributingSource
's RTCRtpReceiver
MediaStreamTrack
in the previous 10 seconds.
MediaStreamTrack
is not attached to any sink for
playout, getSynchronizationSources
and
getContributingSources
returns up-to-date
information as long as the track is not ended; sinks are not a
prerequisite for decoding RTP packets.
RTCRtpSynchronizationSource
and
RTCRtpContributingSource
dictionaries for a particular
RTCRtpReceiver
contain information from a single point in the RTP
stream.
WebIDLdictionaryRTCRtpContributingSource
{ required DOMHighResTimeStamptimestamp
; required unsigned longsource
; doubleaudioLevel
; required unsigned longrtpTimestamp
; };
timestamp
of type
DOMHighResTimeStamp
, required
The
indicating the most recent time a frame
from an RTP packet, originating from this source, was
delivered to the timestamp
's RTCRtpReceiver
MediaStreamTrack
.
The
is defined as timestamp
Performance
.timeOrigin
+
Performance
.now
()
at that time.
source
of type unsigned long, required
The CSRC or SSRC identifier of the contributing or synchronization source.
audioLevel
of type double
Only present for audio receivers. This is a value between 0..1 (linear), where 1.0 represents 0 dBov, 0 represents silence, and 0.5 represents approximately 6 dBSPL change in the sound pressure level from 0 dBov.
For CSRCs, this MUST be converted from the level value defined in [RFC6465] if the RFC 6465 header extension is present, otherwise this member MUST be absent.
For SSRCs, this MUST be converted from the level value defined in [RFC6464]. If the RFC 6464 header extension is not present in the received packets (such as if the other endpoint is not a user agent or is a legacy endpoint), this value SHOULD be absent.
Both RFCs define the level as an integral value from 0 to 127 representing the audio level in negative decibels relative to the loudest signal that the system could possibly encode. Thus, 0 represents the loudest signal the system could possibly encode, and 127 represents silence.
To convert these values to the linear 0..1 range, a value of
127 is converted to 0, and all other values are converted
using the equation: 10^(-rfc_level/20)
.
rtpTimestamp
of type unsigned long, required
The last RTP timestamp, as defined in [RFC3550] Section 5.1, of the media played out at timestamp.
WebIDL dictionaryRTCRtpSynchronizationSource
:RTCRtpContributingSource
{ };
The
dictionary is expected to serve as an extension point for the specification to surface data only available in SSRCs.RTCRtpSynchronizationSource
RTCRtpTransceiver
Interface
The
interface represents a combination of an
RTCRtpTransceiver
and an RTCRtpSender
that share a common media stream "identification-tag". As defined in [RFC8829] (section 3.4.1.), an RTCRtpReceiver
is said
to be associated with a media description if its
"mid" property is non-null and matches a media stream
"identification-tag" in the media description; otherwise it
is said to be disassociated with that media description.
RTCRtpTransceiver
A
may become associated with a new pending
description in RFC8829 while still being disassociated with the
current description. This may happen in check if negotiation is
needed.
RTCRtpTransceiver
The transceiver kind of an
is
defined by the kind of the associated RTCRtpTransceiver
's
RTCRtpReceiver
MediaStreamTrack
object.
To create an RTCRtpTransceiver with an
object, receiver, RTCRtpReceiver
object,
sender, and an RTCRtpSender
value,
direction, run the following steps:
RTCRtpTransceiverDirection
Let transceiver be a new
object.
RTCRtpTransceiver
Let transceiver have a [[Sender]] internal slot, initialized to sender.
Let transceiver have a [[Receiver]] internal slot, initialized to receiver.
Let transceiver have a [[Stopping]]
internal slot, initialized to false
.
Let transceiver have a [[Stopped]]
internal slot, initialized to false
.
Let transceiver have a [[Direction]] internal slot, initialized to direction.
Let transceiver have a [[Receptive]]
internal slot, initialized to false
.
Let transceiver have a
[[CurrentDirection]] internal slot, initialized to
null
.
Let transceiver have a [[FiredDirection]]
internal slot, initialized to null
.
Let transceiver have a [[PreferredCodecs]] internal slot, initialized to an empty list.
Let transceiver have a [[JsepMid]]
internal slot, initialized to null
. This is the
"RtpTransceiver mid property" defined in [RFC8829] (section 5.2.1. and section 5.3.1.), and is only
modified there.
Let transceiver have a [[Mid]] internal
slot, initialized to null
.
Return transceiver.
RTCDtlsTransport
and RTCIceTransport
objects. This will only
occur as part of the process of setting a session description.
WebIDL[Exposed=Window] interfaceRTCRtpTransceiver
{ readonly attribute DOMString?mid
; [SameObject] readonly attributeRTCRtpSender
sender
; [SameObject] readonly attributeRTCRtpReceiver
receiver
; attributeRTCRtpTransceiverDirection
direction
; readonly attributeRTCRtpTransceiverDirection
?currentDirection
; undefinedstop
(); undefinedsetCodecPreferences
(sequence<RTCRtpCodecCapability
> codecs); };
mid
of type DOMString, readonly, nullable
The
attribute is the media stream
"identification-tag" negotiated and present in the local
and remote descriptions. On getting, the attribute MUST
return the value of the [[Mid]] slot.
mid
sender
of type RTCRtpSender
, readonly
The
attribute exposes the sender
corresponding to the RTP media that may be sent with mid =
[[Mid]]. On getting, the attribute MUST return the
value of the [[Sender]] slot.
RTCRtpSender
receiver
of type RTCRtpReceiver
, readonly
The
attribute is the receiver
corresponding to the RTP media that may be received with mid
= [[Mid]]. On getting the attribute MUST return the
value of the [[Receiver]] slot.
RTCRtpReceiver
direction
of type RTCRtpTransceiverDirection
As defined in [RFC8829] (section 4.2.4.), the
direction attribute indicates the preferred
direction of this transceiver, which will be used in calls to
and
createOffer
. An update of
directionality does not take effect immediately. Instead,
future calls to createAnswer
and
createOffer
mark the corresponding media description as createAnswer
sendrecv
,
sendonly
, recvonly
or inactive
as
defined in [RFC8829] (section 5.2.2. and section 5.3.2.)
On getting, the user agent MUST run the following steps:
Let transceiver be the
object on which the getter is invoked.
RTCRtpTransceiver
If transceiver.[[Stopping]] is
true
, return
"
".
stopped
Otherwise, return the value of the [[Direction]] slot.
On setting, the user agent MUST run the following steps:
Let transceiver be the
object on which the setter is invoked.
RTCRtpTransceiver
Let connection be the
object associated with transceiver.
RTCPeerConnection
If transceiver.[[Stopping]] is
true
, throw an
InvalidStateError
.
Let newDirection be the argument to the setter.
If newDirection is equal to transceiver.[[Direction]], abort these steps.
Set transceiver.[[Direction]] to newDirection.
Update the negotiation-needed flag for connection.
currentDirection
of type RTCRtpTransceiverDirection
, readonly,
nullable
As defined in [RFC8829] (section 4.2.5.), the
currentDirection attribute indicates the current
direction negotiated for this transceiver. The value of
currentDirection is independent of the value of
.RTCRtpEncodingParameters
since one cannot be deduced from the other. If this
transceiver has never been represented in an offer/answer
exchange, the value is active
null
. If the transceiver
is stopped
, the value is
"
".
stopped
On getting, the user agent MUST run the following steps:
Let transceiver be the
object on which the getter is invoked.
RTCRtpTransceiver
If transceiver.[[Stopped]] is
true
, return
"
".
stopped
Otherwise, return the value of the [[CurrentDirection]] slot.
stop
Irreversibly marks the transceiver as stopping
, unless it
is already stopped
. This will immediately cause the
transceiver's sender to no longer send, and its receiver to
no longer receive. Calling stop
()
also updates the negotiation-needed flag for the
's associated
RTCRtpTransceiver
.
RTCPeerConnection
A stopping transceiver will cause future calls to
to generate a zero port in
the media description for the corresponding
transceiver, as defined in [RFC8829] (section 4.2.1.) (The user agent MUST treat a
createOffer
stopping
transceiver as stopped
for the purposes of
RFC8829 only in this case). However, to avoid problems with
[RFC8843], a transceiver that is stopping
, but not
stopped
, will not affect
.
createAnswer
A stopped transceiver will cause future calls to
or
createOffer
to generate a zero port in
the media description for the corresponding
transceiver, as defined in [RFC8829] (section 4.2.1.).
createAnswer
The transceiver will remain in the stopping
state, unless
it becomes stopped
by
processing a
rejected m-line in a remote offer or answer.
setRemoteDescription
A transceiver that is stopping
but not stopped
will
always need negotiation. In practice, this means that calling
stop
()
on a transceiver will cause the transceiver to
become stopped
eventually, provided negotiation is
allowed to complete on both ends.
When the
method is invoked, the user agent MUST run
the following steps:
stop
Let transceiver be the
object on which the method is invoked.
RTCRtpTransceiver
Let connection be the
object associated with transceiver.
RTCPeerConnection
If connection.[[IsClosed]] is
true
, throw an
InvalidStateError
.
If transceiver.[[Stopping]] is
true
, abort these steps.
Stop sending and receiving with transceiver.
Update the negotiation-needed flag for connection.
The stop sending and receiving algorithm given a
transceiver and, optionally, a
disappear boolean defaulting to
false
, is as follows:
Let sender be transceiver.[[Sender]].
Let receiver be transceiver.[[Receiver]].
Stop sending media with sender.
Send an RTCP BYE for each RTP stream that was being sent by sender, as specified in [RFC3550].
Stop receiving media with receiver.
If disappear is false
, execute
the steps for
receiver.[[ReceiverTrack]] to be
ended. This
fires an event.
Set transceiver.[[Direction]] to
"
".
inactive
Set transceiver.[[Stopping]] to
true
.
The stop the RTCRtpTransceiver algorithm given a
transceiver and, optionally, a
disappear boolean defaulting to
false
, is as follows:
If transceiver.[[Stopping]] is
false
, stop sending and receiving with
transceiver and disappear.
Set transceiver.[[Stopped]] to
true
.
Set transceiver.[[Receptive]] to
false
.
Set transceiver.[[CurrentDirection]]
to null
.
setCodecPreferences
The
method overrides the default
codec preferences used by the user agent. When
generating a session description using either
setCodecPreferences
or
createOffer
, the user agent
MUST use the indicated codecs, in the order specified in the
codecs argument, for the media section
corresponding to this createAnswer
.
RTCRtpTransceiver
This method allows applications to disable the negotiation of specific codecs (including RTX/RED/FEC). It also allows an application to cause a remote peer to prefer the codec that appears first in the list for sending.
Codec preferences remain in effect for all calls to
and
createOffer
that include this
createAnswer
until this method is called again.
Setting codecs to an empty sequence resets codec
preferences to any default value.
RTCRtpTransceiver
Codecs have their payload types listed under each m= section in the SDP, defining the mapping between payload types and codecs. These payload types are referenced by the m=video or m=audio lines in the order of preference, and codecs that are not negotiated do not appear in this list as defined in section 5.2.1 of [RFC8829]. A previously negotiated codec that is subsequently removed disappears from the m=video or m=audio line, and while its codec payload type is not to be reused in future offers or answers, its payload type may also be removed from the mapping of payload types in the SDP.
The codecs sequence passed into
can only contain codecs that are
returned by
setCodecPreferences
.RTCRtpSender
(kind)
or
getCapabilities
.RTCRtpReceiver
(kind),
where kind is the kind of the
getCapabilities
on which the method is called.
Additionally, the RTCRtpTransceiver
dictionary
members cannot be modified. If codecs does not
fulfill these requirements, the user agent MUST throw an RTCRtpCodecCapability
InvalidModificationError
.
Due to a recommendation in [SDP], calls to
SHOULD use only the common
subset of the codec preferences and the codecs that appear in
the offer. For example, if codec preferences are "C, B, A",
but only codecs "A, B" were offered, the answer should only
contain codecs "B, A". However, [RFC8829] (section 5.3.1.) allows adding codecs that
were not in the offer, so implementations can behave
differently.
createAnswer
When setCodecPreferences
()
in invoked, the user
agent MUST run the following steps:
Let transceiver be the
object this method was invoked on.
RTCRtpTransceiver
Let codecs be the first argument.
If codecs is an empty list, set transceiver.[[PreferredCodecs]] to codecs and abort these steps.
Remove any duplicate values in codecs. Start at the back of the list such that the priority of the codecs is maintained; the index of the first occurrence of a codec within the list is the same before and after this step.
Let kind be the transceiver's transceiver kind.
If the intersection between codecs and
.RTCRtpSender
(kind).getCapabilities
or the intersection between codecs and
codecs
.RTCRtpReceiver
(kind).getCapabilities
only contains RTX, RED or FEC codecs or is an empty set,
throw codecs
InvalidModificationError
. This ensures that we
always have something to offer, regardless of
transceiver.
.
direction
Let codecCapabilities be the union of
.RTCRtpSender
(kind).getCapabilities
and
codecs
.RTCRtpReceiver
(kind).getCapabilities
.
codecs
For each codec in codecs,
InvalidModificationError
.
Set transceiver.[[PreferredCodecs]] to codecs.
If set, the offerer's codec preferences will decide the order of the codecs in the offer. If the answerer does not have any codec preferences then the same order will be used in the answer. However, if the answerer also has codec preferences, these preferences override the order in the answer. In this case, the offerer's preferences would affect which codecs were on offer but not the final order.
Simulcast functionality is provided via the
method of the
addTransceiver
object and the RTCPeerConnection
method of the setParameters
object.
RTCRtpSender
The
method establishes the
simulcast envelope which includes the maximum number of
simulcast streams that can be sent, as well as the ordering of the
addTransceiver
. While characteristics of
individual simulcast streams can be modified using the
encodings
method, the simulcast envelope
cannot be changed. One of the implications of this model is that
the setParameters
addTrack
()
method cannot provide
simulcast functionality since it does not take
as an argument, and
therefore cannot configure an sendEncodings
to send
simulcast.
RTCRtpTransceiver
Another implication is that the answerer cannot set the simulcast envelope directly. Upon calling the
method of the
setRemoteDescription
object, the simulcast envelope is
configured on the RTCPeerConnection
to contain the layers
described by the specified session description. Once the
envelope is determined, layers cannot be removed. They can be
marked as inactive by setting the
RTCRtpTransceiver
member to active
false
effectively disabling the layer.
While
cannot modify the simulcast
envelope, it is still possible to control the number of streams
that are sent and the characteristics of those streams. Using
setParameters
, simulcast streams can be made
inactive by setting the setParameters
member
to active
false
, or can be reactivated by setting the
member to active
true
.
Using
, stream characteristics can be
changed by modifying attributes such as
setParameters
.
maxBitrate
Simulcast is frequently used to send multiple encodings to an SFU,
which will then forward one of the simulcast streams to the end
user. The user agent is therefore expected to allocate bandwidth
between encodings in such a way that all simulcast streams are
usable on their own; for instance, if two simulcast streams have
the same
, one would expect
to see a similar bitrate on both streams. If bandwidth does not
permit all simulcast streams to be sent in an usable form, the user
agent is expected to stop sending some of the simulcast streams.
maxBitrate
As defined in [RFC8829] (section 3.7.), an
offer from a user-agent will only contain a "send" description and
no "recv" description on the a=simulcast
line. Alternatives and restrictions (described in
[RFC8853]) are not supported.
This specification does not define how to configure reception of
multiple RTP encodings using
,
createOffer
or
createAnswer
. However when
addTransceiver
is called with a
corresponding remote description that is able to send multiple RTP
encodings as defined in [RFC8829], and the browser supports
receiving multiple RTP encodings, the setRemoteDescription
may
receive multiple RTP encodings and the parameters retrieved via the
transceiver's
RTCRtpReceiver
.receiver
getParameters
()
will reflect the encodings negotiated.
An
can receive multiple RTP streams in a
scenario where a Selective Forwarding Unit (SFU) switches between
simulcast streams it receives from user agents. If the SFU does not
rewrite RTP headers so as to arrange the switched streams into a
single RTP stream prior to forwarding, the RTCRtpReceiver
will
receive packets from distinct RTP streams, each with their own SSRC
and sequence number space. While the SFU may only forward a single
RTP stream at any given time, packets from multiple RTP streams can
become intermingled at the receiver due to reordering. An
RTCRtpReceiver
equipped to receive multiple RTP streams will
therefore need to be able to correctly order the received packets,
recognize potential loss events and react to them. Correct
operation in this scenario is non-trivial and therefore is optional
for implementations of this specification.
RTCRtpReceiver
This section is non-normative.
Examples of simulcast scenarios implemented with encoding parameters:
// Example of 3-layer spatial simulcast with all but the lowest resolution layer disabled
var encodings = [
{rid: 'q', active: true, scaleResolutionDownBy: 4.0}
{rid: 'h', active: false, scaleResolutionDownBy: 2.0},
{rid: 'f', active: false},
];
This section is non-normative.
Together, the
attribute and the
direction
method enable developers to implement
"hold" scenarios.
replaceTrack
To send music to a peer and cease rendering received audio (music-on-hold):
async function playMusicOnHold() {
try {
// Assume we have an audio transceiver and a music track named musicTrack
await audio.sender.replaceTrack(musicTrack);
// Mute received audio
audio.receiver.track.enabled = false;
// Set the direction to send-only (requires negotiation)
audio.direction = 'sendonly';
} catch (err) {
console.error(err);
}
}
To respond to a remote peer's "sendonly" offer:
async function handleSendonlyOffer() {
try {
// Apply the sendonly offer first,
// to ensure the receiver is ready for ICE candidates.
await pc.setRemoteDescription(sendonlyOffer);
// Stop sending audio
await audio.sender.replaceTrack(null);
// Align our direction to avoid further negotiation
audio.direction = 'recvonly';
// Call createAnswer and send a recvonly answer
await doAnswer();
} catch (err) {
// handle signaling error
}
}
To stop sending music and send audio captured from a microphone, as well to render received audio:
async function stopOnHoldMusic() {
// Assume we have an audio transceiver and a microphone track named micTrack
await audio.sender.replaceTrack(micTrack);
// Unmute received audio
audio.receiver.track.enabled = true;
// Set the direction to sendrecv (requires negotiation)
audio.direction = 'sendrecv';
}
To respond to being taken off hold by a remote peer:
async function onOffHold() {
try {
// Apply the sendrecv offer first, to ensure receiver is ready for ICE candidates.
await pc.setRemoteDescription(sendrecvOffer);
// Start sending audio
await audio.sender.replaceTrack(micTrack);
// Set the direction sendrecv (just in time for the answer)
audio.direction = 'sendrecv';
// Call createAnswer and send a sendrecv answer
await doAnswer();
} catch (err) {
// handle signaling error
}
}
RTCDtlsTransport
Interface
The
interface allows an application access to
information about the Datagram Transport Layer Security (DTLS)
transport over which RTP and RTCP packets are sent and received by
RTCDtlsTransport
and RTCRtpSender
objects, as well other data
such as SCTP packets sent and received by data channels. In
particular, DTLS adds security to an underlying transport, and the
RTCRtpReceiver
interface allows access to information about the
underlying transport and the security added. RTCDtlsTransport
objects are constructed as a result of calls to
RTCDtlsTransport
setLocalDescription
()
and
setRemoteDescription
()
. Each
object represents the DTLS transport layer for
the RTP or RTCP RTCDtlsTransport
of a specific
component
, or a group of RTCRtpTransceiver
s if such a
group has been negotiated via [RFC8843].
RTCRtpTransceiver
RTCRtpTransceiver
will be
represented by an existing RTCDtlsTransport
object, whose
state
will be updated accordingly, as opposed to
being represented by a new object.
An
has a [[DtlsTransportState]]
internal slot initialized to "RTCDtlsTransport
" and a
[[RemoteCertificates]] slot initialized to an empty list.
new
When the underlying DTLS transport experiences an error, such as a certificate validation failure, or a fatal alert (see [RFC5246] section 7.2), the user agent MUST queue a task that runs the following steps:
Let transport be the
object to
receive the state update and error notification.
RTCDtlsTransport
If the state of transport is already
"
", abort these steps.
failed
Set transport.[[DtlsTransportState]] to
"
".
failed
Fire an event named
using the
error
interface with its errorDetail attribute set to
either "RTCErrorEvent
" or
"dtls-failure
", as appropriate, and
other fields set as described under the fingerprint-failure
enum description, at transport.
RTCErrorDetailType
Fire an event named
at
transport.
statechange
When the underlying DTLS transport needs to update the state of the
corresponding
object for any other reason, the
user agent MUST queue a task that runs the following steps:
RTCDtlsTransport
Let transport be the
object to
receive the state update.
RTCDtlsTransport
Let newState be the new state.
Set transport.[[DtlsTransportState]] to newState.
If newState is
then let newRemoteCertificates be the certificate
chain in use by the remote side, with each certificate encoded in
binary Distinguished Encoding Rules (DER) [X690], and set
transport.[[RemoteCertificates]] to
newRemoteCertificates.
connected
Fire an event named
at
transport.
statechange
WebIDL[Exposed=Window] interfaceRTCDtlsTransport
: EventTarget { [SameObject] readonly attributeRTCIceTransport
iceTransport
; readonly attributeRTCDtlsTransportState
state
; sequence<ArrayBuffer>getRemoteCertificates
(); attribute EventHandleronstatechange
; attribute EventHandleronerror
; };
iceTransport
of type RTCIceTransport
, readonly
The
attribute is the underlying transport
that is used to send and receive packets. The underlying
transport may not be shared between multiple active
iceTransport
objects.
RTCDtlsTransport
state
of type RTCDtlsTransportState
, readonly
The
attribute MUST, on getting, return the value of
the [[DtlsTransportState]] slot.
state
onstatechange
of type EventHandler
statechange
.
onerror
of type EventHandler
error
.
getRemoteCertificates
Returns the value of [[RemoteCertificates]].
RTCDtlsTransportState
Enum
WebIDLenumRTCDtlsTransportState
{ "new
", "connecting
", "connected
", "closed
", "failed
" };
Enumeration description | |
---|---|
new
|
DTLS has not started negotiating yet. |
connecting
|
DTLS is in the process of negotiating a secure connection and verifying the remote fingerprint. |
connected
|
DTLS has completed negotiation of a secure connection and verified the remote fingerprint. |
closed
|
The transport has been closed intentionally as the result of
receipt of a close_notify alert, or calling
() .
|
failed
|
The transport has failed as the result of an error (such as receipt of an error alert or failure to validate the remote fingerprint). |
RTCDtlsFingerprint
Dictionary
The
dictionary includes the hash function
algorithm and certificate fingerprint as described in [RFC4572].
RTCDtlsFingerprint
WebIDLdictionaryRTCDtlsFingerprint
{ DOMStringalgorithm
; DOMStringvalue
; };
algorithm
of type DOMString
One of the the hash function algorithms defined in the 'Hash function Textual Names' registry [IANA-HASH-FUNCTION].
value
of type DOMString
The value of the certificate fingerprint in lowercase hex string as expressed utilizing the syntax of 'fingerprint' in [RFC4572] Section 5.
RTCIceTransport
Interface
The
interface allows an application access to
information about the ICE transport over which packets are sent and
received. In particular, ICE manages peer-to-peer connections which
involve state which the application may want to access.
RTCIceTransport
objects are constructed as a result of calls to
RTCIceTransport
setLocalDescription
()
and
setRemoteDescription
()
. The underlying ICE
state is managed by the ICE agent; as such, the state of an
changes when the ICE Agent provides
indications to the user agent as described below. Each
RTCIceTransport
object represents the ICE transport layer for the
RTP or RTCP RTCIceTransport
of a specific
component
, or a group of RTCRtpTransceiver
s if such a
group has been negotiated via [RFC8843].
RTCRtpTransceiver
RTCRtpTransceiver
will be
represented by an existing RTCIceTransport
object, whose
state
will be updated accordingly, as opposed to
being represented by a new object.
When the ICE Agent indicates that it began gathering a generation of candidates for an
, the user
agent MUST queue a task that runs the following steps:
RTCIceTransport
Let connection be the
object
associated with this ICE Agent.
RTCPeerConnection
If connection.[[IsClosed]] is
true
, abort these steps.
Let transport be the
for which
candidate gathering began.
RTCIceTransport
Set transport.[[IceGathererState]] to
.
gathering
Fire an event named
at
transport.
gatheringstatechange
Update the ICE gathering state of connection.
When the ICE Agent is finished gathering a generation of
candidates for an
, and those candidates have been
surfaced to the application, the user agent MUST queue a task that
runs the following steps:
RTCIceTransport
Let connection be the
object
associated with this ICE Agent.
RTCPeerConnection
If connection.[[IsClosed]] is
true
, abort these steps.
Let transport be the
for which
candidate gathering finished.
RTCIceTransport
Let newCandidate be the result of creating an
RTCIceCandidate with a new dictionary whose
and
sdpMid
are set to the values
associated with this sdpMLineIndex
,
RTCIceTransport
is set to the username
fragment of the generation of candidates for which
gathering finished, and usernameFragment
is set
to an empty string.
candidate
Fire an event named
using the
icecandidate
interface with the candidate
attribute set to newCandidate at
connection.
RTCPeerConnectionIceEvent
If another generation of candidates is still being gathered, abort these steps.
Set transport.[[IceGathererState]] to
.
complete
Fire an event named
at
transport.
gatheringstatechange
Update the ICE gathering state of connection.
When the ICE Agent indicates that a new ICE candidate is
available for an
, either by taking one from the
ICE candidate pool or gathering it
from scratch, the user agent MUST queue a task that runs the
following steps:
RTCIceTransport
Let candidate be the available ICE candidate.
Let connection be the
object
associated with this ICE Agent.
RTCPeerConnection
If connection.[[IsClosed]] is
true
, abort these steps.
If either
connection.[[PendingLocalDescription]] or
connection.[[CurrentLocalDescription]] are not
null
, and represent the ICE generation for
which candidate was gathered, surface the candidate with candidate and connection, and abort
these steps.
Otherwise, append candidate to connection.[[EarlyCandidates]].
When the ICE Agent signals that the ICE role has changed due to an ICE binding request with a role collision per [RFC8445] section 7.3.1.1, the UA will queue a task to set the value of [[IceRole]] to the new value.
To release early candidates of a connection, run the following steps:
For each candidate, candidate, in connection.[[EarlyCandidates]], queue a task to surface the candidate with candidate and connection.
Set connection.[[EarlyCandidates]] to an empty list.
To surface a candidate with candidate and connection, run the following steps:
If connection.[[IsClosed]] is
true
, abort these steps.
Let transport be the
for which
candidate is being made available.
RTCIceTransport
If connection.[[PendingLocalDescription]] is
not null
, and represents the ICE generation
for which candidate was gathered, add
candidate to
connection.[[PendingLocalDescription]].sdp.
If connection.[[CurrentLocalDescription]] is
not null
, and represents the ICE generation
for which candidate was gathered, add
candidate to
connection.[[CurrentLocalDescription]].sdp.
Let newCandidate be the result of creating an
RTCIceCandidate with a new dictionary whose
and
sdpMid
are set to the values
associated with this sdpMLineIndex
,
RTCIceTransport
is set to the username
fragment of the candidate, and usernameFragment
is set to a string encoded using the candidate
candidate-attribute
grammar to represent candidate.
Add newCandidate to transport's set of local candidates.
Fire an event named
using the
icecandidate
interface with the candidate
attribute set to newCandidate at
connection.
RTCPeerConnectionIceEvent
The
of an RTCIceTransportState
may change
because a candidate pair with a usable connection was found and
selected or it may change without the selected candidate pair
changing. The selected pair and RTCIceTransport
are related
and are handled in the same task.
RTCIceTransportState
When the ICE Agent indicates that an
has
changed either the selected candidate pair, the
RTCIceTransport
or both, the user agent MUST queue a task
that runs the following steps:
RTCIceTransportState
Let connection be the
object
associated with this ICE Agent.
RTCPeerConnection
If connection.[[IsClosed]] is
true
, abort these steps.
Let transport be the
whose state
is changing.
RTCIceTransport
Let selectedCandidatePairChanged be
false
.
Let transportIceConnectionStateChanged be
false
.
Let connectionIceConnectionStateChanged be
false
.
Let connectionStateChanged be false
.
If transport's selected candidate pair was changed, run the following steps:
Let newCandidatePair be a newly created
representing the indicated pair if
one is selected, and RTCIceCandidatePair
null
otherwise.
Set transport.[[SelectedCandidatePair]] to newCandidatePair.
Set selectedCandidatePairChanged to
true
.
If transport's
was changed,
run the following steps:
RTCIceTransportState
Set transport.[[IceTransportState]] to the
new indicated
.
RTCIceTransportState
Set transportIceConnectionStateChanged to
true
.
Set connection's ICE connection state to the
value of deriving a new state value as described by the
enum.
RTCIceConnectionState
If the ice connection state changed in the previous
step, set connectionIceConnectionStateChanged to
true
.
Set connection's connection state to the
value of deriving a new state value as described by the
enum.
RTCPeerConnectionState
If the connection state changed in the previous step,
set connectionStateChanged to true
.
If selectedCandidatePairChanged is true
,
fire an event named
at
transport.
selectedcandidatepairchange
If transportIceConnectionStateChanged is
true
, fire an event named
at
transport.
statechange
If connectionIceConnectionStateChanged is
true
, fire an event named
at connection.
iceconnectionstatechange
If connectionStateChanged is true
, fire an event named
at
connection.
connectionstatechange
An
object has the following internal slots:
RTCIceTransport
new
"
new
"
null
unknown
"
WebIDL[Exposed=Window] interfaceRTCIceTransport
: EventTarget { readonly attributeRTCIceRole
role
; readonly attributeRTCIceComponent
component
; readonly attributeRTCIceTransportState
state
; readonly attributeRTCIceGathererState
gatheringState
; sequence<RTCIceCandidate
>getLocalCandidates
(); sequence<RTCIceCandidate
>getRemoteCandidates
();RTCIceCandidatePair
?getSelectedCandidatePair
();RTCIceParameters
?getLocalParameters
();RTCIceParameters
?getRemoteParameters
(); attribute EventHandleronstatechange
; attribute EventHandlerongatheringstatechange
; attribute EventHandleronselectedcandidatepairchange
; };
role
of type RTCIceRole
, readonly
The
attribute MUST, on getting, return the value of
the [[IceRole]] internal slot.
role
component
of type
RTCIceComponent
, readonly
The
attribute MUST return the ICE component of
the transport. When RTCP mux is used, a single
component
transports both RTP and RTCP and
RTCIceTransport
is set to "component
".
rtp
state
of type
RTCIceTransportState
,
readonly
The
attribute MUST, on getting, return the value of
the [[IceTransportState]] slot.
state
gatheringState
of type RTCIceGathererState
, readonly
The
attribute MUST, on getting, return the
value of the [[IceGathererState]] slot.
gatheringState
onstatechange
of type EventHandler
statechange
, MUST be fired any time the RTCIceTransport
state
changes.
ongatheringstatechange
of type
EventHandler
gatheringstatechange
, MUST be fired any time the
RTCIceTransport
ICE gathering state changes.
onselectedcandidatepairchange
of type
EventHandler
selectedcandidatepairchange
, MUST be fired any time the
RTCIceTransport
's selected candidate pair changes.
getLocalCandidates
Returns a sequence describing the local ICE candidates
gathered for this
and sent in
RTCIceTransport
.
onicecandidate
getRemoteCandidates
Returns a sequence describing the remote ICE candidates
received by this
via
RTCIceTransport
addIceCandidate
()
.
getRemoteCandidates
will not expose peer reflexive
candidates since they are not received via
addIceCandidate
()
.
getSelectedCandidatePair
Returns the selected candidate pair on which packets are
sent. This method MUST return the value of the
[[SelectedCandidatePair]] slot. When
.RTCIceTransport
is
"state
" or
"new
"
closed
returns getSelectedCandidatePair
null
.
getLocalParameters
Returns the local ICE parameters received by this
via
RTCIceTransport
, or
setLocalDescription
null
if the parameters have not yet been
received.
getRemoteParameters
Returns the remote ICE parameters received by this
via
RTCIceTransport
or
setRemoteDescription
null
if the parameters have not yet been
received.
RTCIceParameters
Dictionary
WebIDLdictionaryRTCIceParameters
{ DOMStringusernameFragment
; DOMStringpassword
; };
RTCIceParameters
Members
RTCIceCandidatePair
Dictionary
WebIDLdictionaryRTCIceCandidatePair
{RTCIceCandidate
local
;RTCIceCandidate
remote
; };
RTCIceCandidatePair
Members
local
of type RTCIceCandidate
The local ICE candidate.
remote
of type RTCIceCandidate
The remote ICE candidate.
RTCIceGathererState
Enum
WebIDLenumRTCIceGathererState
{ "new
", "gathering
", "complete
" };
Enumeration description
|
|
---|---|
new
|
The was just created, and has not
started gathering candidates yet.
|
gathering
|
The is in the process of gathering
candidates.
|
complete
|
The has completed gathering and the
end-of-candidates indication for this transport has been
sent. It will not gather candidates again until an ICE
restart causes it to restart.
|
RTCIceTransportState
Enum
WebIDLenumRTCIceTransportState
{ "new
", "checking
", "connected
", "completed
", "disconnected
", "failed
", "closed
" };
Enumeration description
|
|
---|---|
new
|
The is gathering candidates and/or
waiting for remote candidates to be supplied, and has not
yet started checking.
|
checking
|
The has received at least one remote
candidate and is checking candidate pairs and has either
not yet found a connection or consent checks [RFC7675]
have failed on all previously successful candidate pairs.
In addition to checking, it may also still be gathering.
|
connected
|
The has found a usable connection, but
is still checking other candidate pairs to see if there is
a better connection. It may also still be gathering and/or
waiting for additional remote candidates. If consent checks
[RFC7675] fail on the connection in use, and there are
no other successful candidate pairs available, then the
state transitions to " "
(if there are candidate pairs remaining to be checked) or
" " (if there are no
candidate pairs to check, but the peer is still gathering
and/or waiting for additional remote candidates).
|
completed
|
The has finished gathering, received an
indication that there are no more remote candidates,
finished checking all candidate pairs and found a
connection. If consent checks [RFC7675] subsequently
fail on all successful candidate pairs, the state
transitions to " ".
|
disconnected
|
The ICE Agent has determined that connectivity is
currently lost for this . This is a
transient state that may trigger intermittently (and
resolve itself without action) on a flaky network. The way
this state is determined is implementation dependent.
Examples include:
has finished
checking all existing candidates pairs and not found a
connection (or consent checks [RFC7675] once successful,
have now failed), but it is still gathering and/or waiting
for additional remote candidates.
|
failed
|
The has finished gathering, received an
indication that there are no more remote candidates,
finished checking all candidate pairs, and all pairs have
either failed connectivity checks or have lost consent.
This is a terminal state until ICE is restarted. Since an
ICE restart may cause connectivity to resume, entering the
" " state does not cause DTLS
transports, SCTP associations or the data channels that run
over them to close, or tracks to mute.
|
closed
|
The has shut down and is no longer
responding to STUN requests.
|
The most common transitions for a successful call will be new ->
checking -> connected -> completed, but under specific
circumstances (only the last checked candidate succeeds, and
gathering and the no-more candidates indication both occur prior to
success), the state can transition directly from
"
" to
"checking
".
completed
An ICE restart causes candidate gathering and connectivity checks to
begin anew, causing a transition to
"
" if begun in the
"connected
" state. If begun in the
transient "completed
" state, it causes
a transition to "disconnected
", effectively
forgetting that connectivity was previously lost.
checking
The "
" and
"failed
" states require an indication
that there are no additional remote candidates. This can be
indicated by calling completed
with a
candidate value whose addIceCandidate
property is set
to an empty string or by
candidate
being set to
canTrickleIceCandidates
false
.
Some example state transitions are:
RTCIceTransport
first created, as a result of
setLocalDescription
or
setRemoteDescription
):
"new
"
new
", remote candidates received):
"checking
"
checking
", found usable connection):
"connected
"
checking
", checks fail but gathering
still in progress): "disconnected
"
checking
", gave up):
"failed
"
disconnected
", new local
candidates): "checking
"
connected
", finished all checks):
"completed
"
completed
", lost connectivity):
"disconnected
"
disconnected
" or
"failed
", ICE restart occurs):
"checking
"
completed
", ICE restart occurs):
"connected
"
RTCPeerConnection
.close
()
:
"closed
"
RTCIceRole
Enum
WebIDLenumRTCIceRole
{ "unknown
", "controlling
", "controlled
" };
Enumeration description
|
|
---|---|
unknown
|
An agent whose role as defined by [RFC5245], Section 3, has not yet been determined. |
controlling
|
A controlling agent as defined by [RFC5245], Section 3. |
controlled
|
A controlled agent as defined by [RFC5245], Section 3. |
RTCIceComponent
Enum
WebIDLenumRTCIceComponent
{ "rtp
", "rtcp
" };
Enumeration description
|
|
---|---|
rtp
|
The ICE Transport is used for RTP (or RTCP multiplexing),
as defined in [RFC5245], Section 4.1.1.1. Protocols
multiplexed with RTP (e.g. data channel) share its
component ID. This represents the component-id value 1 when encoded
in candidate-attribute .
|
rtcp
|
The ICE Transport is used for RTCP as defined by [RFC5245],
Section 4.1.1.1. This represents the component-id value 2 when encoded
in candidate-attribute .
|
RTCTrackEvent
The
event uses the track
interface.
RTCTrackEvent
WebIDL[Exposed=Window] interfaceRTCTrackEvent
: Event {constructor
(DOMString type,RTCTrackEventInit
eventInitDict); readonly attributeRTCRtpReceiver
receiver
; readonly attribute MediaStreamTracktrack
; [SameObject] readonly attribute FrozenArray<MediaStream>streams
; readonly attributeRTCRtpTransceiver
transceiver
; };
RTCTrackEvent.constructor()
receiver
of type
RTCRtpReceiver
, readonly
The
attribute represents the receiver
object associated with the event.
RTCRtpReceiver
track
of type MediaStreamTrack
, readonly
The
attribute represents the track
MediaStreamTrack
object that is associated with the
identified by RTCRtpReceiver
.
receiver
streams
of type FrozenArray<MediaStream
>,
readonly
The
attribute returns an array of streams
MediaStream
objects representing the MediaStream
s that this event's
is a part of.
track
transceiver
of type
RTCRtpTransceiver
,
readonly
The
attribute represents the
transceiver
object associated with the event.
RTCRtpTransceiver
WebIDLdictionaryRTCTrackEventInit
: EventInit { requiredRTCRtpReceiver
receiver
; required MediaStreamTracktrack
; sequence<MediaStream>streams
= []; requiredRTCRtpTransceiver
transceiver
; };
RTCTrackEventInit
Members
receiver
of type RTCRtpReceiver
, required
The
member represents the receiver
object associated with the event.
RTCRtpReceiver
track
of type MediaStreamTrack
, required
The
member represents the track
MediaStreamTrack
object that is associated with the
identified by RTCRtpReceiver
.
receiver
streams
of type sequence<MediaStream
>,
defaulting to []
The
member is an array of streams
MediaStream
objects
representing the MediaStream
s that this event's
is a part of.
track
transceiver
of type RTCRtpTransceiver
, required
The
attribute represents the
transceiver
object associated with the event.
RTCRtpTransceiver
The Peer-to-peer Data API lets a web application send and receive generic application data peer-to-peer. The API for sending and receiving data models the behavior of Web Sockets.
The Peer-to-peer data API extends the
interface
as described below.
RTCPeerConnection
WebIDL partial interfaceRTCPeerConnection
{ readonly attributeRTCSctpTransport
?sctp
;RTCDataChannel
createDataChannel
(USVString label, optionalRTCDataChannelInit
dataChannelDict = {}); attribute EventHandlerondatachannel
; };
sctp
of type RTCSctpTransport
, readonly, nullable
The SCTP transport over which SCTP data is sent and received.
If SCTP has not been negotiated, the value is null. This
attribute MUST return the
object stored
in the [[SctpTransport]] internal slot.
RTCSctpTransport
ondatachannel
of type EventHandler
datachannel
.
createDataChannel
Creates a new
object with the given label.
The RTCDataChannel
dictionary can be used to
configure properties of the underlying channel such as data
reliability.
RTCDataChannelInit
When the
method is invoked, the user
agent MUST run the following steps.
createDataChannel
Let connection be the
object on which the method is invoked.
RTCPeerConnection
If connection.[[IsClosed]] is
true
, throw an
InvalidStateError
.
Create an RTCDataChannel, channel.
Initialize channel.[[DataChannelLabel]] to the value of the first argument.
If the UTF-8 representation of
[[DataChannelLabel]] is longer than 65535 bytes,
throw a TypeError
.
Let options be the second argument.
Initialize
channel.[[MaxPacketLifeTime]] to
option.
,
if present, otherwise maxPacketLifeTime
null
.
Initialize channel.[[MaxRetransmits]]
to
option.
,
if present, otherwise maxRetransmits
null
.
Initialize channel.[[Ordered]] to
option.
.
ordered
Initialize
channel.[[DataChannelProtocol]] to
option.
.
protocol
If the UTF-8 representation of
[[DataChannelProtocol]] is longer than 65535
bytes, throw a TypeError
.
Initialize channel.[[Negotiated]] to
option.
.
negotiated
Initialize channel.[[DataChannelId]]
to the value of
option.
, if it is
present and [[Negotiated]] is true, otherwise
id
null
.
If [[Negotiated]] is true
and
[[DataChannelId]] is null
, throw a TypeError
.
If both [[MaxPacketLifeTime]] and
[[MaxRetransmits]] attributes are set (not null),
throw a TypeError
.
If a setting, either [[MaxPacketLifeTime]] or [[MaxRetransmits]], has been set to indicate unreliable mode, and that value exceeds the maximum value supported by the user agent, the value MUST be set to the user agents maximum value.
If [[DataChannelId]] is equal to 65535, which is
greater than the maximum allowed ID of 65534 but still
qualifies as an unsigned
short, throw a TypeError
.
If the [[DataChannelId]] slot is
null
(due to no ID being passed into
, or [[Negotiated]] being
false), and the DTLS role of the SCTP transport has
already been negotiated, then initialize
[[DataChannelId]] to a value generated by the
user agent, according to [RFC8832], and
skip to the next step. If no available ID could be
generated, or if the value of the
[[DataChannelId]] slot is being used by an
existing createDataChannel
, throw an
RTCDataChannel
OperationError
exception.
null
after this step, it will be populated
during the RTCSctpTransport connected procedure.
Let transport be connection.[[SctpTransport]].
If the [[DataChannelId]] slot is not
null
, transport is in the
"
" state and
[[DataChannelId]] is greater or equal to
transport.[[MaxChannels]], throw an connected
OperationError
.
If channel is the first
created on connection, update the
negotiation-needed flag for connection.
RTCDataChannel
Return channel and continue the following steps in parallel.
Create channel's associated underlying data transport and configure it according to the relevant properties of channel.
RTCSctpTransport
Interface
The
interface allows an application access to
information about the SCTP data channels tied to a particular SCTP
association.
RTCSctpTransport
To create an
with an initial
state, initialState, run the following steps:
RTCSctpTransport
Let transport be a new
object.
RTCSctpTransport
Let transport have a [[SctpTransportState]] internal slot initialized to initialState.
Let transport have a [[MaxMessageSize]] internal slot and run the steps labeled update the data max message size to initialize it.
Let transport have a [[MaxChannels]]
internal slot initialized to null
.
Return transport.
To update the data max message size of an
run the following steps:
RTCSctpTransport
Let transport be the
object
to be updated.
RTCSctpTransport
Let remoteMaxMessageSize be the value of the
max-message-size
SDP attribute read
from the remote description, as described in [RFC8841]
(section 6), or 65536 if the attribute is missing.
Let canSendSize be the number of bytes that this client can send (i.e. the size of the local send buffer) or 0 if the implementation can handle messages of any size.
If both remoteMaxMessageSize and canSendSize are 0, set [[MaxMessageSize]] to the positive Infinity value.
Else, if either remoteMaxMessageSize or canSendSize is 0, set [[MaxMessageSize]] to the larger of the two.
Else, set [[MaxMessageSize]] to the smaller of remoteMaxMessageSize or canSendSize.
Once an SCTP transport
is connected, meaning the SCTP association of an
has been established, run the following steps:
RTCSctpTransport
Let transport be the
object.
RTCSctpTransport
Let connection be the
object
associated with transport.
RTCPeerConnection
Set [[MaxChannels]] to the minimum of the negotiated amount of incoming and outgoing SCTP streams.
For each of connection's
:
RTCDataChannel
Let channel be the
object.
RTCDataChannel
If channel.[[DataChannelId]] is
null
, initialize [[DataChannelId]]
to the value generated by the underlying sctp data
channel, according to [RFC8832].
If channel.[[DataChannelId]] is greater or equal to transport.[[MaxChannels]], or the previous step failed to assign an id, close the channel due to a failure. Otherwise, announce the channel as open.
Fire an event named
at
transport.
statechange
This event is fired before the open
events fired by announcing the channel as open;
the open
events are fired from a
queued task.
WebIDL[Exposed=Window] interfaceRTCSctpTransport
: EventTarget { readonly attributeRTCDtlsTransport
transport
; readonly attributeRTCSctpTransportState
state
; readonly attribute unrestricted doublemaxMessageSize
; readonly attribute unsigned short?maxChannels
; attribute EventHandleronstatechange
; };
transport
of type RTCDtlsTransport
, readonly
The transport over which all SCTP packets for data channels will be sent and received.
state
of type RTCSctpTransportState
, readonly
The current state of the SCTP transport. On getting, this attribute MUST return the value of the [[SctpTransportState]] slot.
maxMessageSize
of type unrestricted double, readonly
The maximum size of data that can be passed to
's RTCDataChannel
send
()
method. The
attribute MUST, on getting, return the value of the
[[MaxMessageSize]] slot.
maxChannels
of type unsigned short , readonly, nullable
The maximum amount of
's that can be used
simultaneously. The attribute MUST, on getting, return the
value of the [[MaxChannels]] slot.
RTCDataChannel
null
until the
SCTP transport goes into the
"connected
" state.
onstatechange
of type EventHandler
The event type of this event handler is
.
statechange
RTCSctpTransportState
Enum
indicates the state of the SCTP
transport.
RTCSctpTransportState
WebIDLenumRTCSctpTransportState
{ "connecting
", "connected
", "closed
" };
Enumeration description | |
---|---|
connecting
|
The |
connected
|
When the negotiation of an association is completed, a
task is queued to update the [[SctpTransportState]] slot
to " |
closed
|
A task is queued to update the [[SctpTransportState]]
slot to "
Note that the last transition is logical due to the fact that an SCTP association requires an established DTLS connection - [RFC8261] section 6.1 specifies that SCTP over DTLS is single-homed - and that no way of of switching to an alternate transport is defined in this API. |
RTCDataChannel
The
interface represents a bi-directional data
channel between two peers. An RTCDataChannel
is created via a
factory method on an RTCDataChannel
object. The messages sent
between the browsers are described in [RFC8831] and
[RFC8832].
RTCPeerConnection
There are two ways to establish a connection with
.
The first way is to simply create an RTCDataChannel
at one of the
peers with the RTCDataChannel
negotiated
dictionary member unset or set to its default
value false. This will announce the new channel in-band and trigger
an RTCDataChannelInit
with the corresponding RTCDataChannelEvent
object at the other peer. The second way is to let the application
negotiate the RTCDataChannel
. To do this, create an
RTCDataChannel
object with the RTCDataChannel
negotiated
dictionary member set to true, and signal
out-of-band (e.g. via a web server) to the other side that it SHOULD
create a corresponding RTCDataChannelInit
with the
RTCDataChannel
negotiated
dictionary
member set to true and the same RTCDataChannelInit
. This will
connect the two separately created id
objects. The
second way makes it possible to create channels with asymmetric
properties and to create channels in a declarative way by specifying
matching RTCDataChannel
s.
id
Each
has an associated underlying data transport that is used to
transport actual data to the other peer. In the case of SCTP data
channels utilizing an RTCDataChannel
(which represents the
state of the SCTP association), the underlying data transport is the
SCTP stream pair. The transport properties of the underlying data
transport, such as in order delivery settings and reliability
mode, are configured by the peer as the channel is created. The
properties of a channel cannot change after the channel has been
created. The actual wire protocol between the peers is specified by
the WebRTC DataChannel Protocol specification [RFC8831].
RTCSctpTransport
An
can be configured to operate in different
reliability modes. A reliable channel ensures that the data is
delivered at the other peer through retransmissions. An unreliable
channel is configured to either limit the number of retransmissions (
RTCDataChannel
) or set a time during which
transmissions (including retransmissions) are allowed (
maxRetransmits
). These properties can not
be used simultaneously and an attempt to do so will result in an
error. Not setting any of these properties results in a reliable
channel.
maxPacketLifeTime
An
, created with
RTCDataChannel
or dispatched via an
createDataChannel
, MUST initially be in the
"RTCDataChannelEvent
" state. When the
connecting
object's underlying data transport is ready,
the user agent MUST announce the RTCDataChannel as open.
RTCDataChannel
To create an
, run the following
steps:
RTCDataChannel
Let channel be a newly created
object.
RTCDataChannel
Let channel have a [[ReadyState]]
internal slot initialized to
"
".
connecting
Let channel have a [[BufferedAmount]]
internal slot initialized to 0
.
Let channel have internal slots named [[DataChannelLabel]], [[Ordered]], [[MaxPacketLifeTime]], [[MaxRetransmits]], [[DataChannelProtocol]], [[Negotiated]], and [[DataChannelId]].
Return channel.
When the user agent is to announce an
as
open, the user agent MUST queue a task to run the following
steps:
RTCDataChannel
If the associated
object's
[[IsClosed]] slot is RTCPeerConnection
true
, abort these
steps.
Let channel be the
object to be
announced.
RTCDataChannel
If channel.[[ReadyState]] is
"
" or
"closing
", abort these steps.
closed
Set channel.[[ReadyState]] to
"
".
open
Fire an event named
at channel.
open
When an underlying data transport is to be announced (the
other peer created a channel with
unset or set to false), the user agent of the peer that did not
initiate the creation process MUST queue a task to run the
following steps:
negotiated
Let connection be the
object
associated with the underlying data transport.
RTCPeerConnection
If connection.[[IsClosed]] is
true
, abort these steps.
Create an RTCDataChannel, channel.
Let configuration be an information bundle received from the other peer as a part of the process to establish the underlying data transport described by the WebRTC DataChannel Protocol specification [RFC8832].
Initialize channel.[[DataChannelLabel]], [[Ordered]], [[MaxPacketLifeTime]], [[MaxRetransmits]], [[DataChannelProtocol]], and [[DataChannelId]] internal slots to the corresponding values in configuration.
Initialize channel.[[Negotiated]] to
false
.
Set channel.[[ReadyState]] to
"
" (but do not fire the open
event, yet).
open
datachannel
event handler prior to the open
event being
fired.
Fire an event named datachannel
using the
interface with the
RTCDataChannelEvent
attribute set to
channel at connection.
channel
An
object's underlying data transport may
be torn down in a non-abrupt manner by running the closing procedure. When
that happens the user agent MUST queue a task to run the following
steps:
RTCDataChannel
Let channel be the
object whose
underlying data transport was closed.
RTCDataChannel
Unless the procedure was initiated by
channel.
, set
channel.[[ReadyState]] to
"close
" and fire an event named
closing
at channel.
closing
Run the following steps in parallel:
Finish sending all currently pending messages of the channel.
Follow the closing procedure defined for the channel's underlying data transport :
Render the channel's data transport
closed
by following the associated procedure.
When an
object's underlying data transport
has been closed, the user agent MUST queue a task to run the
following steps:
RTCDataChannel
Let channel be the
object whose
underlying data transport was closed.
RTCDataChannel
closed
", abort these steps.
Set channel.[[ReadyState]] to
"
".
closed
If the transport was closed
with an error, fire an event named
using the error
interface with its RTCErrorEvent
attribute set to
"errorDetail
" at channel.
sctp-failure
Fire an event named
at channel.
close
In some cases, the user agent may be unable to create an
's underlying data transport. For
example, the data channel's RTCDataChannel
may be outside
the range negotiated by the [RFC8831] implementations in the
SCTP handshake. When the user agent determines that an
id
's underlying data transport cannot be
created, the user agent MUST queue a task to run the following
steps:
RTCDataChannel
Let channel be the
object for
which the user agent could not create an underlying data
transport.
RTCDataChannel
Set channel.[[ReadyState]] to
"
".
closed
Fire an event named
using the
error
interface with the RTCErrorEvent
attribute set to "errorDetail
"
at channel.
data-channel-failure
Fire an event named
at channel.
close
When an
message has
been received via the underlying data transport with
type type and data rawData, the user agent
MUST queue a task to run the following steps:
RTCDataChannel
Let channel be the
object for
which the user agent has received a message.
RTCDataChannel
Let connection be the
object
associated with channel.
RTCPeerConnection
If channel.[[ReadyState]] is not
"
", abort these steps and discard
rawData.
open
Execute the sub step by switching on type and
channel.
:
binaryType
If type indicates that rawData is a
string
:
Let data be a DOMString that represents the result of decoding rawData as UTF-8.
If type indicates that rawData is
binary and
is binaryType
"blob"
:
Let data be a new Blob
object containing
rawData as its raw data source.
If type indicates that rawData is
binary and
is binaryType
"arraybuffer"
:
Let data be a new ArrayBuffer
object
containing rawData as its raw data source.
Fire an event named
using the
message
MessageEvent
interface with its origin
attribute initialized to the
serialization of an origin of
connection.[[DocumentOrigin]], and the
data
attribute initialized to
data at channel.
WebIDL[Exposed=Window] interfaceRTCDataChannel
: EventTarget { readonly attribute USVStringlabel
; readonly attribute booleanordered
; readonly attribute unsigned short?maxPacketLifeTime
; readonly attribute unsigned short?maxRetransmits
; readonly attribute USVStringprotocol
; readonly attribute booleannegotiated
; readonly attribute unsigned short?id
; readonly attributeRTCDataChannelState
readyState
; readonly attribute unsigned longbufferedAmount
; [EnforceRange] attribute unsigned longbufferedAmountLowThreshold
; attribute EventHandleronopen
; attribute EventHandleronbufferedamountlow
; attribute EventHandleronerror
; attribute EventHandleronclosing
; attribute EventHandleronclose
; undefinedclose
(); attribute EventHandleronmessage
; attribute BinaryTypebinaryType
; undefinedsend
(USVString data); undefinedsend
(Blob data); undefinedsend
(ArrayBuffer data); undefinedsend
(ArrayBufferView data); };
label
of type
USVString, readonly
The
attribute represents a label that can be used
to distinguish this label
object from other
RTCDataChannel
objects. Scripts are allowed to create
multiple RTCDataChannel
objects with the same label. On
getting, the attribute MUST return the value of the
[[DataChannelLabel]] slot.
RTCDataChannel
ordered
of type
boolean, readonly
The
attribute returns true if the
ordered
is ordered, and false if out of order
delivery is allowed. On getting, the attribute MUST return
the value of the [[Ordered]] slot.
RTCDataChannel
maxPacketLifeTime
of
type unsigned short, readonly,
nullable
The
attribute returns the length of the
time window (in milliseconds) during which transmissions and
retransmissions may occur in unreliable mode. On getting, the
attribute MUST return the value of the
[[MaxPacketLifeTime]] slot.
maxPacketLifeTime
maxRetransmits
of type unsigned short,
readonly, nullable
The
attribute returns the maximum number
of retransmissions that are attempted in unreliable mode. On
getting, the attribute MUST return the value of the
[[MaxRetransmits]] slot.
maxRetransmits
protocol
of type
USVString, readonly
The
attribute returns the name of the
sub-protocol used with this protocol
. On getting,
the attribute MUST return the value of the
[[DataChannelProtocol]] slot.
RTCDataChannel
negotiated
of type
boolean, readonly
The
attribute returns true if this
negotiated
was negotiated by the application, or
false otherwise. On getting, the attribute MUST return the
value of the [[Negotiated]] slot.
RTCDataChannel
id
of type unsigned short, readonly, nullable
The
attribute returns the ID for this
id
. The value is initially null, which is
what will be returned if the ID was not provided at channel
creation time, and the DTLS role of the SCTP transport has
not yet been negotiated. Otherwise, it will return the ID
that was either selected by the script or generated by the
user agent according to [RFC8832]. After the
ID is set to a non-null value, it will not change. On
getting, the attribute MUST return the value of the
[[DataChannelId]] slot.
RTCDataChannel
readyState
of type
RTCDataChannelState
,
readonly
The
attribute represents the state of the
readyState
object. On getting, the attribute MUST
return the value of the [[ReadyState]] slot.
RTCDataChannel
bufferedAmount
of type unsigned long,
readonly
The
attribute MUST, on getting, return the
value of the [[BufferedAmount]] slot. The attribute
exposes the number of bytes of application data (UTF-8 text
and binary data) that have been queued using
bufferedAmount
send
()
. Even though the data transmission
can occur in parallel, the returned value MUST NOT be
decreased before the current task yielded back to the event
loop to prevent race conditions. The value does not include
framing overhead incurred by the protocol, or buffering done
by the operating system or network hardware. The value of the
[[BufferedAmount]] slot will only increase with each
call to the send
()
method as long as the
[[ReadyState]] slot is
"
"; however, the slot does not
reset to zero once the channel closes. When the underlying
data transport sends data from its queue, the user agent
MUST queue a task that reduces [[BufferedAmount]]
with the number of bytes that was sent.
open
bufferedAmountLowThreshold
of type
unsigned long
The
attribute sets the
threshold at which the bufferedAmountLowThreshold
is
considered to be low. When the
bufferedAmount
decreases from above this
threshold to equal or below it, the bufferedAmount
event fires. The
bufferedamountlow
is initially
zero on each new bufferedAmountLowThreshold
, but the application may
change its value at any time.
RTCDataChannel
onopen
of type EventHandler
open
.
onbufferedamountlow
of type EventHandler
bufferedamountlow
.
onerror
of type EventHandler
The event type of this event handler is
.
RTCErrorEvent
contains "sctp-failure",
errorDetail
contains the SCTP Cause Code
value, and sctpCauseCode
contains the SCTP
Cause-Specific-Information, possibly with additional text.
message
onclosing
of type EventHandler
The event type of this event handler is
.
closing
onclose
of type EventHandler
The event type of this event handler is
.
close
onmessage
of type EventHandler
The event type of this event handler is
.
message
binaryType
of type
BinaryType
The
attribute MUST, on getting, return the
value to which it was last set. On setting, if the new value
is either the string binaryType
"blob"
or the
string "arraybuffer"
, then set the
IDL attribute to this new value. Otherwise, throw a SyntaxError
. When an
object is created, the
RTCDataChannel
attribute MUST be initialized
to the string binaryType
"blob"
.
This attribute controls how binary data is exposed to
scripts. See Web Socket's binaryType
.
close
Closes the
. It may be called regardless of
whether the RTCDataChannel
object was created by this
peer or the remote peer.
RTCDataChannel
When the
method is called, the user agent MUST run
the following steps:
close
Let channel be the
object
which is about to be closed.
RTCDataChannel
If channel.[[ReadyState]] is
"
" or
"closing
", then abort these steps.
closed
Set channel.[[ReadyState]] to
"
".
closing
If the closing procedure has not started yet, start it.
send
Run the steps described by the send() algorithm with
argument type string
object.
send
Run the steps described by the send() algorithm with
argument type Blob
object.
send
Run the steps described by the send() algorithm with
argument type ArrayBuffer
object.
send
Run the steps described by the send() algorithm with
argument type ArrayBufferView
object.
The send()
method is overloaded to
handle different data argument types. When any version of the
method is called, the user agent MUST run the following
steps:
Let channel be the
object on
which data is to be sent.
RTCDataChannel
If channel.[[ReadyState]] is not
"
", throw an
open
InvalidStateError
.
Execute the sub step that corresponds to the type of the methods argument:
string
object:
Let data be a byte buffer that represents the result of encoding the method's argument as UTF-8.
Blob
object:
Let data be the raw data represented by the
Blob
object.
Blob
object can happen asynchronously, the user agent
will make sure to queue the data on the
channel's underlying data transport in
the same order as the send method is called. The byte
size of data needs to be known synchronously.
ArrayBuffer
object:
Let data be the data stored in the buffer
described by the ArrayBuffer
object.
ArrayBufferView
object:
Let data be the data stored in the section of
the buffer described by the ArrayBuffer
object that
the ArrayBufferView
object references.
TypeError
. This includes
null
and undefined
.
If the byte size of data exceeds the value of
on channel's
associated maxMessageSize
, throw a
RTCSctpTransport
TypeError
.
Queue data for transmission on
channel's underlying data transport. If
queuing data is not possible because not enough
buffer space is available, throw an
OperationError
.
onerror
.
Increase the value of the [[BufferedAmount]] slot by the byte size of data.
WebIDLdictionaryRTCDataChannelInit
{ booleanordered
= true; [EnforceRange] unsigned shortmaxPacketLifeTime
; [EnforceRange] unsigned shortmaxRetransmits
; USVStringprotocol
= ""; booleannegotiated
= false; [EnforceRange] unsigned shortid
; };
RTCDataChannelInit
Members
ordered
of type boolean, defaulting to true
If set to false, data is allowed to be delivered out of order. The default value of true, guarantees that data will be delivered in order.
maxPacketLifeTime
of type unsigned short
Limits the time (in milliseconds) during which the channel will transmit or retransmit data if not acknowledged. This value may be clamped if it exceeds the maximum value supported by the user agent.
maxRetransmits
of type unsigned short
Limits the number of times a channel will retransmit data if not successfully delivered. This value may be clamped if it exceeds the maximum value supported by the user agent.
protocol
of type USVString, defaulting to ""
Subprotocol name used for this channel.
negotiated
of type boolean, defaulting to
false
The default value of false tells the user agent to announce
the channel in-band and instruct the other peer to dispatch a
corresponding
object. If set to true, it
is up to the application to negotiate the channel and create
an RTCDataChannel
object with the same
RTCDataChannel
at the other peer.
id
id
of type unsigned short
Sets the channel ID when
is
true. Ignored when negotiated
is
false.
negotiated
WebIDLenumRTCDataChannelState
{ "connecting
", "open
", "closing
", "closed
" };
RTCDataChannelState Enumeration description
|
|
---|---|
connecting
|
The user agent is attempting to establish the underlying
data transport. This is the initial state of an
|
open
|
The underlying data transport is established and communication is possible. |
closing
|
The procedure to close down the underlying data transport has started. |
closed
|
The underlying data transport has been |
RTCDataChannelEvent
The datachannel
event uses the
interface.
RTCDataChannelEvent
WebIDL[Exposed=Window] interfaceRTCDataChannelEvent
: Event {constructor
(DOMString type,RTCDataChannelEventInit
eventInitDict); readonly attributeRTCDataChannel
channel
; };
RTCDataChannelEvent.constructor()
channel
of type
RTCDataChannel
, readonly
The
attribute represents the channel
object associated with the event.
RTCDataChannel
WebIDLdictionaryRTCDataChannelEventInit
: EventInit { requiredRTCDataChannel
channel
; };
RTCDataChannelEventInit
Members
channel
of type RTCDataChannel
, required
The
object to be announced by the event.
RTCDataChannel
An
object MUST not be garbage collected if its
RTCDataChannel
[[ReadyState]] slot is
"
" and at least one event
listener is registered for connecting
open
events, message
events, error
events, closing
events, or close
events.
[[ReadyState]] slot is "
" and
at least one event listener is registered for open
message
events, error
events, closing
events, or
close
events.
[[ReadyState]] slot is "
"
and at least one event listener is registered for closing
error
events, or close
events.
underlying data transport is established and data is queued to be transmitted.
This section describes an interface on
to send DTMF
(phone keypad) values across an RTCRtpSender
. Details of how
DTMF is sent to the other peer are described in [RFC7874].
RTCPeerConnection
The Peer-to-peer DTMF API extends the
interface as
described below.
RTCRtpSender
WebIDL partial interfaceRTCRtpSender
{ readonly attributeRTCDTMFSender
?dtmf
; };
dtmf
of type RTCDTMFSender
, readonly, nullable
On getting, the
attribute returns the value of the
[[Dtmf]] internal slot, which represents a
dtmf
which can be used to send DTMF, or
RTCDTMFSender
null
if unset. The [[Dtmf]] internal
slot is set when the kind of an
's
[[SenderTrack]] is RTCRtpSender
"audio"
.
RTCDTMFSender
To create an RTCDTMFSender, the user agent MUST run the following steps:
Let dtmf be a newly created
object.
RTCDTMFSender
Let dtmf have a [[Duration]] internal slot.
Let dtmf have a [[InterToneGap]] internal slot.
Let dtmf have a [[ToneBuffer]] internal slot.
WebIDL[Exposed=Window] interfaceRTCDTMFSender
: EventTarget { undefinedinsertDTMF
(DOMString tones, optional unsigned long duration = 100, optional unsigned long interToneGap = 70); attribute EventHandlerontonechange
; readonly attribute booleancanInsertDTMF
; readonly attribute DOMStringtoneBuffer
; };
ontonechange
of type EventHandler
The event type of this event handler is
.
tonechange
canInsertDTMF
of type boolean, readonly
Whether the
dtmfSender is
capable of sending DTMF. On getting, the user agent MUST
return the result of running determine if DTMF can be sent for dtmfSender.
RTCDTMFSender
toneBuffer
of type
DOMString, readonly
The
attribute MUST return a list of the tones
remaining to be played out. For the syntax, content, and
interpretation of this list, see toneBuffer
.
insertDTMF
insertDTMF
An
object's RTCDTMFSender
method is used
to send DTMF tones.
insertDTMF
The tones parameter is treated as a series of characters. The characters 0 through 9, A through D, #, and * generate the associated DTMF tones. The characters a to d MUST be normalized to uppercase on entry and are equivalent to A to D. As noted in [RTCWEB-AUDIO] Section 3, support for the characters 0 through 9, A through D, #, and * are required. The character ',' MUST be supported, and indicates a delay of 2 seconds before processing the next character in the tones parameter. All other characters (and only those other characters) MUST be considered unrecognized.
The duration parameter indicates the duration in ms to use for each character passed in the tones parameters. The duration cannot be more than 6000 ms or less than 40 ms. The default duration is 100 ms for each tone.
The interToneGap parameter indicates the gap between tones in ms. The user agent clamps it to at least 30 ms and at most 6000 ms. The default value is 70 ms.
The browser MAY increase the duration and interToneGap times to cause the times that DTMF start and stop to align with the boundaries of RTP packets but it MUST not increase either of them by more than the duration of a single RTP audio packet.
When the insertDTMF
()
method is invoked, the user agent
MUST run the following steps:
RTCRtpSender
used to send DTMF.
Let transceiver be the
object associated with sender.
RTCRtpTransceiver
RTCDTMFSender
associated with sender.
false
, throw an InvalidStateError
.
unrecognized
characters, throw an InvalidCharacterError
.
sendrecv
" nor
"sendonly
", abort these
steps.
tonechange
using
the RTCDTMFToneChangeEvent
interface with the
tone
attribute set to an empty
string at the RTCDTMFSender
object and abort these
steps.
","
delay sending
tones for 2000
ms on the associated RTP
media stream, and queue a task to be executed in
2000
ms from now that runs the steps
labelled Playout task.
","
start
playout of tone for [[Duration]] ms on
the associated RTP media stream, using the appropriate
codec, then queue a task to be executed in
[[Duration]] + [[InterToneGap]] ms from
now that runs the steps labelled Playout task.
tonechange
using the
RTCDTMFToneChangeEvent
interface with the
tone
attribute set to
tone at the RTCDTMFSender
object.
Since
replaces the tone buffer, in order to
add to the DTMF tones being played, it is necessary to call
insertDTMF
with a string containing both the remaining
tones (stored in the [[ToneBuffer]] slot) and the new
tones appended together. Calling insertDTMF
with an empty
tones parameter can be used to cancel all tones queued to
play after the currently playing tone.
insertDTMF
To determine if DTMF can be sent for an
instance dtmfSender, the user agent MUST queue a task that
runs the following steps:
RTCDTMFSender
RTCRtpSender
associated with dtmfSender.
RTCRtpTransceiver
associated with sender.
RTCPeerConnection
associated with transceiver.
RTCPeerConnectionState
is not
"connected
" return false
.
null
return false
.
sendrecv
" nor
"sendonly
" return false
.
[0]
.active
is false
return false
.
"audio/telephone-event"
has been negotiated for sending
with this sender, return false
.
true
.
RTCDTMFToneChangeEvent
The
event uses the tonechange
interface.
RTCDTMFToneChangeEvent
WebIDL[Exposed=Window] interfaceRTCDTMFToneChangeEvent
: Event {constructor
(DOMString type, optionalRTCDTMFToneChangeEventInit
eventInitDict = {}); readonly attribute DOMStringtone
; };
RTCDTMFToneChangeEvent.constructor()
tone
of type DOMString, readonly
The
attribute contains the character for the tone
(including tone
","
) that has just begun playout (see
). If the value is the empty
string, it indicates that the [[ToneBuffer]] slot is
an empty string and that the previous tones have completed
playback.
insertDTMF
WebIDL dictionaryRTCDTMFToneChangeEventInit
: EventInit { DOMStringtone
= ""; };
RTCDTMFToneChangeEventInit
Members
tone
of type DOMString, defaulting to ""
The
attribute contains the character for the tone
(including tone
","
) that has just begun playout (see
). If the value is the empty
string, it indicates that the [[ToneBuffer]] slot is
an empty string and that the previous tones have completed
playback.
insertDTMF
The basic statistics model is that the browser maintains a set of statistics for monitored objects, in the form of stats objects.
A group of related objects may be referenced by a selector. The selector may, for example, be a
MediaStreamTrack
. For a track to be a valid selector, it MUST be
a MediaStreamTrack
that is sent or received by the
object on which the stats request was issued.
The calling Web application provides the selector to the
RTCPeerConnection
getStats
()
method and the browser emits (in the
JavaScript) a set of statistics that are relevant to the selector,
according to the stats selection algorithm. Note that that
algorithm takes the sender or receiver of a selector.
The statistics returned in stats objects are designed in such a
way that repeated queries can be linked by the RTCStats
dictionary member. Thus, a Web application can make
measurements over a given time period by requesting measurements at
the beginning and end of that period.
id
With a few exceptions, monitored objects, once created, exist
for the duration of their associated
. This
ensures statistics from them are available in the result from
RTCPeerConnection
getStats
()
even past the associated peer
connection being
d.
close
Only a few monitored objects have shorter lifetimes. Statistics from these objects are no longer available in subsequent getStats() results. The object descriptions in [WEBRTC-STATS] describe when these monitored objects are deleted.
The Statistics API extends the
interface as
described below.
RTCPeerConnection
WebIDL partial interfaceRTCPeerConnection
{ Promise<RTCStatsReport
>getStats
(optional MediaStreamTrack? selector = null); };
getStats
Gathers stats for the given selector and reports the result asynchronously.
When the getStats
()
method is invoked, the user agent
MUST run the following steps:
Let selectorArg be the method's first argument.
Let connection be the
object on which the method was invoked.
RTCPeerConnection
If selectorArg is null
, let
selector be null
.
If selectorArg is a MediaStreamTrack
let
selector be an
or
RTCRtpSender
on connection which
RTCRtpReceiver
attribute matches
selectorArg. If no such sender or receiver
exists, or if more than one sender or receiver fit this
criteria, return a promise rejected with a newly created track
InvalidAccessError
.
Let p be a new promise.
Run the following steps in parallel:
Gather the stats indicated by selector according to the stats selection algorithm.
Resolve p with the resulting
object, containing the gathered
stats.
RTCStatsReport
Return p.
RTCStatsReport
Object
The getStats
()
method delivers a successful
result in the form of an
object. An
RTCStatsReport
object is a map between strings that identify the
inspected objects (RTCStatsReport
attribute in id
instances), and their corresponding RTCStats
-derived
dictionaries.
RTCStats
An
may be composed of several RTCStatsReport
-derived
dictionaries, each reporting stats for one underlying object that the
implementation thinks is relevant for the selector. One
achieves the total for the selector by summing over all the
stats of a certain type; for instance, if an RTCStats
uses
multiple SSRCs to carry its track over the network, the
RTCRtpSender
may contain one RTCStatsReport
-derived dictionary
per SSRC (which can be distinguished by the value of the
RTCStats
ssrc
stats attribute).
WebIDL[Exposed=Window]
interface RTCStatsReport
{
readonly maplike<DOMString, object>;
};
Use these to retrieve the various dictionaries descended from
that this stats report is composed of. The set of
supported property names [WEBIDL] is defined as the ids of all
the RTCStats
-derived dictionaries that have been generated for
this stats report.
RTCStats
RTCStats
Dictionary
An
dictionary represents the stats object
constructed by inspecting a specific monitored object. The
RTCStats
dictionary is a base type that specifies as set of
default attributes, such as RTCStats
and
timestamp
. Specific stats are added by extending the
type
dictionary.
RTCStats
Note that while stats names are standardized, any given implementation may be using experimental values or values not yet known to the Web application. Thus, applications MUST be prepared to deal with unknown stats.
Statistics need to be synchronized with each other in order to yield
reasonable values in computation; for instance, if
bytesSent
and
packetsSent
are both reported, they both
need to be reported over the same interval, so that "average packet
size" can be computed as "bytes / packets" - if the intervals are
different, this will yield errors. Thus implementations MUST return
synchronized values for all stats in an
-derived
dictionary.
RTCStats
WebIDLdictionaryRTCStats
{ required DOMHighResTimeStamptimestamp
; required RTCStatsTypetype
; required DOMStringid
; };
RTCStats
Members
timestamp
of type DOMHighResTimeStamp
The
, of type timestamp
DOMHighResTimeStamp
,
associated with this object. The time is relative to the UNIX
epoch (Jan 1, 1970, UTC). For statistics that came from a
remote source (e.g., from received RTCP packets),
represents the time at which the information
arrived at the local endpoint. The remote timestamp can be
found in an additional field in an timestamp
-derived
dictionary, if applicable.
RTCStats
type
of type RTCStatsType
The type of this object.
The
attribute MUST be initialized to the name of the
most specific type this type
dictionary represents.
RTCStats
id
of type DOMString
A unique
that is associated with the object that was
inspected to produce this id
object. Two
RTCStats
objects, extracted from two different
RTCStats
objects, MUST have the same id if they
were produced by inspecting the same underlying object.
RTCStatsReport
Stats ids MUST NOT be predictable by an application. This prevents applications from depending on a particular user agent's way of generating ids, since this prevents an application from getting stats objects by their id unless they have already read the id of that specific stats object.
User agents are free to pick any format for the id as long as it meets the requirements above.
A user agent can turn a predictably generated string into an unpredictable string using a hash function, as long as it uses a salt that is unique to the peer connection. This allows an implementation to have predictable ids internally, which may make it easier to guarantee that stats objects have stable ids across getStats() calls.
The set of valid values for RTCStatsType
, and the dictionaries
derived from RTCStats that they indicate, are documented in
[WEBRTC-STATS].
The stats selection algorithm is as follows:
RTCStatsReport
.
null
,
gather stats for the whole connection, add them to
result, return result, and abort these steps.
RTCRtpSender
, gather stats for and
add the following objects to result:
RTCOutboundRtpStreamStats
objects representing RTP
streams being sent by selector.
RTCOutboundRtpStreamStats
objects added.
RTCRtpReceiver
, gather stats for and
add the following objects to result:
RTCInboundRtpStreamStats
objects representing RTP
streams being received by selector.
RTCInboundRtpStreamStats
added.
The stats listed in [WEBRTC-STATS] are intended to cover a wide range of use cases. Not all of them have to be implemented by every WebRTC implementation.
An implementation MUST support generating statistics of the following
s when the corresponding objects exist on a
type
, with the fields that are listed when they are
valid for that object in addition to the generic fields defined in
the RTCPeerConnection
dictionary:
RTCStats
An implementation MAY support generating any other statistic defined in [WEBRTC-STATS], and MAY generate statistics that are not documented.
Consider the case where the user is experiencing bad sound and the application wants to determine if the cause of it is packet loss. The following example code might be used:
async function gatherStats(pc) {
try {
const [sender] = pc.getSenders();
const baselineReport = await sender.getStats();
await new Promise(resolve => setTimeout(resolve, aBit)); // wait a bit
const currentReport = await sender.getStats();
// compare the elements from the current report with the baseline
for (const now of currentReport.values()) {
if (now.type != 'outbound-rtp') continue;
// get the corresponding stats from the baseline report
const base = baselineReport.get(now.id);
if (!base) continue;
const remoteNow = currentReport.get(now.remoteId);
const remoteBase = baselineReport.get(base.remoteId);
const packetsSent = now.packetsSent - base.packetsSent;
const packetsReceived = remoteNow.packetsReceived -
remoteBase.packetsReceived;
const fractionLost = (packetsSent - packetsReceived) / packetsSent;
if (fractionLost > 0.3) {
// if fractionLost is > 0.3, we have probably found the culprit
}
}
} catch (err) {
console.error(err);
}
}
The MediaStreamTrack
interface, as defined in the
[GETUSERMEDIA] specification, typically represents a stream of
data of audio or video. One or more MediaStreamTrack
s can be
collected in a MediaStream
(strictly speaking, a MediaStream
as defined in [GETUSERMEDIA] may contain zero or more
MediaStreamTrack
objects).
A MediaStreamTrack
may be extended to represent a media flow that
either comes from or is sent to a remote peer (and not just the local
camera, for instance). The extensions required to enable this
capability on the MediaStreamTrack
object will be described in
this section. How the media is transmitted to the peer is described
in [RFC8834], [RFC7874], and [RFC8835].
A MediaStreamTrack
sent to another peer will appear as one and
only one MediaStreamTrack
to the recipient. A peer is defined as
a user agent that supports this specification. In addition, the
sending side application can indicate what MediaStream
object(s)
the MediaStreamTrack
is a member of. The corresponding
MediaStream
object(s) on the receiver side will be created (if
not already present) and populated accordingly.
As also described earlier in this document, the objects
and RTCRtpSender
can be used by the
application to get more fine grained control over the transmission
and reception of RTCRtpReceiver
MediaStreamTrack
s.
Channels are the smallest unit considered in the Media Capture and
Streams specification. Channels are intended to be encoded together
for transmission as, for instance, an RTP payload type. All of the
channels that a codec needs to encode jointly MUST be in the same
MediaStreamTrack
and the codecs SHOULD be able to encode, or
discard, all the channels in the track.
The concepts of an input and output to a given MediaStreamTrack
apply in the case of MediaStreamTrack
objects transmitted over
the network as well. A MediaStreamTrack
created by an
object (as described previously in this
document) will take as input the data received from a remote peer.
Similarly, a RTCPeerConnection
MediaStreamTrack
from a local source, for instance a
camera via [GETUSERMEDIA], will have an output that represents
what is transmitted to a remote peer if the object is used with an
object.
RTCPeerConnection
The concept of duplicating MediaStream
and MediaStreamTrack
objects as described in [GETUSERMEDIA] is also applicable here.
This feature can be used, for instance, in a video-conferencing
scenario to display the local video from the user's camera and
microphone in a local monitor, while only transmitting the audio to
the remote peer (e.g. in response to the user using a "video mute"
feature). Combining different MediaStreamTrack
objects into new
MediaStream
objects is useful in certain situations.
In this document, we only specify aspects of the following objects
that are relevant when used along with an
.
Please refer to the original definitions of the objects in the
[GETUSERMEDIA] document for general information on using
RTCPeerConnection
MediaStream
and MediaStreamTrack
.
The id
attribute specified in MediaStream
returns an id that is unique to this stream, so that streams can be
recognized at the remote end of the
API.
RTCPeerConnection
When a MediaStream
is created to represent a stream obtained
from a remote peer, the id
attribute is initialized
from information provided by the remote source.
The id
of a MediaStream
object is unique to the
source of the stream, but that does not mean it is not possible to
end up with duplicates. For example, the tracks of a locally
generated stream could be sent from one user agent to a remote peer
using
and then sent back to the original user
agent in the same manner, in which case the original user agent
will have multiple streams with the same id (the locally-generated
one and the one received from the remote peer).
RTCPeerConnection
A MediaStreamTrack
object's reference to its MediaStream
in
the non-local media source case (an RTP source, as is the case for
each MediaStreamTrack
associated with an
) is
always strong.
RTCRtpReceiver
Whenever an
receives data on an RTP source whose
corresponding RTCRtpReceiver
MediaStreamTrack
is muted, but not ended, and the
[[Receptive]] slot of the
object the
RTCRtpTransceiver
is a member of is RTCRtpReceiver
true
, it MUST queue
a task to set the muted state of the corresponding
MediaStreamTrack
to false
.
When one of the SSRCs for RTP source media streams received by an
is removed either due to reception of a BYE or via
timeout, it MUST queue a task to set the muted state of the
corresponding RTCRtpReceiver
MediaStreamTrack
to true
. Note that
can also lead to the setting of the muted state of the
setRemoteDescription
to the value track
true
.
The procedures add a track, remove a track and set a track's muted state are specified in [GETUSERMEDIA].
When a MediaStreamTrack
track produced by an
receiver has RTCRtpReceiver
ended
[GETUSERMEDIA] (such as via a call to
receiver.
.track
stop
), the user agent MAY choose to free resources
allocated for the incoming stream, by for instance turning off the
decoder of receiver.
The concept of constraints and constrainable properties, including
MediaTrackConstraints
(MediaStreamTrack
.getConstraints()
, MediaStreamTrack
.applyConstraints()
), and MediaTrackSettings
(MediaStreamTrack
.getSettings()
) are
outlined in [GETUSERMEDIA]. However, the constrainable
properties of tracks sourced from a peer connection are different
than those sourced by getUserMedia()
; the
constraints and settings applicable to MediaStreamTrack
s
sourced from a remote source are defined here. The settings
of a remote track represent the latest frame received.
MediaStreamTrack
.getCapabilities()
MUST always return the empty set and
MediaStreamTrack
.applyConstraints()
MUST always reject with OverconstrainedError
on remote tracks for constraints
defined here.
The following constrainable properties are defined to apply to
video MediaStreamTrack
s sourced from a remote source:
Property Name | Values | Notes |
---|---|---|
width |
ConstrainULong
|
As a setting, this is the width, in pixels, of the latest frame received. |
height |
ConstrainULong
|
As a setting, this is the height, in pixels, of the latest frame received. |
frameRate |
ConstrainDouble
|
As a setting, this is an estimate of the frame rate based on recently received frames. |
aspectRatio |
ConstrainDouble
|
As a setting, this is the aspect ratio of the latest frame; this is the width in pixels divided by height in pixels as a double rounded to the tenth decimal place. |
This document does not define any constrainable properties to apply
to audio MediaStreamTrack
s sourced from a remote source.
This section is non-normative.
When two peers decide they are going to set up a connection to each other, they both go through these steps. The STUN/TURN server configuration describes a server they can use to get things like their public IP address or to set up NAT traversal. They also have to send data for the signaling channel to each other using the same out-of-band mechanism they used to establish that they were going to communicate in the first place.
const signaling = new SignalingChannel(); // handles JSON.stringify/parse
const constraints = {audio: true, video: true};
const configuration = {iceServers: [{urls: 'stun:stun.example.org'}]};
const pc = new RTCPeerConnection(configuration);
// send any ice candidates to the other peer
pc.onicecandidate = ({candidate}) => signaling.send({candidate});
// let the "negotiationneeded" event trigger offer generation
pc.onnegotiationneeded = async () => {
try {
await pc.setLocalDescription();
// send the offer to the other peer
signaling.send({description: pc.localDescription});
} catch (err) {
console.error(err);
}
};
pc.ontrack = ({track, streams}) => {
// once media for a remote track arrives, show it in the remote video element
track.onunmute = () => {
// don't set srcObject again if it is already set.
if (remoteView.srcObject) return;
remoteView.srcObject = streams[0];
};
};
// call start() to initiate
function start() {
addCameraMic();
}
// add camera and microphone to connection
async function addCameraMic() {
try {
// get a local stream, show it in a self-view and add it to be sent
const stream = await navigator.mediaDevices.getUserMedia(constraints);
for (const track of stream.getTracks()) {
pc.addTrack(track, stream);
}
selfView.srcObject = stream;
} catch (err) {
console.error(err);
}
}
signaling.onmessage = async ({data: {description, candidate}}) => {
try {
if (description) {
await pc.setRemoteDescription(description);
// if we got an offer, we need to reply with an answer
if (description.type == 'offer') {
if (!selfView.srcObject) {
// blocks negotiation on permission (not recommended in production code)
await addCameraMic();
}
await pc.setLocalDescription();
signaling.send({description: pc.localDescription});
}
} else if (candidate) {
await pc.addIceCandidate(candidate);
}
} catch (err) {
console.error(err);
}
};
When two peers decide they are going to set up a connection to each other and want to have the ICE, DTLS, and media connections "warmed up" such that they are ready to send and receive media immediately, they both go through these steps.
const signaling = new SignalingChannel(); // handles JSON.stringify/parse
const constraints = {audio: true, video: true};
const configuration = {iceServers: [{urls: 'stun:stun.example.org'}]};
let pc;
let audio;
let video;
let started = false;
// Call warmup() before media is ready, to warm-up ICE, DTLS, and media.
async function warmup(isAnswerer) {
pc = new RTCPeerConnection(configuration);
if (!isAnswerer) {
audio = pc.addTransceiver('audio');
video = pc.addTransceiver('video');
}
// send any ice candidates to the other peer
pc.onicecandidate = ({candidate}) => signaling.send({candidate});
// let the "negotiationneeded" event trigger offer generation
pc.onnegotiationneeded = async () => {
try {
await pc.setLocalDescription();
// send the offer to the other peer
signaling.send({description: pc.localDescription});
} catch (err) {
console.error(err);
}
};
pc.ontrack = async ({track, transceiver}) => {
try {
// once media for the remote track arrives, show it in the video element
event.track.onunmute = () => {
// don't set srcObject again if it is already set.
if (!remoteView.srcObject) {
remoteView.srcObject = new MediaStream();
}
remoteView.srcObject.addTrack(track);
}
if (isAnswerer) {
if (track.kind == 'audio') {
audio = transceiver;
} else if (track.kind == 'video') {
video = transceiver;
}
if (started) await addCameraMicWarmedUp();
}
} catch (err) {
console.error(err);
}
};
try {
// get a local stream, show it in a self-view and add it to be sent
selfView.srcObject = await navigator.mediaDevices.getUserMedia(constraints);
if (started) await addCameraMicWarmedUp();
} catch (err) {
console.error(err);
}
}
// call start() after warmup() to begin transmitting media from both ends
function start() {
signaling.send({start: true});
signaling.onmessage({data: {start: true}});
}
// add camera and microphone to already warmed-up connection
async function addCameraMicWarmedUp() {
const stream = selfView.srcObject;
if (audio && video && stream) {
await Promise.all([
audio.sender.replaceTrack(stream.getAudioTracks()[0]),
video.sender.replaceTrack(stream.getVideoTracks()[0]),
]);
}
}
signaling.onmessage = async ({data: {start, description, candidate}}) => {
if (!pc) warmup(true);
try {
if (start) {
started = true;
await addCameraMicWarmedUp();
} else if (description) {
await pc.setRemoteDescription(description);
// if we got an offer, we need to reply with an answer
if (description.type == 'offer') {
await pc.setLocalDescription();
signaling.send({description: pc.localDescription});
}
} else {
await pc.addIceCandidate(candidate);
}
} catch (err) {
console.error(err);
}
};
A client wants to send multiple RTP encodings (simulcast) to a server.
const signaling = new SignalingChannel(); // handles JSON.stringify/parse
const constraints = {audio: true, video: true};
const configuration = {'iceServers': [{'urls': 'stun:stun.example.org'}]};
let pc;
// call start() to initiate
async function start() {
pc = new RTCPeerConnection(configuration);
// let the "negotiationneeded" event trigger offer generation
pc.onnegotiationneeded = async () => {
try {
await pc.setLocalDescription();
// send the offer to the other peer
signaling.send({description: pc.localDescription});
} catch (err) {
console.error(err);
}
};
try {
// get a local stream, show it in a self-view and add it to be sent
const stream = await navigator.mediaDevices.getUserMedia(constraints);
selfView.srcObject = stream;
pc.addTransceiver(stream.getAudioTracks()[0], {direction: 'sendonly'});
pc.addTransceiver(stream.getVideoTracks()[0], {
direction: 'sendonly',
sendEncodings: [
{rid: 'q', scaleResolutionDownBy: 4.0}
{rid: 'h', scaleResolutionDownBy: 2.0},
{rid: 'f'},
]
});
} catch (err) {
console.error(err);
}
}
signaling.onmessage = async ({data: {description, candidate}}) => {
try {
if (description) {
await pc.setRemoteDescription(description);
// if we got an offer, we need to reply with an answer
if (description.type == 'offer') {
await pc.setLocalDescription();
signaling.send({description: pc.localDescription});
}
} else if (candidate) {
await pc.addIceCandidate(candidate);
}
} catch (err) {
console.error(err);
}
};
This example shows how to create an
object and
perform the offer/answer exchange required to connect the channel
to the other peer. The RTCDataChannel
is used in the context of
a simple chat application using an RTCDataChannel
input
field for
user input.
const signaling = new SignalingChannel(); // handles JSON.stringify/parse
const configuration = {iceServers: [{urls: 'stun:stun.example.org'}]};
let pc, channel;
// call start() to initiate
function start() {
pc = new RTCPeerConnection(configuration);
// send any ice candidates to the other peer
pc.onicecandidate = ({candidate}) => signaling.send({candidate});
// let the "negotiationneeded" event trigger offer generation
pc.onnegotiationneeded = async () => {
try {
await pc.setLocalDescription();
// send the offer to the other peer
signaling.send({description: pc.localDescription});
} catch (err) {
console.error(err);
}
};
// create data channel and setup chat using "negotiated" pattern
channel = pc.createDataChannel('chat', {negotiated: true, id: 0});
channel.onopen = () => input.disabled = false;
channel.onmessage = ({data}) => showChatMessage(data);
input.onkeypress = ({keyCode}) => {
// only send when user presses enter
if (keyCode != 13) return;
channel.send(input.value);
}
}
signaling.onmessage = async ({data: {description, candidate}}) => {
if (!pc) start(false);
try {
if (description) {
await pc.setRemoteDescription(description);
// if we got an offer, we need to reply with an answer
if (description.type == 'offer') {
await pc.setLocalDescription();
signaling.send({description: pc.localDescription});
}
} else if (candidate) {
await pc.addIceCandidate(candidate);
}
} catch (err) {
console.error(err);
}
};
This shows an example of one possible call flow between two browsers. This does not show the procedure to get access to local media or every callback that gets fired but instead tries to reduce it down to only show the key events and messages.
Examples assume that sender is an
.
RTCRtpSender
Sending the DTMF signal "1234" with 500 ms duration per tone:
if (sender.dtmf.canInsertDTMF) {
const duration = 500;
sender.dtmf.insertDTMF('1234', duration);
} else {
console.log('DTMF function not available');
}
Send the DTMF signal "123" and abort after sending "2".
async function sendDTMF() {
if (sender.dtmf.canInsertDTMF) {
sender.dtmf.insertDTMF('123');
await new Promise(r => sender.dtmf.ontonechange = e => e.tone == '2' && r());
// empty the buffer to not play any tone after "2"
sender.dtmf.insertDTMF('');
} else {
console.log('DTMF function not available');
}
}
Send the DTMF signal "1234", and light up the active key using
lightKey(key)
while the tone is playing
(assuming that lightKey("")
will darken
all the keys):
const wait = ms => new Promise(resolve => setTimeout(resolve, ms));
if (sender.dtmf.canInsertDTMF) {
const duration = 500; // ms
sender.dtmf.insertDTMF(sender.dtmf.toneBuffer + '1234', duration);
sender.dtmf.ontonechange = async ({tone}) => {
if (!tone) return;
lightKey(tone); // light up the key when playout starts
await wait(duration);
lightKey(''); // turn off the light after tone duration
};
} else {
console.log('DTMF function not available');
}
It is always safe to append to the tone buffer. This example appends before any tone playout has started as well as during playout.
if (sender.dtmf.canInsertDTMF) {
sender.dtmf.insertDTMF('123');
// append more tones to the tone buffer before playout has begun
sender.dtmf.insertDTMF(sender.dtmf.toneBuffer + '456');
sender.dtmf.ontonechange = ({tone}) => {
// append more tones when playout has begun
if (tone != '1') return;
sender.dtmf.insertDTMF(sender.dtmf.toneBuffer + '789');
};
} else {
console.log('DTMF function not available');
}
Send a 1-second "1" tone followed by a 2-second "2" tone:
if (sender.dtmf.canInsertDTMF) {
sender.dtmf.ontonechange = ({tone}) => {
if (tone == '1') {
sender.dtmf.insertDTMF(sender.dtmf.toneBuffer + '2', 2000);
}
};
sender.dtmf.insertDTMF(sender.dtmf.toneBuffer + '1', 1000);
} else {
console.log('DTMF function not available');
}
Perfect negotiation is a recommended pattern to manage negotiation
transparently, abstracting this asymmetric task away from the rest of
an application. This pattern has advantages over one side always
being the offerer, as it lets applications operate on both peer
connection objects simultaneously without risk of glare (an offer
coming in outside of "
" state). The rest
of the application may use any and all modification methods and
attributes, without worrying about signaling state races.
stable
It designates different roles to the two peers, with behavior to resolve signaling collisions between them:
The polite peer uses rollback to avoid collision with an incoming offer.
The impolite peer ignores an incoming offer when this would collide with its own.
Together, they manage signaling for the rest of the application in a
manner that doesn't deadlock. The example assumes a
polite
boolean variable indicating the designated role:
const signaling = new SignalingChannel(); // handles JSON.stringify/parse
const constraints = {audio: true, video: true};
const configuration = {iceServers: [{urls: 'stun:stun.example.org'}]};
const pc = new RTCPeerConnection(configuration);
// call start() anytime on either end to add camera and microphone to connection
async function start() {
try {
const stream = await navigator.mediaDevices.getUserMedia(constraints);
for (const track of stream.getTracks()) {
pc.addTrack(track, stream);
}
selfView.srcObject = stream;
} catch (err) {
console.error(err);
}
}
pc.ontrack = ({track, streams}) => {
// once media for a remote track arrives, show it in the remote video element
track.onunmute = () => {
// don't set srcObject again if it is already set.
if (remoteView.srcObject) return;
remoteView.srcObject = streams[0];
};
};
// - The perfect negotiation logic, separated from the rest of the application ---
// keep track of some negotiation state to prevent races and errors
let makingOffer = false;
let ignoreOffer = false;
let isSettingRemoteAnswerPending = false;
// send any ice candidates to the other peer
pc.onicecandidate = ({candidate}) => signaling.send({candidate});
// let the "negotiationneeded" event trigger offer generation
pc.onnegotiationneeded = async () => {
try {
makingOffer = true;
await pc.setLocalDescription();
signaling.send({description: pc.localDescription});
} catch (err) {
console.error(err);
} finally {
makingOffer = false;
}
};
signaling.onmessage = async ({data: {description, candidate}}) => {
try {
if (description) {
// An offer may come in while we are busy processing SRD(answer).
// In this case, we will be in "stable" by the time the offer is processed
// so it is safe to chain it on our Operations Chain now.
const readyForOffer =
!makingOffer &&
(pc.signalingState == "stable" || isSettingRemoteAnswerPending);
const offerCollision = description.type == "offer" && !readyForOffer;
ignoreOffer = !polite && offerCollision;
if (ignoreOffer) {
return;
}
isSettingRemoteAnswerPending = description.type == "answer";
await pc.setRemoteDescription(description); // SRD rolls back as needed
isSettingRemoteAnswerPending = false;
if (description.type == "offer") {
await pc.setLocalDescription();
signaling.send({description: pc.localDescription});
}
} else if (candidate) {
try {
await pc.addIceCandidate(candidate);
} catch (err) {
if (!ignoreOffer) throw err; // Suppress ignored offer's candidates
}
}
} catch (err) {
console.error(err);
}
}
Note that this is timing sensitive, and deliberately uses versions of
(without arguments) and
setLocalDescription
(with implicit rollback)
to avoid races with other signaling messages being serviced.
setRemoteDescription
The ignoreOffer variable is needed, because the
object on the impolite side is never told about
ignored offers. We must therefore suppress errors from incoming
candidates belonging to such offers.
RTCPeerConnection
Some operations throw or fire
. This is an extension of
RTCError
DOMException
that carries additional WebRTC-specific information.
RTCError
Interface
WebIDL[Exposed=Window] interfaceRTCError
: DOMException {constructor
(RTCErrorInit
init, optional DOMString message = ""); readonly attributeRTCErrorDetailType
errorDetail
; readonly attribute long?sdpLineNumber
; readonly attribute long?sctpCauseCode
; readonly attribute unsigned long?receivedAlert
; readonly attribute unsigned long?sentAlert
; };
constructor()
Run the following steps:
Let init be the constructor's first argument.
Let message be the constructor's second argument.
Let e be a new
object.
RTCError
Invoke the DOMException
constructor of e
with the
argument set to
message and the message
name
argument
set to "OperationError"
.
This name does not have a mapping to a legacy code so
e.code
will return 0.
Set all
attributes of e to the
value of the corresponding attribute in init if
it is present, otherwise set it to RTCError
null
.
Return e.
errorDetail
of type RTCErrorDetailType, readonly
The WebRTC-specific error code for the type of error that occurred.
sdpLineNumber
of type long, readonly, nullable
If
is
"errorDetail
" this is the line
number where the error was detected (the first line has line
number 1).
sdp-syntax-error
sctpCauseCode
of type long, readonly, nullable
If
is
"errorDetail
" this is the SCTP cause
code of the failed SCTP negotiation.
sctp-failure
receivedAlert
of type unsigned long, readonly, nullable
If
is
"errorDetail
" and a fatal DTLS alert
was received, this is the value of the DTLS alert received.
dtls-failure
sentAlert
of type unsigned long, readonly, nullable
If
is
"errorDetail
" and a fatal DTLS alert
was sent, this is the value of the DTLS alert sent.
dtls-failure
All attributes defined in
are marked at risk due
to lack of implementation (RTCError
,
errorDetail
, sdpLineNumber
, sctpCauseCode
and
receivedAlert
). This does not include attributes inherited
from sentAlert
DOMException
.
RTCErrorInit
Dictionary
WebIDLdictionaryRTCErrorInit
{ requiredRTCErrorDetailType
errorDetail
; longsdpLineNumber
; longsctpCauseCode
; unsigned longreceivedAlert
; unsigned longsentAlert
; };
The errorDetail
, sdpLineNumber
, sctpCauseCode
,
receivedAlert
and sentAlert
members of
have the same
definitions as the attributes of the same name of RTCErrorInit
.
RTCError
RTCErrorDetailType
Enum
WebIDLenumRTCErrorDetailType
{ "data-channel-failure
", "dtls-failure
", "fingerprint-failure
", "sctp-failure
", "sdp-syntax-error
", "hardware-encoder-not-available
", "hardware-encoder-error
" };
Enumeration description | |
---|---|
data-channel-failure
|
The data channel has failed. |
dtls-failure
|
The DTLS negotiation has failed or the connection has been
terminated with a fatal error. The
contains information relating to the nature of error. If a
fatal DTLS alert was received, the
attribute is set to the value of the DTLS alert received. If a
fatal DTLS alert was sent, the attribute
is set to the value of the DTLS alert sent.
|
fingerprint-failure
|
The 's remote certificate did not match any
of the fingerprints provided in the SDP. If the remote peer
cannot match the local certificate against the provided
fingerprints, this error is not generated. Instead a
"bad_certificate" (42) DTLS alert might be received from the
remote peer, resulting in a
" ".
|
sctp-failure
|
The SCTP negotiation has failed or the connection has been
terminated with a fatal error. The
attribute is set to the SCTP cause code.
|
sdp-syntax-error
|
The SDP syntax is not valid. The
attribute is set to the line number in the SDP where the syntax
error was detected.
|
hardware-encoder-not-available
|
The hardware encoder resources required for the requested operation are not available. |
hardware-encoder-error
|
The hardware encoder does not support the provided parameters. |
RTCErrorEvent
Interface
The
interface is defined for cases when an
RTCErrorEvent
is raised as an event:
RTCError
WebIDL[Exposed=Window] interfaceRTCErrorEvent
: Event {constructor
(DOMString type,RTCErrorEventInit
eventInitDict); [SameObject] readonly attributeRTCError
error
; };
constructor()
Constructs a new
.
RTCErrorEvent
error
of type RTCError
, readonly
The
describing the error that triggered the event.
RTCError
RTCErrorEventInit
Dictionary
WebIDL dictionaryRTCErrorEventInit
: EventInit { requiredRTCError
error
; };
error
of type RTCError
The
describing the error associated with the event
(if any).
RTCError
This section is non-normative.
The following events fire on
objects:
RTCDataChannel
Event name | Interface | Fired when... |
---|---|---|
open
|
Event
|
The object's underlying data transport
has been established (or re-established).
|
message
|
MessageEvent
[html]
|
A message was successfully received. |
bufferedamountlow
|
Event
|
The object's
decreases from above its
to less than or
equal to its .
|
error
|
|
An error occurred on the data channel. |
closing
|
Event
|
The object transitions to the
" " state
|
close
|
Event
|
The object's underlying data transport
has been closed.
|
The following events fire on
objects:
RTCPeerConnection
Event name | Interface | Fired when... |
---|---|---|
track
|
|
New incoming media has been negotiated for a specific
, and that receiver's
has been added to any associated remote MediaStream s.
|
negotiationneeded
|
Event
|
The browser wishes to inform the application that session negotiation needs to be done (i.e. a createOffer call followed by setLocalDescription). |
signalingstatechange
|
Event
|
The signaling state has changed. This state change is the
result of either or
being invoked.
|
iceconnectionstatechange
|
Event
|
The 's ICE connection state has
changed.
|
icegatheringstatechange
|
Event
|
The 's ICE gathering state has
changed.
|
icecandidate
|
|
A new is made available to the script.
|
connectionstatechange
|
Event
|
The .
has changed.
|
icecandidateerror
|
|
A failure occured when gathering ICE candidates. |
datachannel |
|
A new is dispatched to the script in response
to the other peer creating a channel.
|
The following events fire on
objects:
RTCDTMFSender
Event name | Interface | Fired when... |
---|---|---|
tonechange
|
|
The object has either just begun playout of a
tone (returned as the attribute)
or just ended the playout of tones in the
(returned as an empty value in the
attribute).
|
The following events fire on
objects:
RTCIceTransport
Event name | Interface | Fired when... |
---|---|---|
statechange
|
Event
|
The state changes.
|
gatheringstatechange
|
Event
|
The gathering state changes.
|
selectedcandidatepairchange
|
Event
|
The 's selected candidate pair changes.
|
The following events fire on
objects:
RTCDtlsTransport
Event name | Interface | Fired when... |
---|---|---|
statechange
|
Event
|
The state changes.
|
error
|
|
An error occurred on the (either
" " or
" ").
|
The following events fire on
objects:
RTCSctpTransport
Event name | Interface | Fired when... |
---|---|---|
statechange
|
Event
|
The state changes.
|
This section is non-normative.
This section is non-normative; it specifies no new behaviour, but instead summarizes information already present in other parts of the specification. The overall security considerations of the general set of APIs and protocols used in WebRTC are described in [RFC8827].
This document extends the Web platform with the ability to set up real-time, direct communication between browsers and other devices, including other browsers.
This means that data and media can be shared between applications running in different browsers, or between an application running in the same browser and something that is not a browser, something that is an extension to the usual barriers in the Web model against sending data between entities with different origins.
The WebRTC specification provides no user prompts or chrome indicators for communication; it assumes that once the Web page has been allowed to access media, it is free to share that media with other entities as it chooses. Peer-to-peer exchanges of data view WebRTC datachannels can thus occur without any user explicit consent or involvement, similarly as a server-mediated exchange (e.g. via Web Sockets) could occur without user involvement.
Even without WebRTC, the Web server providing a Web application will know the public IP address to which the application is delivered. Setting up communications exposes additional information about the browser’s network context to the web application, and may include the set of (possibly private) IP addresses available to the browser for WebRTC use. Some of this information has to be passed to the corresponding party to enable the establishment of a communication session.
Revealing IP addresses can leak location and means of connection; this can be sensitive. Depending on the network environment, it can also increase the fingerprinting surface and create persistent cross-origin state that cannot easily be cleared by the user.
A connection will always reveal the IP addresses proposed for
communication to the corresponding party. The application can limit
this exposure by choosing not to use certain addresses using the
settings exposed by the
dictionary, and by
using relays (for instance TURN servers) rather than direct
connections between participants. One will normally assume that the
IP address of TURN servers is not sensitive information. These
choices can for instance be made by the application based on whether
the user has indicated consent to start a media connection with the
other party.
RTCIceTransportPolicy
Mitigating the exposure of IP addresses to the application itself requires limiting the IP addresses that can be used, which will impact the ability to communicate on the most direct path between endpoints. Browsers are encouraged to provide appropriate controls for deciding which IP addresses are made available to applications, based on the security posture desired by the user. The choice of which addresses to expose is controlled by local policy (see [RFC8828] for details).
Since the browser is an active platform executing in a trusted network environment (inside the firewall), it is important to limit the damage that the browser can do to other elements on the local network, and it is important to protect data from interception, manipulation and modification by untrusted participants.
Mitigations include:
These measures are specified in the relevant IETF documents.
The fact that communication is taking place cannot be hidden from adversaries that can observe the network, so this has to be regarded as public information.
Communication certificates may be opaquely shared using
postMessage
(message, options)
in anticipation of future needs. User
agents are strongly encouraged to isolate the private keying material
these objects hold a handle to, from the processes that have access
to the
objects, to reduce memory attack surface.
RTCCertificate
As described above, the list of IP addresses exposed by the WebRTC API can be used as a persistent cross-origin state.
Beyond IP addresses, the WebRTC API exposes information about the
underlying media system via the
.RTCRtpSender
and
getCapabilities
.RTCRtpReceiver
methods,
including detailed and ordered information about the codecs that the
system is able to produce and consume. A subset of that information
is likely to be represented in the SDP session descriptions
generated, exposed and transmitted during session negotiation. That
information is in most cases persistent across time and origins, and
increases the fingerprint surface of a given device.
getCapabilities
When establishing DTLS connections, the WebRTC API can generate certificates that can be persisted by the application (e.g. in IndexedDB). These certificates are not shared across origins, and get cleared when persistent storage is cleared for the origin.
guards against malformed
and invalid SDP by throwing exceptions, but makes no attempt to guard
against SDP that might be unexpected by the application. Setting the
remote description can cause significant resources to be allocated
(including image buffers and network ports), media to start flowing
(which may have privacy and bandwidth implications) among other
things. An application that does not guard against malicious SDP
could be at risk of resource deprivation, unintentionally allowing
incoming media or at risk of not having certain events fire like
setRemoteDescription
if the other endpoint does not
negotiate sending. Applications need to be on guard against
malevolent SDP.
ontrack
This section is non-normative.
The WebRTC 1.0 specification exposes an API to control protocols (defined within the IETF) necessary to establish real-time audio, video and data exchange.
The Telecommunications Device for the Deaf (TDD/TTY) enables individuals who are hearing or speech impaired (among others) to communicate over telephone lines. Real-Time Text, defined in [RFC4103], utilizes T.140 encapsulated in RTP to enable the transition from TDD/TTY devices to IP-based communications, including emergency communication with Public Safety Access Points (PSAP).
Since Real-Time Text requires the ability to send and receive data in near real time, it can be best supported via the WebRTC 1.0 data channel API. As defined by the IETF, the data channel protocol utilizes the SCTP/DTLS/UDP protocol stack, which supports both reliable and unreliable data channels. The IETF chose to standardize SCTP/DTLS/UDP over proposals for an RTP data channel which relied on SRTP key management and were focused on unreliable communications.
Since the IETF chose a different approach than the RTP data channel as part of the WebRTC suite of protocols, as of the time of this publication there is no standardized way for the WebRTC APIs to directly support Real-Time Text as defined at IETF and implemented in U.S. (FCC) regulations. The WebRTC working Group will evaluate whether the developing IETF protocols in this space warrant direct exposure in the browser APIs and is looking for input from the relevant user communities on this potential gap.
Within the IETF MMUSIC Working Group, work is ongoing to enable Real-time text to be sent over the WebRTC data channel, allowing gateways to be deployed to translate between the SCTP data channel protocol and RFC 4103 Real-Time Text. This work, once completed, is expected to enable a unified and interoperable approach for integrating real-time text in WebRTC user-agents (including browsers) - through a gateway or otherwise.
At the time of this publication, gateways that enable effective RTT support in WebRTC clients can be developed e.g. through a custom WebRTC data channel. This is deemed sufficient until such time as future standardized gateways are enabled via IETF protocols such as the SCTP data channel protocol and RFC 4103 Real-Time Text. This will need to be defined at IETF in conjunction with related work at W3C groups to effectively and consistently standardise RTT support internationally.
The editors wish to thank the Working Group chairs and Team Contact, Harald Alvestrand, Stefan Håkansson, Erik Lagerway and Dominique Hazaël-Massieux, for their support. Substantial text in this specification was provided by many people including Martin Thomson, Harald Alvestrand, Justin Uberti, Eric Rescorla, Peter Thatcher, Jan-Ivar Bruaroey and Peter Saint-Andre. Dan Burnett would like to acknowledge the significant support received from Voxeo and Aspect during the development of this specification.
The
and RTCRtpSender
objects were initially
described in the W3C ORTC
CG, and have been adapted for use in this specification.
RTCRtpReceiver