This document defines a set of ECMAScript APIs in WebIDL to allow media
to be sent to and received from another browser or device implementing the
appropriate set of real-time protocols. This specification is being
developed in conjunction with a protocol specification developed by the
IETF RTCWEB group and an API specification to get access to local media
devices.
Status of This Document
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.
The specification is feature complete and is expected to be stable with no further substantive change. Since the previous Candidate Recommendation, the following substantive changes have been brought to the specification:
the specification allows administrative prohibiting of ICE candidates to help prevent WebRTC to be used for portscanning
the support for setting multiple DTLS certificates has been deprecated
voiceActivityFlag has been marked at risk for lack of implementation
the state machine when closing a PeerConnection has been clarified
To go into Proposed Recommendation status, the group expects to demonstrate implementation of each feature in at least two deployed browsers, and at least one implementation of each optional feature. Mandatory feature with only one implementation may be marked as optional in a revised Candidate Recommendation where applicable.
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).
W3C publishes a Candidate Recommendation to indicate that the document is believed to be
stable and to encourage implementation by the developer community. This Candidate
Recommendation is expected to advance to Proposed Recommendation no earlier than
24 September 2020.
Publication as a Candidate Recommendation does not imply endorsement by the W3C
Membership. This is a draft document and may be updated, replaced or
obsoleted by other documents at any time. It is inappropriate to cite this
document as other than work in progress.
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.
There are a number of facets to peer-to-peer communications and
video-conferencing in HTML covered by this specification:
Connecting to remote peers using NAT-traversal technologies such as
ICE, STUN, and TURN.
Sending the locally-produced tracks to remote peers and receiving
tracks from remote peers.
Sending arbitrary data directly to remote peers.
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
[RTCWEB-OVERVIEW] and [RTCWEB-SECURITY].
2. Conformance
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.
3. Terminology
The EventHandler
interface, representing a callback used for event handlers, is defined in [HTML].
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.
4. Peer-to-peer connections
4.1 Introduction
An RTCPeerConnection instance allows an
application to establish peer-to-peer communications with another
RTCPeerConnection instance in another browser, or to
another endpoint implementing the required protocols. Communications are coordinated by the
exchange of control messages (called a signaling protocol) over a
signaling channel which is provided by unspecified means, but generally
by a script in the page via the server, e.g. using
Web Sockets or XMLHttpRequest [xhr].
4.2 Configuration
4.2.1 RTCConfiguration Dictionary
The RTCConfiguration defines a set of parameters to
configure how the peer-to-peer communication established via
RTCPeerConnection is established or
re-established.
A set of certificates that the
RTCPeerConnection uses to authenticate.
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
RTCPeerConnection implementation selects which of
the certificates is used for a given connection; how
certificates are selected is outside the scope of this
specification.
Note
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 RTCPeerConnection
instance.
This option allows applications to establish key continuity.
An RTCCertificate can be persisted in
[INDEXEDDB] and reused. Persistence and reuse also avoids the
cost of key generation.
The value for this configuration option cannot change after
its value is initially selected.
iceCandidatePoolSize of type
octet, defaulting to
0
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 RTCIceServer object represents a
TURN server, and credentialType is
"password", then this attribute specifies the
username to use with that TURN server.
If this RTCIceServer object represents a
TURN server, then this attribute specifies how
credential should be used when that TURN server
requests authorization.
As described in [JSEP] (section 4.1.1.), if the
iceTransportPolicy member of
the RTCConfiguration is specified, it defines the
ICE candidate policy
[JSEP] (section 3.5.3.) the browser uses to surface the permitted candidates
to the application; only these candidates will be used for connectivity
checks.
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
RTCIceCandidate.address.
4.2.5 RTCBundlePolicy Enum
As described in [JSEP] (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.
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.
4.2.6 RTCRtcpMuxPolicy Enum
As described in [JSEP] (section 4.1.1.), the
RTCRtcpMuxPolicy affects what ICE candidates are gathered to support
non-multiplexed RTCP. The only value defined in this spec is
"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.
4.2.7 Offer/Answer Options
These dictionaries describe the options that can be used to control
the offer/answer creation process.
When the value of this dictionary member is true,
or the relevant RTCPeerConnection 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
currentLocalDescription attribute's
SDP). Applying the generated description will restart ICE, as
described in section 9.1.1.1 of [ICE].
Performing an ICE restart is recommended when
iceConnectionState transitions to
"failed".
An application may additionally choose to listen for the
iceConnectionState transition to
"disconnected"
and then use other sources of information (such as using
getStats to measure if the number of bytes sent
or received over the next couple of seconds increases) to determine
whether an ICE restart is advisable.
The RTCAnswerOptions dictionary describe options specific to session description of type "answer" (none in this version of the specification).
Note that if an RTCIceTransport 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.
4.4 RTCPeerConnection Interface
The [JSEP] specification, as a whole, describes the details of how
the RTCPeerConnection operates. References to
specific subsections of [JSEP] are provided as appropriate.
configuration.iceServers 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.
An RTCPeerConnection 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.
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 message attribute set to an appropriate description.
Else, generate one or more new RTCCertificate instances
with this RTCPeerConnection instance and store them. This MAY happen
asynchronously and the value of certificates remains
undefined for the subsequent steps. As noted in Section 4.3.2.3 of
[RTCWEB-SECURITY], 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.
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.
4.4.1.2 Chain an asynchronous operation
An RTCPeerConnection 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.
An RTCPeerConnection object has an aggregated
connection state. Whenever the state of an
RTCDtlsTransport changes or when the
[[IsClosed]] slot turns true, the user agent MUST
update the connection state by queueing a task that runs the
following steps:
The null candidate event is fired to ensure
legacy compatibility. New code should monitor the gathering state
of RTCIceTransport and/or
RTCPeerConnection.
Let jsepSetOfTransceivers be a shallow copy of
connection's set of transceivers.
In parallel, start the process to apply description
as described in [JSEP] (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 the content of description is not
valid SDP syntax, then rejectp with an
RTCError (with errorDetail
set to "sdp-syntax-error" and the sdpLineNumber
attribute set to the line number in the SDP where
the syntax error was detected) and abort these steps.
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 "offer",
then if any 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 "answer", and it
initiates the closure of an existing SCTP association, as
defined in [SCTP-SDP], Sections 10.3 and 10.4, set the
value of connection.[[SctpTransport]] to
null.
Let trackEventInits, muteTracks,
addList, removeList and
errorList be empty lists.
If description is of type "answer" or
"pranswer", then run the following steps:
If description negotiates the DTLS role of
the SCTP transport, then for each
RTCDataChannel, channel,
with a nullid,
run the following step:
Give channel a new ID generated
according to [RTCWEB-DATA-PROTOCOL]. If no
available ID could be generated, set
channel.[[ReadyState]] to
"closed", and add channnel
to errorList.
If description is not of type "rollback", then
run the following steps:
If remote is false, then run
the following steps for each media description in
description:
Set
transceiver.[[Receiver]].[[ReceiveCodecs]]
to the codecs that description
negotiates for receiving and which the user
agent is currently prepared to receive.
Note
If the direction is
"sendonly" or "inactive",
the receiver is not prepared to receive
anything, and the list will be empty.
If description is of type
"answer" or "pranswer",
then run the following steps:
Otherwise, (if remote is true)
run the following steps for each media description
in description:
If the description is of type "offer"
and contains a request to receive simulcast, use the order
of the rid values specified in the simulcast attribute to create
an RTCRtpEncodingParameters dictionary for
each of the simulcast layers, populating the rid member
according to the corresponding rid value, and let sendEncodings
be the list containing the created dictionaries. Otherwise,
let sendEncodings be an empty list.
Let supportedEncodings be the maximum number of encodings
that the implementation can support. If the length of
sendEncodings is greater than supportedEncodings,
truncate sendEncodings so that its length is
supportedEncodings.
If sendEncodings is non-empty, set each
encoding's
scaleResolutionDownBy
to 2^(length of sendEncodings -
encoding index - 1).
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 [MMUSIC-SIMULCAST] of
each simulcast layer by setting the
active
member on the corresponding dictionaries in
transceiver.[[Sender]].[[SendEncodings]]
to true for unpaused or to false for paused.
If direction is "sendrecv"
or "recvonly", 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.
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
"answer" or "pranswer", then run
the following steps:
Set
transceiver.[[Sender]].[[SendCodecs]]
to the codecs that description
negotiates for sending and which the user agent
is currently capable of sending.
Set the ICE Agent's ICE transports setting to
the value of configuration.iceTransportPolicy. As defined
in [JSEP] (section 4.1.16.), 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.
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.
Set the ICE
Agent's ICE servers
list to validatedServers.
As defined in [JSEP] (section 4.1.16.), 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.
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.
It represents the local description that was successfully
negotiated the last time the RTCPeerConnection
transitioned into the stable state plus any local candidates
that have been generated by the ICE Agent since the offer
or answer was created.
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 RTCPeerConnection is in
the stable state, the value is null.
It represents the last remote description that was successfully
negotiated the last time the RTCPeerConnection
transitioned into the stable state plus any remote candidates
that have been supplied via addIceCandidate() since the
offer or answer was created.
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
RTCPeerConnection is in the stable state, the
value is null.
canTrickleIceCandidates of type boolean, readonly, nullable
The canTrickleIceCandidates
attribute indicates whether the remote peer is able to accept
trickled ICE candidates [TRICKLE-ICE]. The value is
determined based on whether a remote description indicates
support for trickle ICE, as defined in [JSEP] (section 4.1.15.). Prior to the completion of
setRemoteDescription, this
value is null.
The createOffer method generates a blob of SDP that contains
an RFC 3264 offer with the supported configurations for the
session, including descriptions of the local
MediaStreamTracks attached to this
RTCPeerConnection, 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.
If a system has limited resources (e.g. a finite number of
decoders), createOffer needs to return an offer
that reflects the current state of the system, so that
setLocalDescription 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.
Creating the SDP
MUST follow the appropriate process for generating an offer
described in [JSEP], except the user agent MUST treat a
stopping transceiver as
stopped for the
purposes of JSEP 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
createOffer 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.
The generated SDP will also contain the ICE agent's
usernameFragment,
password and
ICE options (as defined in [ICE], Section 14) and may also contain
any local candidates that have been gathered by the agent.
The certificates value in configuration
for the RTCPeerConnection 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.
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
RTCPeerConnection object on which the
method was invoked.
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 [JSEP] (section 4.1.6.).
Given the information that was obtained from previous
inspection, the current state of connection
and its RTCRtpTransceivers,
generate an SDP offer, sdpString, as described
in [JSEP] (section 5.2.).
As described in [BUNDLE] (Section 7), if bundling
is used (see RTCBundlePolicy) 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.
The filtering MUST NOT change the order of the codec
preferences.
If the length of the [[SendEncodings]] slot
of the RTCRtpSender is
larger than 1, then
for each encoding given in [[SendEncodings]]
of the RTCRtpSender, add an 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 encodings field. No RID restrictions
are set.
Note
[SDP-SIMULCAST] 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
RTCSessionDescriptionInit dictionary
with its type member initialized to the string
"offer" and its sdp member
initialized to sdpString.
The createAnswer method generates an [SDP]
answer with the supported configuration for the session that is
compatible with the parameters in the remote configuration.
Like createOffer, the returned blob of SDP contains
descriptions of the local MediaStreamTracks
attached to this RTCPeerConnection, 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.
Like createOffer, the
returned description SHOULD reflect the current state of the
system. 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.
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 [JSEP].
The generated SDP will also contain the ICE agent's
usernameFragment,
password and
ICE options (as defined in [ICE], Section 14) and may also contain
any local candidates that have been gathered by the agent.
The certificates value in configuration
for the RTCPeerConnection 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.
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 [JSEP] (section 4.1.7.).
Given the information that was obtained from previous
inspection and the current state of connection
and its RTCRtpTransceivers,
generate an SDP answer, sdpString, as described
in [JSEP] (section 5.3.).
The codec preferences of an m= section's
associated transceiver is said to be the value of the
RTCRtpTransceiver.[[PreferredCodecs]] with the following
filtering applied (or said not to be set if
[[PreferredCodecs]] is empty):
The filtering MUST NOT change the order of the codec
preferences.
If the length of the [[SendEncodings]] slot
of the RTCRtpSender is
larger than 1, then
for each encoding given in [[SendEncodings]]
of the RTCRtpSender, add an 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 encodings field. No RID restrictions
are set.
Let answer be a newly created
RTCSessionDescriptionInit dictionary
with its type member initialized to the string
"answer" and its sdp member
initialized to sdpString.
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 RTCPeerConnectionMUST 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.
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.
The addIceCandidate
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 candidate member. The only
members of the argument used by this method are candidate, sdpMid, sdpMLineIndex, and
usernameFragment; the rest
are ignored. When the method is invoked, the user agent MUST
run the following steps:
Let candidate be the method's argument.
Let connection be the
RTCPeerConnection object on which the
method was invoked.
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.
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 iceTransportPolicy member of the RTCConfiguration is relay, candidates requiring external
resolution, such as mDNS candidates and DNS candidates,
MUST be prohibited.
Note
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 restartIce method tells the
RTCPeerConnection that ICE should be
restarted. Subsequent calls to createOffer will
create descriptions that will restart ICE, as described in
section 9.1.1.1 of [ICE].
When this method is invoked, the user agent MUST run the
following steps:
Let connection be the
RTCPeerConnection on which the method
was invoked.
The setConfiguration method updates the
configuration of this RTCPeerConnection
object. This includes changing the configuration of the ICE
Agent. As noted in [JSEP] (section 3.5.1.), when the ICE
configuration changes in a way that requires a new gathering
phase, an ICE restart is required.
When the setConfiguration method is
invoked, the user agent MUST run the following steps:
Let connection be the
RTCPeerConnection on which the method
was invoked.
Let transceivers be the result of executing the
CollectTransceivers algorithm. For every
RTCRtpTransceivertransceiver in
transceivers, run the following steps:
If transceiver.[[Stopped]]
is true, abort these sub steps.
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.
Note
The addStream method that used to exist on
RTCPeerConnection is easy to polyfill as:
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 RTCPeerConnection. Developers are encouraged to
use the RTCRtpTransceiver API instead.
When createOffer
is called with any of the legacy options specified in this section,
run the followings steps instead of the regular createOffer
steps:
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.
4.4.4 Garbage collection
An RTCPeerConnection 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 true, no such event handler can be triggered and
it is therefore safe to garbage collect the object.
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.
An RTCSdpType of "offer" indicates
that a description MUST be treated as an [SDP] offer.
pranswer
An RTCSdpType of "pranswer"
indicates that a description MUST be treated as an [SDP]
answer, but not a final answer. A description used as an SDP
pranswer may be applied as a response to an SDP
offer, or an update to a previously sent SDP pranswer.
answer
An RTCSdpType of "answer"
indicates that a description MUST be treated as an [SDP]
final answer, and the offer-answer exchange MUST be
considered complete. A description used as an SDP answer may
be applied as a response to an SDP offer or as an update to a
previously sent SDP pranswer.
rollback
An RTCSdpType of "rollback"
indicates that a description MUST be treated as canceling the
current SDP negotiation and moving the SDP [SDP] offer
back to what it was in the previous stable state. Note
the local or remote SDP descriptions in the previous stable
state could be null if there has not yet been a successful
offer-answer negotiation. An "answer" or
"pranswer" cannot be rolled back.
The RTCSessionDescription()
constructor takes a dictionary argument,
description, whose content is used to
initialize the new RTCSessionDescription
object. This constructor is deprecated; it exists for
legacy compatibility reasons only.
The string representation of the SDP [SDP]; if
type is "rollback", this member is unused.
4.7 Session Negotiation Model
Many changes to state of an RTCPeerConnection 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
negotiationneeded event. This event is fired according to
the state of the connection's negotiation-needed flag,
represented by a [[NegotiationNeeded]] internal slot.
4.7.1 Setting Negotiation-Needed
This section is non-normative.
If an operation is performed on an
RTCPeerConnection that requires signaling, the
connection will be marked as needing negotiation. Examples of such
operations include adding or stopping an
RTCRtpTransceiver, or adding the first RTCDataChannel.
Internal changes within the implementation can also result in the
connection being marked as needing negotiation.
The negotiation-needed flag is cleared when an
RTCSessionDescription of type "answer" is applied, and the supplied description matches
the state of the
RTCRtpTransceivers and
RTCDataChannels that currently exist on the
RTCPeerConnection. Specifically, this means that all
non-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.
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 MUSTupdate the negotiation-needed flag.
To update the negotiation-needed flag for
connection, run the following steps:
The task queueing prevents negotiationneeded from
firing prematurely, in the common situation where multiple
modifications to connection are being made at once.
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
"sendrecv" or "sendonly",
and the associated m= section in description
either doesn't contain a single 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 "answer",
and the direction of the associated m=
section in the description does not match
transceiver.[[Direction]]
intersected with the offered direction (as described in
[JSEP] (section 5.3.1.)), return
true.
If all the preceding checks were performed and true
was not returned, nothing remains to be negotiated; return
false.
4.8 Interfaces for Interactive Connectivity Establishment
4.8.1 RTCIceCandidate Interface
This interface describes an ICE candidate, described in
[ICE] Section 2. Other than
candidate, sdpMid,
sdpMLineIndex, and usernameFragment,
the remaining attributes are derived from parsing the
candidate member in candidateInitDict,
if it is well formed.
To maintain backward compatibility, any error on parsing the
candidate attribute is ignored. In such case, the
candidate attribute holds the raw
candidate string given in candidateInitDict,
but derivative attributes such as foundation,
priority, etc are set to null.
Attributes
Most attributes below are defined in section 15.1 of [ICE].
candidate of type DOMString, readonly
This carries the candidate-attribute as defined
in section 15.1 of [ICE]. 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
If not 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
If not 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
A unique identifier that allows ICE to correlate candidates
that appear on multiple
RTCIceTransports.
The assigned network component of the candidate
("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
The assigned priority of the candidate.
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]].remote.
By default, the user agent MUST leave the address attribute
as null for any exposed remote candidate.
Once a RTCPeerConnection instance learns on an address
by the web application using addIceCandidate, the user agent
can expose the address attribute value in any RTCIceCandidate
of the RTCPeerConnection instance representing a remote
candidate with that newly learnt address.
Note
The addresses exposed in candidates gathered via ICE
and made visibile to the application in
RTCIceCandidate 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.
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 iceTransportPolicy member of
RTCConfiguration.
To limit the addresses exposed to the application
itself, browsers can offer their users different policies
regarding sharing local addresses, as defined in
[RTCWEB-IP-HANDLING].
relatedAddress of type DOMString, readonly, nullable
For a candidate that is derived from another, such as a relay
or reflexive candidate, the 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
For a candidate that is derived from another, such as a relay
or reflexive candidate, the 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
This carries the ufrag as defined in section
15.4 of [ICE].
Methods
toJSON()
To invoke the toJSON() operation of the RTCIceCandidate interface, run the following steps:
This carries the candidate-attribute as defined
in section 15.1 of [ICE]. If this represents an
end-of-candidates indication, candidate
is an empty string.
sdpMid of type DOMString, nullable, defaulting to
null
The primary grammar for candidate-attribute
is defined in section 15.1 of [ICE]. 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.
4.8.1.2 RTCIceProtocol Enum
The RTCIceProtocol represents the protocol of the ICE
candidate.
An "active" TCP candidate is one for which the
transport will attempt to open an outbound connection but
will not receive incoming connection requests.
passive
A "passive" TCP candidate is one for which the
transport will receive incoming connection attempts but not
attempt a connection.
so
An "so" candidate is one for which the
transport will attempt to open a connection simultaneously
with its peer.
Note
The user agent will typically only gather active
ICE TCP candidates.
4.8.1.4 RTCIceCandidateType Enum
The RTCIceCandidateType represents the type of the ICE
candidate, as defined in [ICE] section 15.1.
The icecandidate event is used for three
different types of indications:
A candidate has been gathered. The candidate
member of the event will be populated normally. It should be
signaled to the remote peer and passed into
addIceCandidate.
An RTCIceTransport has finished gathering a
generation of candidates, and is providing an end-of-candidates
indication as defined by Section 8.2 of [TRICKLE-ICE]. This is
indicated by candidate.candidate being set to an
empty string. The candidate object
should be signaled to the remote peer and passed into
addIceCandidate
like a typical ICE candidate, in order to provide the
end-of-candidates indication to the remote peer.
All RTCIceTransports have finished
gathering candidates, and the RTCPeerConnection's
RTCIceGatheringState has transitioned to
"complete".
This is indicated by the
candidate
member of the event being set to 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
icegatheringstatechange event with the
"complete"
state.
The candidate attribute is the
RTCIceCandidate object with the new ICE
candidate that caused the event.
This attribute is set to null when an event is
generated to indicate the end of candidate gathering.
Note
Even where there are multiple media components,
only one event containing a null candidate is
fired.
url of type DOMString, readonly, nullable
The url 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
null.
The address attribute is the local IP
address used to communicate with the STUN or TURN
server.
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 address
attribute will be set to null.
port of type unsigned short, readonly, nullable
The port attribute is the port used to
communicate with the STUN or TURN server.
If the address attribute is null,
the port attribute is also set to null.
url of type DOMString, readonly
The url attribute is the STUN or TURN URL that
identifies the STUN or TURN server for which the failure
occurred.
errorCode of type unsigned short, readonly
The errorCode attribute is the numeric STUN
error code returned by the STUN or TURN server
[STUN-PARAMETERS].
If no host candidate can reach the server,
errorCode 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
RTCIceGatheringState of "gathering".
errorText of type USVString, readonly
The errorText attribute is the STUN reason text
returned by the STUN or TURN server [STUN-PARAMETERS].
If the server could not be reached, errorText
will be set to an implementation-specific value providing
details about the error.
The explicit certificate management functions provided here are
optional. If an application does not provide the
certificates configuration option when constructing an
RTCPeerConnection 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.
The generateCertificate 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
RTCCertificate interface. The returned
RTCCertificate can be used to control the
certificate that is offered in the DTLS sessions established by
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"
}.
Note
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
RTCPeerConnection, 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.
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:
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
RTCPeerConnection, return a promise that is
rejected with a
DOMException of type NotSupportedError. In
particular, normalizedKeygenAlgorithmMUST 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.
An optional expires attribute MAY be added to the
definition of the algorithm that is passed to generateCertificate. If this
parameter is present it indicates the maximum time that the
RTCCertificate is valid for relative to the
current time.
4.9.2 RTCCertificate Interface
The RTCCertificate 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 RTCPeerConnection
uses to authenticate with a peer, and the origin ([[Origin]])
that created the object.
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
RTCPeerConnection using this certificate fail.
Note that this value might not be reflected in a
notAfter parameter in the certificate itself.
Methods
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
RTCCertificate 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.
Initialize value.expires attribute to
contain serialized.[[Expires]].
Set value.[[Certificate]] to a copy of
serialized.[[Certificate]].
Set value.[[Origin]] to a copy of
serialized.[[Origin]].
Set value.[[KeyingMaterialHandle]] to the
private keying material handle
resulting from deserializing serialized.[[KeyingMaterialHandle]].
Note
Supporting structured cloning in this manner
allows RTCCertificate instances to be persisted to stores. It
also allows instances to be passed to other origins using APIs
like postMessage() [html]. However, the object cannot
be used by any other origin than the one that originally created it.
5. RTP Media API
The RTP media API lets a web application send and receive
MediaStreamTracks over a peer-to-peer connection. Tracks, when
added to an RTCPeerConnection, result in signaling; when this
signaling is forwarded to a remote peer, it causes corresponding tracks to
be created on the remote side.
Note
There is not an exact 1:1 correspondence between tracks sent
by one RTCPeerConnection and received by the other. For one,
IDs of tracks sent have no mapping to the IDs of tracks received. Also,
replaceTrack changes the
track sent by an RTCRtpSender without creating a new
track on the receiver side; the corresponding
RTCRtpReceiver will only have a single track,
potentially representing multiple sources of media stitched together. Both
addTransceiver and
replaceTrack 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
RTCRtpSender on one side and an
RTCRtpReceiver's track on the other side, matching senders
and receivers using the RTCRtpTransceiver's mid if necessary.
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 [JSEP] (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.
Note
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
scaleResolutionDownBy 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.
The actual encoding and transmission of MediaStreamTracks
is managed through objects called RTCRtpSenders.
Similarly, the reception and decoding of MediaStreamTracks is
managed through objects called RTCRtpReceivers. Each
RTCRtpSender is associated with at most one track,
and each track to be received is associated with exactly
one RTCRtpReceiver.
The encoding and transmission of each MediaStreamTrackSHOULD be made such that its characteristics (width, height and frameRate
for video tracks; volume, 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 RTCRtpSender settings applied that
instruct the implementation to act differently.
An RTCPeerConnection object contains a set of
RTCRtpTransceivers, representing the paired
senders and receivers with some shared state. This set is initialized to
the empty set when the RTCPeerConnection object is
created.RTCRtpSenders and
RTCRtpReceivers are always created at the same time
as an RTCRtpTransceiver, which they will remain
attached to for their lifetime.
RTCRtpTransceivers are created implicitly when the
application attaches a MediaStreamTrack to an
RTCPeerConnection via the addTrack()
method, or explicitly when the application uses the
addTransceiver 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
MediaStreamTrack and RTCRtpReceiver are
surfaced to the application via the track event.
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.
5.1 RTCPeerConnection Interface Extensions
The RTP media API extends the RTCPeerConnection
interface as described below.
When the addTrack method is invoked,
the user agent MUST run the following steps:
Let connection be the
RTCPeerConnection object on which this
method was invoked.
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.
The steps below describe how to determine if an existing
sender can be reused. Doing so will cause future calls to
createOffer and createAnswer to
mark the corresponding media description as
sendrecv or sendonly and add the
MSID of the sender's streams, as defined in [JSEP] (section 5.2.2. and section 5.3.2.).
If any RTCRtpSender object in
senders matches all the following criteria, let
sender be that object, or null
otherwise:
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
RTCRtpSender 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.
When the other peer stops sending a track in this manner, the
track is removed from any remote MediaStreams
that were initially revealed in the track event, and
if the MediaStreamTrack is not already muted,
a mute event is
fired at the track.
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 an RTCSessionDescription
of type "rollback"), then abort these steps.
Validate sendEncodings by running the following steps:
Verify that each rid
value in sendEncodings conforms to the grammar specified in
Section 10 of [MMUSIC-RID]. If one of the RIDs does not meet
these requirements, throw a TypeError.
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 the number of RTCRtpEncodingParameters
stored in sendEncodings exceeds maxN,
then trim sendEncodings from the tail until its length
is maxN.
If the 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 RTCRtpEncodingParameters now
stored in sendEncodings is 1, then remove any
rid
member from the lone entry.
Note
Providing a single, default
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 createOffer will be configured to send
multiple RTP encodings as defined in [JSEP] (section 5.2.2. and section 5.2.1.). When
setRemoteDescription is called with a
corresponding remote description that is able to receive
multiple RTP encodings as defined in [JSEP] (section 3.7.), the
RTCRtpSender may send multiple RTP
encodings and the parameters retrieved via the transceiver's
sender.getParameters() will reflect the
encodings negotiated.
When the remote PeerConnection's track event fires
corresponding to the RTCRtpReceiver being
added, these are the streams that will be put in the event.
The RTCRtpTransceiver will neither send
nor receive RTP. It will generate a zero port in the offer. In
answers, its RTCRtpSender will not offer to
send RTP, and its RTCRtpReceiver will not
offer to receive RTP. This is a terminal state.
5.1.1 Processing Remote MediaStreamTracks
An application can reject incoming media descriptions by setting
the transceiver's direction to either "inactive" to turn
off both directions temporarily, or to "sendonly" 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
RTCRtpTransceiver.stop()
and subsequently initiate negotiation from its end.
To
process remote tracks given an RTCRtpTransceivertransceiver, direction, msids,
addList, removeList, and trackEventInits,
run the following steps:
To set the associated remote streams given
RTCRtpReceiverreceiver, msids,
addList, and removeList, run the following steps:
Let connection be the
RTCPeerConnection object associated with
receiver.
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 ids corresponding to
msids.
The RTCRtpSender interface allows an
application to control how a given MediaStreamTrack is
encoded and transmitted to a remote peer. When setParameters
is called on an RTCRtpSender object, the encoding is
changed appropriately.
To create an RTCRtpSender with a
MediaStreamTrack, track, a string,
kind, a list of
MediaStream objects, streams, and
optionally a list of RTCRtpEncodingParameters
objects, sendEncodings, run the following steps:
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
[JSEP] (section 5.2.1.).
Let sender have a [[SendEncodings]]
internal slot, representing a list of
RTCRtpEncodingParameters dictionaries.
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 RTCRtpEncodingParameters with
active set to true.
Let sender have a [[SendCodecs]]
internal slot, representing a list of
RTCRtpCodecParameters dictionaries, and
initialized to an empty list.
Let sender have a [[LastReturnedParameters]]
internal slot, which will be used to match
getParameters and
setParameters transactions.
The track attribute is the track that is
associated with this RTCRtpSender object. If
track is ended, or if the track's output is disabled,
i.e. the track is disabled and/or muted, the
RTCRtpSenderMUST send black frames (video) and
MUST NOT send (audio). In the case of video,
the RTCRtpSenderSHOULD send one
black frame per second. If track is null then
the RTCRtpSender does not send. On getting, the
attribute MUST return the value of the [[SenderTrack]]
slot.
The transport attribute is the transport over
which media from track is sent in the form of RTP
packets. Prior to construction of the
RTCDtlsTransport object, the
transport attribute will be null. When bundling is
used, multiple RTCRtpSender objects will
share one transport and will all send RTP and RTCP
over the same transport.
On getting, the attribute MUST return the value of the
[[SenderTransport]] slot.
Methods
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, getCapabilities returns 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.
setParameters
The setParameters method updates how
track is encoded and transmitted to a remote
peer.
When the setParameters method is called, the user
agent MUST run the following steps:
Any parameter in parameters is marked as a
Read-only parameter (such as RID) and has a
value that is different from the corresponding parameter
value in sender.[[LastReturnedParameters]].
Note that this also applies to transactionId.
setParameters 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 RTCRtpSendParameters dictionary
are designed to not enable this, so attributes like
cname that cannot be changed are read-only. Other
things, like bitrate, are controlled using limits such as
maxBitrate, 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.
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.
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 rejectp with a newly
createdInvalidModificationError, 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.
Changing dimensions and/or frame rates might not require
negotiation. Cases that may require negotiation include:
Changing a resolution to a value outside of the
negotiated imageattr bounds, as described in
[RFC6236].
Changing a frame rate to a value that causes the block
rate for the codec to be exceeded.
A video track differing in raw vs. pre-encoded
format.
An audio track having a different number of
channels.
Sources that also encode (typically hardware encoders)
might be unable to produce the negotiated codec; similarly,
software sources might not implement the codec that was
negotiated for an encoding source.
setStreams
Sets the MediaStreams to be associated with this
sender's track.
When the setStreams method is invoked,
the user agent MUST run the following steps:
Let sender be the
RTCRtpSender object on which this method
was invoked.
Let connection be the
RTCPeerConnection object on which this
method was invoked.
A sequence containing the media codecs that an
RTCRtpSender 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 codecs with a mimeType
attribute indicating retransmission via audio/rtx or
video/rtx, and an sdpFmtpLine attribute (providing
the "apt" and "rtx-time" parameters). Read-only parameter.
A unique identifier for the last set of parameters applied.
Ensures that setParameters can only be called based on a previous
getParameters, and that there are no intervening changes.
Read-only parameter.
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 maxBitrate 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.
Note
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.
scaleResolutionDownBy will be
1.0).
The RTCRtpHeaderExtensionParameters dictionary
enables an application to determine whether a header extension
is configured for use within an RTCRtpSender
or RTCRtpReceiver. For an
RTCRtpTransceivertransceiver, an
application can determine the "direction" parameter (defined in
Section 5 of [RFC5285]) of a header extension as follows
without having to parse SDP:
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 [JSEP] (section 5.8.). For an
RTCRtpSender, these parameters come from the
remote description, and for an
RTCRtpReceiver, they come from the local
description. Read-only parameter.
Supported media codecs as well as entries for RTX, RED and FEC
mechanisms. There will only be a single entry in
codecs for retransmission via RTX, with
sdpFmtpLine not present.
The RTCRtpCodecCapability dictionary provides
information about codec capabilities. Only capability
combinations that would utilize distinct payload types in a
generated SDP offer are provided. For example:
Two H.264/AVC codecs, one for each of two supported
packetization-mode values.
Two CN codecs with different clock rates.
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.
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
RTCRtpCodecParameters dictionaries, and
initialized to an empty list.
Let receiver have a
[[LastStableStateReceiveCodecs]] internal slot and initialize
it to an empty list.
The track
attribute is the track that is associated with this
RTCRtpReceiver object receiver.
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.
The transport attribute is the
transport over which media for the receiver's track
is received in the form of RTP packets. Prior to construction of
the RTCDtlsTransport object, the
transport attribute will be null. When bundling is
used, multiple RTCRtpReceiver objects will
share one transport and will all receive RTP and
RTCP over the same transport.
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,
getCapabilities returns 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.
Both the local and remote description may
affect this list of codecs. For example, if three codecs are
offered, the receiver will be prepared to receive each of
them and will return them all from
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.
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 RTCRtpReceiver's
MediaStreamTrack, the user agent MUST queue a task to
update the relevant information for the
RTCRtpContributingSource and
RTCRtpSynchronizationSource 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
RTCRtpContributingSource 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 RTCRtpReceiver's
MediaStreamTrack in the previous 10 seconds.
As stated in the conformance
section, requirements phrased as algorithms may be implemented in
any manner so long as the end result is equivalent. So, an
implementation does not need to literally queue a task for every frame,
as long as the end result is that within a single event loop task
execution, all returned RTCRtpSynchronizationSource
and RTCRtpContributingSource dictionaries for a
particular RTCRtpReceiver contain information from a
single point in the RTP stream.
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.
Only present for audio receivers. Whether the last RTP packet,
delivered from this source, contains voice activity (true) or not
(false). If the RFC 6464 extension header was not present, or if
the peer has signaled that it is not using the V bit by setting the
"vad" extension attribute to "off", as described in [RFC6464],
Section 4, voiceActivityFlag will be absent.
(Feature at Risk) Issue 1
voiceActivityFlag is marked as a feature at risk, since there is no
clear commitment from implementers.
A RTCRtpTransceiver may become associated with a new pending
description in JSEP while still being disassociated with the current
description. This may happen in check if negotiation is needed.
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
[JSEP] (section 5.2.1. and section 5.3.1.), and is only modified
there.
Let transceiver have a [[Mid]] internal slot,
initialized to null.
The mid
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.
The sender attribute exposes the
RTCRtpSender corresponding to the RTP media
that may be sent with mid = [[Mid]]. On getting,
the attribute MUST return the value of the [[Sender]]
slot.
The receiver attribute is the
RTCRtpReceiver corresponding to the RTP media
that may be received with mid = [[Mid]]. On
getting the attribute MUST return the value of the
[[Receiver]] slot.
As defined in [JSEP] (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.active since one cannot be
deduced from the other. If this transceiver has never been
represented in an offer/answer exchange, the value is
null. If the transceiver is
stopped, the value is "stopped".
On getting, the user agent MUST run the following steps:
Let transceiver be the
RTCRtpTransceiver object on which the
getter is invoked.
A stopping transceiver will cause future calls to
createOffer to generate a zero port in the
media description for the corresponding transceiver, as
defined in [JSEP] (section 4.2.1.) (The user agent
MUST treat a stopping transceiver as stopped for the
purposes of JSEP only in this case). However, to avoid problems
with [BUNDLE], a transceiver that is stopping, but not
stopped, will not affect createAnswer.
The transceiver will remain in the stopping state,
unless it becomes stopped by setRemoteDescription
processing a rejected m-line in a remote offer or answer.
Note
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 stop method is
invoked, the user agent MUST run the following steps:
Let transceiver be the
RTCRtpTransceiver object on which the
method is invoked.
Let connection be the
RTCPeerConnection object associated with
transceiver.
The setCodecPreferences method overrides the
default codec preferences used by the user agent. When
generating a session description using either
createOffer or createAnswer, the
user agentMUST use the indicated codecs, in the order
specified in the codecs argument, for the media
section corresponding to this 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
createOffer and createAnswer that
include this RTCRtpTransceiver until this method is
called again. Setting codecs to an empty sequence
resets codec preferences to any default value.
Note
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 apear in this list as
defined in section 5.2.1 of [JSEP].
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.
Due to a recommendation in [SDP], calls to
createAnswerSHOULD 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, [JSEP] (section 5.3.1.)
allows adding codecs that were not in the offer, so
implementations can behave differently.
Let transceiver be the RTCRtpTransceiver
object this method was invoked on.
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.
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.
The addTransceiver method establishes the simulcast envelope which
includes the maximum number of simulcast streams that can be sent, as well as the ordering of the
encodings. While characteristics of individual simulcast streams can be modified using
the setParameters method, the simulcast envelope cannot be changed. One of the
implications of this model is that the addTrack() method cannot provide simulcast
functionality since it does not take sendEncodings as an argument, and therefore cannot
configure an RTCRtpTransceiver to send simulcast.
While setParameters 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 active
member to false, or can be reactivated by setting the active
member to true. Using setParameters, stream characteristics can be
changed by modifying attributes such as maxBitrate.
Note
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 maxBitrate, 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.
As defined in [JSEP] (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 [MMUSIC-SIMULCAST]) are not supported.
This specification does not define how to configure reception of multiple
RTP encodings using createOffer, createAnswer
or addTransceiver. However when setRemoteDescription
is called with a corresponding remote description that is able to send multiple RTP
encodings as defined in [JSEP], and the browser supports receiving multiple RTP
encodings, the RTCRtpReceiver may receive multiple RTP encodings and the parameters
retrieved via the transceiver's receiver.getParameters()
will reflect the encodings negotiated.
Note
An RTCRtpReceiver 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.
5.4.1.1 Encoding Parameter Examples
This section is non-normative.
Examples of simulcast scenarios implemented with encoding parameters:
asyncfunctionplayMusicOnHold() {
try {
// Assume we have an audio transceiver and a music track named musicTrackawait 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);
}
}
asyncfunctionstopOnHoldMusic() {
// Assume we have an audio transceiver and a microphone track named micTrackawait 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:
asyncfunctiononOffHold() {
try {
// Apply the sendrecv offer first, to ensure receiver is ready for ICE candidates.await pc.setRemoteDescription(sendrecvOffer);
// Start sending audioawait audio.sender.replaceTrack(micTrack);
// Set the direction sendrecv (just in time for the answer)
audio.direction = 'sendrecv';
// Call createAnswer and send a sendrecv answerawait doAnswer();
} catch (err) {
// handle signaling error
}
}
5.5 RTCDtlsTransport Interface
The RTCDtlsTransport 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 RTCRtpSender and
RTCRtpReceiver 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
RTCDtlsTransport interface allows access to information
about the underlying transport and the security added.
RTCDtlsTransport objects are constructed as a result
of calls to setLocalDescription() and
setRemoteDescription(). Each
RTCDtlsTransport object represents the DTLS transport
layer for the RTP or RTCP component
of a specific RTCRtpTransceiver, or a group of
RTCRtpTransceivers if such a group has been
negotiated via [BUNDLE].
Note
A new DTLS association for an existing
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 RTCDtlsTransport has a
[[DtlsTransportState]] internal slot initialized to
"new" and a
[[RemoteCertificates]] slot initialized to an empty list.
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
RTCDtlsTransport object to receive the state update
and error notification.
If the state of transport is
already "failed", abort these steps.
When the underlying DTLS transport needs to update the state of the
corresponding RTCDtlsTransport object for any other
reason, the user agent
MUST queue a task that runs the following steps:
Let transport be the
RTCDtlsTransport object to receive the state update.
If newState is
connected
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.
The iceTransport attribute is the underlying
transport that is used to send and receive packets. The
underlying transport may not be shared between multiple active
RTCDtlsTransport objects.
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.
5.6 RTCIceTransport Interface
The RTCIceTransport 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 setLocalDescription() and
setRemoteDescription(). The underlying ICE state is managed
by the ICE agent; as such, the state of an
RTCIceTransport 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 component
of a specific RTCRtpTransceiver, or a group of
RTCRtpTransceivers if such a group has been
negotiated via [BUNDLE].
Note
An ICE restart for an existing
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 RTCIceTransport, the
user agent MUST queue a task that runs the following steps:
When the ICE Agent is finished gathering a generation of
candidates for an RTCIceTransport, and those
candidates have been surfaced to the application, the user agent MUST
queue a task that runs the following steps:
When the ICE Agent indicates that a new ICE candidate is
available for an RTCIceTransport, 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:
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:
The RTCIceTransportState of an RTCIceTransport 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 RTCIceTransportState are related and are handled in
the same task.
When the ICE Agent indicates that an RTCIceTransport has
changed either the selected candidate pair, the RTCIceTransportState or
both, the user agent MUST queue a task that runs the following steps:
The component
attribute MUST return the ICE component of the transport. When
RTCP mux is used, a single
RTCIceTransport transports both RTP and RTCP
and component is set to "rtp".
The RTCIceTransport was just created, and
has not started gathering candidates yet.
gathering
The RTCIceTransport is in the process of
gathering candidates.
complete
The RTCIceTransport 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.
The RTCIceTransport is gathering
candidates and/or waiting for remote candidates to be supplied,
and has not yet started checking.
checking
The RTCIceTransport 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 RTCIceTransport 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 "checking" (if there are candidate pairs remaining
to be checked) or "disconnected" (if there are no candidate pairs
to check, but the peer is still gathering and/or waiting for
additional remote candidates).
completed
The RTCIceTransport 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
"failed".
disconnected
The ICE Agent has determined that connectivity is
currently lost for this RTCIceTransport.
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:
Losing the network interface for the connection in
use.
Repeatedly failing to receive a response to STUN
requests.
Alternatively, the RTCIceTransport 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 RTCIceTransport 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
"failed" state does not cause DTLS transports, SCTP
associations or the data channels that run over them to close, or
tracks to mute.
closed
The RTCIceTransport has shut down and is
no longer responding to STUN requests.
Note
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 "checking" to "completed".
An ICE restart causes candidate gathering and connectity checks to
begin anew, causing a transition to "connected" if begun in the
"completed" state. If begun in the transient
"disconnected" state, it causes a transition to
"checking", effectively forgetting that connectivity was
previously lost.
The "failed" and "completed" states require an
indication that there are no additional remote candidates. This can be
indicated by calling addIceCandidate with
a candidate value whose candidate property is set
to an empty string or by canTrickleIceCandidates being set to
false.
The ICE Transport is used for RTP (or RTCP multiplexing),
as defined in [ICE], 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 [ICE],
Section 4.1.1.1. This represents the component-id
value 2 when encoded in
candidate-attribute.
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.
6.1 RTCPeerConnection Interface Extensions
The Peer-to-peer data API extends the
RTCPeerConnection interface as described below.
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 RTCSctpTransport
object stored in the [[SctpTransport]]
internal slot.
ondatachannel of type EventHandler
The event type of this event handler is
datachannel.
Methods
createDataChannel
Creates a new RTCDataChannel object with
the given label. The RTCDataChannelInit
dictionary can be used to configure properties of the underlying
channel such as data reliability.
When the createDataChannel
method is invoked, the user agent MUST run the following
steps.
Let connection be the
RTCPeerConnection object on which the
method is invoked.
This means the id member will be ignored if
the data channel is negotiated in-band; this is
intentional. Data channels negotiated in-band should have
IDs selected based on the DTLS role, as specified in
[RTCWEB-DATA-PROTOCOL].
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.
Let remoteMaxMessageSize be the value of the
max-message-size SDP attribute read from the remote description,
as described in [SCTP-SDP] (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.
6.1.1.3 Connected procedure
Once an SCTP transport
is connected, meaning the SCTP association of an RTCSctpTransport has been established, run the following
steps:
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
RTCDataChannel's 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 RTCDataChannel's
that can be used simultaneously. The attribute MUST, on
getting, return the value of the [[MaxChannels]] slot.
Note
This attribute's value will be
null until the SCTP transport goes into the
"connected" state.
onstatechange of type EventHandler
The event type of this event handler is
statechange.
The RTCSctpTransport is in the process of
negotiating an association. This is the initial state of the
[[SctpTransportState]] slot when an
RTCSctpTransport is created.
connected
When the negotiation of an association is completed, a task is
queued to update the [[SctpTransportState]] slot to
"connected".
closed
A task is queued to update the [[SctpTransportState]]
slot to "closed" when:
a SHUTDOWN or ABORT chunk
is received.
the SCTP association has been closed
intentionally, such as by closing the peer connection
or applying a remote description that rejects data or
changes the SCTP port.
the underlying DTLS association has transitioned
to "closed" state.
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.
There are two ways to establish a connection with
RTCDataChannel. The first way is to simply create an
RTCDataChannel at one of the peers with the
negotiatedRTCDataChannelInit dictionary member unset or set to
its default value false. This will announce the new channel in-band and
trigger an RTCDataChannelEvent with the corresponding
RTCDataChannel 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 negotiatedRTCDataChannelInit 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 RTCDataChannel with the
negotiatedRTCDataChannelInit dictionary member set to true and
the same id. This will
connect the two separately created RTCDataChannel
objects. The second way makes it possible to create channels with
asymmetric properties and to create channels in a declarative way by
specifying matching ids.
Each RTCDataChannel 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 RTCSctpTransport (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 [RTCWEB-DATA].
An RTCDataChannel 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 (
maxRetransmits ) or set
a time during which transmissions (including retransmissions) are allowed
( maxPacketLifeTime ).
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.
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.
6.2.2 Announcing a data channel as open
When the user agent is to announce an RTCDataChannel as
open, the user agent MUST queue a task to run the following
steps:
When an underlying data transport is to be announced (the other
peer created a channel with negotiated 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:
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 [RTCWEB-DATA-PROTOCOL].
An RTCDataChannel 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:
In some cases, the user agent may be
unable to create an RTCDataChannel
's underlying data transport.
For example, the data channel's id may be outside the range negotiated by the
[RTCWEB-DATA] implementations in the SCTP handshake. When the user
agent determines that an RTCDataChannel's
underlying data transport cannot be created, the user agent MUST
queue a task to run the following steps:
When an
RTCDataChannel 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:
Let channel be the RTCDataChannel
object for which the user agent has received a message.
Let connection be the
RTCPeerConnection object associated with
channel.
If channel.[[ReadyState]] is not
"open", abort these steps and discard rawData.
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 binaryType is "blob":
Let data be a new Blob object
containing rawData as its raw data source.
If type indicates that rawData is binary
and binaryType is "arraybuffer":
Let data be a new ArrayBuffer object
containing rawData as its raw data source.
The label
attribute represents a label that can be used to distinguish this
RTCDataChannel 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.
ordered of type boolean, readonly
The ordered attribute
returns true if the RTCDataChannel is
ordered, and false if out of order delivery is allowed. On
getting, the attribute MUST return the value of the [[Ordered]] slot.
maxPacketLifeTime of type unsigned short, readonly,
nullable
The maxPacketLifeTime
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.
maxRetransmits of type unsigned short, readonly,
nullable
The maxRetransmits
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.
The negotiated
attribute returns true if this RTCDataChannel
was negotiated by the application, or false otherwise. On getting,
the attribute MUST return the value of the [[Negotiated]] slot.
id of type unsigned short, readonly, nullable
The id attribute returns the ID for this
RTCDataChannel. 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
[RTCWEB-DATA-PROTOCOL]. 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.
The bufferedAmount
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
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 "open"; 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.
The event type of this event handler is RTCErrorEvent.
errorDetail contains "sctp-failure",
sctpCauseCode contains the SCTP
Cause Code value, and message
contains the SCTP Cause-Specific-Information,
possibly with additional text.
The binaryType
attribute MUST, on getting, return the value to which it was
last set. On setting, if the new value is either the string
"blob" or the string "arraybuffer",
then set the IDL attribute to this new value. Otherwise,
throw a SyntaxError. When an
RTCDataChannel object is
created, the binaryType attribute MUST be
initialized to the string "blob".
This attribute controls how binary data is exposed to scripts.
See Web Socket's binaryType.
Methods
close
Closes the RTCDataChannel. It may be
called regardless of whether the
RTCDataChannel object was created by this
peer or the remote peer.
When the close method is called, the user agent
MUST run the following steps:
Let channel be the
RTCDataChannel object which is about to
be closed.
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 RTCDataChannel
object on which data is to be sent.
Let data be the raw data represented by the
Blob object.
Note
Although the actual retrieval of data from a 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.
The actual transmission of data occurs in
parallel. If sending data leads to an SCTP-level error, the
application will be notified asynchronously through onerror.
Increase the value of the [[BufferedAmount]] slot by
the byte size of data.
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 RTCDataChannel object. If set
to true, it is up to the application to negotiate the channel and
create an RTCDataChannel object with the same
id at the other
peer.
Note
If set to true, the application must also take
care to not send a message until the other peer has created a
data channel to receive it. Receiving a message on an SCTP stream
with no associated data channel is undefined behavior, and it may
be silently dropped. This will not be possible as long as both
endpoints create their data channel before the first offer/answer
exchange is complete.
An RTCDataChannel object MUST not be garbage
collected if its
[[ReadyState]] slot is
"connecting" and at least one event listener is registered
for open events, message events,
error events, closing events, or close events.
[[ReadyState]] slot is
"open" and at least one event listener is registered for
message events, error events, closing events, or
close events.
[[ReadyState]] slot is
"closing" and at least one event listener is registered
for error events, or close events.
This section describes an interface on RTCRtpSender
to send DTMF (phone keypad) values across an
RTCPeerConnection. Details of how DTMF is sent to the
other peer are described in [RTCWEB-AUDIO].
7.1 RTCRtpSender Interface Extensions
The Peer-to-peer DTMF API extends the RTCRtpSender
interface as described below.
On getting, the dtmf attribute returns the value
of the [[Dtmf]] internal slot, which represents a
RTCDTMFSender which can be used to send DTMF, or
null if unset. The [[Dtmf]] internal slot
is set when the kind of an RTCRtpSender's
[[SenderTrack]] is "audio".
7.2 RTCDTMFSender
To create an RTCDTMFSender, the user agent MUST
run the following steps:
The event type of this event handler is
tonechange.
canInsertDTMF of type boolean, readonly
Whether the RTCDTMFSenderdtmfSender is capable of sending
DTMF. On getting, the user agent MUST return the result of running
determine if DTMF can be sent for dtmfSender.
toneBuffer of type DOMString, readonly
The toneBuffer
attribute MUST return a list of the tones remaining to be played
out. For the syntax, content, and interpretation of this list,
see 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:
Remove the first character from the [[ToneBuffer]]
slot and let that character be tone.
If tone is "," 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.
If tone is not "," 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.
Since insertDTMF 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.
7.3 canInsertDTMF algorithm
To determine if DTMF can be sent for an RTCDTMFSender
instance dtmfSender, the user agent MUST queue a task that runs the following steps:
Let sender be the RTCRtpSender associated with
dtmfSender.
The tone attribute contains the
character for the tone (including ",") that has just
begun playout (see insertDTMF ). 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.
The tone attribute contains the
character for the tone (including ",") that has just
begun playout (see insertDTMF ). 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.
8. Statistics Model
8.1 Introduction
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 RTCPeerConnection
object on which the stats request was issued. The calling Web application
provides the selector to the 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
RTCStatsid 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.
With a few exceptions, monitored objects, once created, exist for
the duration of their associated RTCPeerConnection.
This ensures statistics from them are available in the result from getStats()
even past the associated peer connection being closed.
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.
8.2 RTCPeerConnection Interface Extensions
The Statistics API extends the RTCPeerConnection
interface as described below.
The getStats() method
delivers a successful result in the form of an
RTCStatsReport object. An
RTCStatsReport object is a map between strings that
identify the inspected objects (id attribute in RTCStats
instances), and their corresponding RTCStats-derived
dictionaries.
An RTCStatsReport may be composed of several
RTCStats-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
RTCRtpSender uses multiple SSRCs to carry its track over the
network, the RTCStatsReport may contain one
RTCStats-derived dictionary per SSRC (which can be
distinguished by the value of the ssrc stats attribute).
Use these to retrieve the various dictionaries descended from
RTCStats 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.
8.4 RTCStats Dictionary
An RTCStats 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 timestamp and type. Specific
stats are added by extending the RTCStats
dictionary.
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
RTCStats-derived dictionary.
The timestamp, of type
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), timestamp represents
the time at which the information arrived at the local endpoint.
The remote timestamp can be found in an additional field in an
RTCStats-derived dictionary, if
applicable.
The type attribute MUST be initialized
to the name of the most specific type this
RTCStats dictionary represents.
id of type DOMString
A unique id that is associated with
the object that was inspected to produce this
RTCStats object. Two
RTCStats objects, extracted from two
different RTCStatsReport objects, MUST have
the same id if they were produced by inspecting the same
underlying object.
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.
Note
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 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
types when the corresponding objects exist on a RTCPeerConnection, with the
fields that are listed when they are valid for that object in addition to the generic fields defined in the RTCStats dictionary:
An implementation MAY support generating any other statistic defined
in [WEBRTC-STATS], and MAY generate statistics that are not
documented.
8.7 GetStats Example
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:
asyncfunctiongatherStats(pc) {
try {
const [sender] = pc.getSenders();
const baselineReport = await sender.getStats();
awaitnewPromise(resolve => setTimeout(resolve, aBit)); // wait a bitconst currentReport = await sender.getStats();
// compare the elements from the current report with the baselinefor (const now of currentReport.values()) {
if (now.type != 'outbound-rtp') continue;
// get the corresponding stats from the baseline reportconst 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);
}
}
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
[RTCWEB-RTP], [RTCWEB-AUDIO], and [RTCWEB-TRANSPORT].
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
RTCRtpSender and RTCRtpReceiver can be used by
the application to get more fine grained control over the transmission
and reception of MediaStreamTracks.
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
RTCPeerConnection object (as described previously in
this document) will take as input the data received from a remote peer.
Similarly, a 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 RTCPeerConnection object.
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.
Note
In this document, we only specify aspects of the
following objects that are relevant when used along with an
RTCPeerConnection. Please refer to the original
definitions of the objects in the [GETUSERMEDIA] document for general
information on using MediaStream and
MediaStreamTrack.
9.2 MediaStream
9.2.1 id
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 RTCPeerConnection API.
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.
Note
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 RTCPeerConnection 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).
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 RTCRtpReceiverreceiver has
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.
9.3.1 MediaTrackSupportedConstraints, MediaTrackCapabilities,
MediaTrackConstraints and MediaTrackSettings
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
MediaStreamTracks 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
MediaStreamTracks sourced from a remote
source:
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 MediaStreamTracks sourced from a remote
source.
10. Examples and Call Flows
This section is non-normative.
10.1 Simple Peer-to-peer Example
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/parseconst 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 initiatefunctionstart() {
addCameraMic();
}
// add camera and microphone to connectionasyncfunctionaddCameraMic() {
try {
// get a local stream, show it in a self-view and add it to be sentconst 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 answerif (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});
}
} elseif (candidate) {
await pc.addIceCandidate(candidate);
}
} catch (err) {
console.error(err);
}
};
10.2 Advanced Peer-to-peer Example with Warm-up
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/parseconst 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.asyncfunctionwarmup(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;
} elseif (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 endsfunctionstart() {
signaling.send({start: true});
signaling.onmessage({data: {start: true}});
}
// add camera and microphone to already warmed-up connectionasyncfunctionaddCameraMicWarmedUp() {
const stream = selfView.srcObject;
if (audio && video && stream) {
awaitPromise.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();
} elseif (description) {
await pc.setRemoteDescription(description);
// if we got an offer, we need to reply with an answerif (description.type == 'offer') {
await pc.setLocalDescription();
signaling.send({description: pc.localDescription});
}
} else {
await pc.addIceCandidate(candidate);
}
} catch (err) {
console.error(err);
}
};
10.3 Simulcast Example
A client wants to send multiple RTP encodings (simulcast) to a
server.
const signaling = new SignalingChannel(); // handles JSON.stringify/parseconst constraints = {audio: true, video: true};
const configuration = {'iceServers': [{'urls': 'stun:stun.example.org'}]};
let pc;
// call start() to initiateasyncfunctionstart() {
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 sentconst 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 answerif (description.type == 'offer') {
await pc.setLocalDescription();
signaling.send({description: pc.localDescription});
}
} elseif (candidate) {
await pc.addIceCandidate(candidate);
}
} catch (err) {
console.error(err);
}
};
10.4 Peer-to-peer Data Example
This example shows how to create an
RTCDataChannel 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 input field for user input.
const signaling = new SignalingChannel(); // handles JSON.stringify/parseconst configuration = {iceServers: [{urls: 'stun:stun.example.org'}]};
let pc, channel;
// call start() to initiatefunctionstart() {
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 enterif (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 answerif (description.type == 'offer') {
await pc.setLocalDescription();
signaling.send({description: pc.localDescription});
}
} elseif (candidate) {
await pc.addIceCandidate(candidate);
}
} catch (err) {
console.error(err);
}
};
10.5 Call Flow Browser to Browser
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.
asyncfunctionsendDTMF() {
if (sender.dtmf.canInsertDTMF) {
sender.dtmf.insertDTMF('123');
awaitnewPromise(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 =>newPromise(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 startsawait 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 begunif (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');
}
10.7 Perfect Negotiation Example
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 "stable" state). The rest of the
application may use any and all modification methods and attributes,
without worrying about signaling state races.
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/parseconst 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 connectionasyncfunctionstart() {
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 errorslet 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});
}
} elseif (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 setLocalDescription (without arguments) and
setRemoteDescription (with implicit rollback) to avoid
races with other signaling messages being serviced.
The ignoreOffer variable is needed, because
the RTCPeerConnection object on the impolite side is never
told about ignored offers. We must therefore suppress errors from
incoming candidates belonging to such offers.
11. Error Handling
Some operations throw or fire RTCError. This is an extension
of DOMException
that carries additional WebRTC-specific information.
The errorDetail, sdpLineNumber, sctpCauseCode, receivedAlert and sentAlert members of RTCErrorInit have the same definitions as the attributes of the same name of RTCError.
The DTLS negotiation has failed or the connection
has been terminated with a fatal error. The
message contains information relating to
the nature of error. If a fatal DTLS alert was received,
the receivedAlert attribute is set to the
value of the DTLS alert received. If a fatal DTLS alert was
sent, the sentAlert attribute is set to
the value of the DTLS alert sent.
fingerprint-failure
The RTCDtlsTransport'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 "dtls-failure".
sctp-failure
The SCTP negotiation has failed or the connection
has been terminated with a fatal error. The
sctpCauseCode attribute is set to the
SCTP cause code.
sdp-syntax-error
The SDP syntax is not valid. The sdpLineNumber
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.
11.3 RTCErrorEvent Interface
The RTCErrorEvent interface is defined for cases when an
RTCError is raised as an event:
The RTCDTMFSender object has either just
begun playout of a tone (returned as the tone attribute) or just ended
the playout of tones in the
toneBuffer
(returned as an empty value in the
tone
attribute).
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
[RTCWEB-SECURITY-ARCH].
13.1 Impact on same origin policy
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.
13.2 Revealing IP addresses
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 RTCIceTransportPolicy 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.
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 [RTCWEB-IP-HANDLING] for
details).
13.3 Impact on local network
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:
A user agent will always request permission from the correspondent
user agent to communicate using ICE. This ensures that the user agent
can only send to partners who you have shared credentials with.
A user agent will always request ongoing permission to continue
sending using ICE continued consent. This enables a receiver to
withdraw consent to receive.
A user agent will always encrypt data, with strong per-session
keying (DTLS-SRTP).
A user agent will always use congestion control. This ensures that
WebRTC cannot be used to flood the network.
These measures are specified in the relevant IETF documents.
13.4 Confidentiality of Communications
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() 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
RTCCertificate objects, to reduce memory attack surface.
13.5 Persistent information exposed by WebRTC
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.getCapabilities
and RTCRtpReceiver.getCapabilities 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.
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.
13.6 Setting SDP from remote endpoints
setRemoteDescription 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 ontrack if the other endpoint does not
negotiate sending. Applications need to be on guard against malevolent
SDP.
14. Accessibility Considerations
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.
A. Acknowledgements
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.
ITU-T Recommendation X.509 version 3 (1997). "Information Technology - Open Systems Interconnection - The Directory Authentication Framework" ISO/IEC 9594-8:1997.. ITU.
