Object RTC (ORTC) API for WebRTC
This document defines a set of ECMAScript APIs in WebIDL to allow media to be sent and received
from another browser or device implementing the appropriate set of real-time protocols. However,
unlike the WebRTC 1.0 API, Object Realtime Communications (ORTC) API does not mandate a media signaling protocol or
format. As a result, ORTC does not utilize Session Description Protocol (SDP), nor does it mandate support for
the Offer/Answer state machine.
Instead, ORTC focuses on "sender", "receiver" and "transport" objects, which have
"capabilities" describing what they are capable of doing,
as well as "parameters" which define what they are configured to do.
"Tracks" and "data channels" are sent over the transports, between senders and receivers.
Overview
Object RealTime Communications (ORTC) provides a powerful API for the development of WebRTC based applications.
ORTC does not mandate a media signaling protocol or format (as the current WebRTC 1.0 does by mandating SDP Offer/Answer).
Instead, ORTC focuses on "sender", "receiver" and "transport" objects, which have
"capabilities" describing what they are capable of doing,
as well as "parameters" which define what they are configured to do.
"Tracks" and "data channels" are sent over the transports, between senders and receivers.
This specification defines several objects: RTCDtlsTransport (
Section 2
),
RTCIceTransport (
Section 3
),
RTCIceTransportController (
Section 4
),
RTCIceListener (
Section 5
), RTCRtpSender (
Section 6
),
RTCRtpReceiver (
Section 7
),
RTCRtpListener (
Section 8
),
RTCDtmfSender (
Section 10
), RTCDataChannel
Section 11
), and RTCSctpTransport (
Section 12
).
RTP dictionaries are described in
Section 9
the Statistics API is described in
Section 13
the Identity API is described in
Section 14
an event summary is provided in
Section 15
and WebRTC 1.0 compatibility issues are discussed in
Section 16
In a Javascript application utilizing the ORTC API,
the relationship between the application and the objects, as well
as between the objects themselves is shown below.
Horizontal or slanted arrows denote the flow of media or data,
whereas vertical arrows denote interactions via methods and events.
Terminology
The
EventHandler
interface represents a callback used for event handlers as defined in
[[!HTML5]].
The concepts
queue a
task
and
fires a
simple event
are defined in [[!HTML5]].
The terms
event
event
handlers
and
event
handler event types
are defined in [[!HTML5]].
The terms
MediaStream
and
MediaStreamTrack
are defined in
[[!GETUSERMEDIA]].
For Scalable Video Coding (SVC), the terms single-session transmission (
SST
) and multi-session transmission (
MST
are defined in [[RFC6190]]. This specification only supports
SST
but not
MST
However, within a single RTP session, this specification allows a codec implementation to indicate
whether it supports SVC, and if so, whether it is capable of configuring SVC layers to utilize distinct SSRCs, or whether all layers
must utilize the same SSRC.
The RTCDtlsTransport Object
The
RTCDtlsTransport
object includes information relating to Datagram Transport Layer Security (DTLS) transport.
Overview
An
RTCDtlsTransport
instance is associated to an
RTCRtpSender
an
RTCRtpReceiver
, or an
RTCSctpTransport
Operation
RTCDtlsTransport
instance is optionally constructed with an
RTCIceTransport
object.
Interface Definition
readonly attribute RTCIceTransport transport
The associated
RTCIceTransport
instance.
readonly attribute RTCDtlsTransportState state
The current state of the DTLS transport.
RTCDtlsParameters getLocalParameters()
Obtain the DTLS parameters of the local
RTCDtlsTransport
RTCDtlsParameters? getRemoteParameters()
Obtain the current DTLS parameters of the remote
RTCDtlsTransport
sequence getRemoteCertificates()
Obtain the certificates used by the remote peer.
void start(RTCDtlsParameters remoteParameters)
Start DTLS transport negotiation with the parameters of the remote DTLS transport.
If
remoteParameters
is invalid, throw an
InvalidParameter
exception.
void stop()
Stops and closes the DTLS transport object.
stop()
. is final - calling
start()
afterwards
will throw an
InvalidState
exception.<
attribute EventHandler? ondtlsstatechange
This event handler, of event handler type
dtlsstatechange
uses the
RTCDtlsStateChangedEvent
interface.
It
MUST
be supported by
all objects implementing the
RTCDtlsTransport
interface.
It is called any time the
RTCDtlsTransportState
changes.
attribute EventHandler? onerror
This event handler, of event handler type
error
MUST
be supported by all objects implementing the
RTCDtlsTransport
interface.
This event
MUST
be fired on reception of a DTLS alert.
The RTCDtlsParameters Object
The
RTCDtlsParameters
object includes information relating to DTLS configuration.
RTCDtlsRole role="auto"
The DTLS role, with a default of auto.
sequence fingerprints
Sequence of fingerprints.
The RTCDtlsFingerprint Object
The
RTCDtlsFingerprint
object includes the hash function algorithm and certificate fingerprint as described in [[!RFC4572]].
DOMString algorithm
One of the the hash function algorithms defined in the 'Hash function Textual Names' registry, initially specified in [[!RFC4572]] Section 8.
DOMString value
The value of the certificate fingerprint in lowercase hex string as expressed utilizing the syntax of 'fingerprint' in [[!RFC4572]] Section 5.
enum RTCDtlsRole
RTCDtlsRole
indicates the role of the DTLS agent.
auto
The DTLS role is be determined based on the resolved ICE role: the
'Controlled' role acts as the DTLS client,
the 'Controlling' role acts as the DTLS server.
client
The DTLS client role.
server
The DTLS server role.
enum RTCDtlsTransportState
RTCDtlsTransportState
indicates the state of the DTLS transport.
new
DTLS connection object has been created and has not started negotiating yet.
connecting
DTLS is in the process of negotiating a secure connection.
connected
DTLS has completed negotiation of a secure connection (including DTLS/SRTP).
closed
The DTLS connection has been closed intentionally or as the result of an error.
RTCDtlsTransportStateChangedEvent
The
dtlsstatechange
event of the
RTCDtlsTransport
object uses
the
RTCDtlsTransportStateChangedEvent
interface.
Firing an
RTCDtlsTransportStateChangedEvent
event named
with an
RTCDtlsTransportState
state
means that an event with the name
which does not bubble (except where otherwise stated) and is not
cancelable (except where otherwise stated), and which uses the
RTCDtlsTransportStateChangedEvent
interface with the
state
attribute set to the new
RTCDtlsTransportState
MUST
be
created and dispatched at the given target.
readonly attribute RTCDtlsTransportState state
The
state
attribute is the new
RTCDtlsTransportState
that caused the event.
RTCDtlsTransportState? state
The
state
attribute is the new
RTCDtlsTransportState
that caused the event.
The RTCIceTransport Object
The
RTCIceTransport
includes information relating to Interactive Connectivity Establishment (ICE).
Overview
An
RTCIceTransport
instance is associated to a transport object (such as
RTCDtlsTransport
),
and provides RTC related methods to it. To do an ICE restart, construct a new
RTCIceTransport
object
(or alternatively, a new
RTCIceListener
object).
Operation
An
RTCIceTransport
instance is constructed from
either an
RTCIceListener
or an
RTCIceOptions
object.
If an
RTCIceListener
was not passed into the constructor,
an
RTCIceListener
object is automatically created.
Interface Definition
readonly attribute RTCIceListener iceListener
The
RTCIceListener
specified in the
RTCIceTransport
constructor.
If none was specified, the system will create an
RTCIceListener
automatically using the
RTCIceOptions
specified.
readonly attribute RTCIceRole role
The current role of the ICE transport.
readonly attribute RTCIceComponent component
The component-id of the
RTCIceTransport
readonly attribute RTCIceTransportState state
The current state of the ICE transport.
readonly attribute RTCIceGatheringState iceGatheringState
Represents the current state of ICE candidate gathering.
sequence getLocalCandidates()
Retrieve the sequence of valid candidates associated with the local
RTCIceTransport
This retrieves all candidates currently known, even if an
onlocalcandidate
event hasn't been processed yet.
sequence getRemoteCandidates()
Retrieve the sequence of candidates associated with the remote
RTCIceTransport
. Only returns the candidates previously
added using
setRemoteCandidates
or
addRemoteCandidate
void gather(RTCIceGatherPolicy gatherPolicy)
Start gathering
RTCIceCandidate
objects, based on the gather policy
(set on the local system, not negotiated). If
gatherPolicy
is invalid, throw an
InvalidParameter
exception.
void start(RTCIceParameters remoteParameters, optional RTCIceRole role)
Starts candidate connectivity checks and attempts to connect to the remote
RTCIceTransport
. If
start
is called with invalid parameters,
throw an
InvalidParameters
exception.
void stop()
Stops and closes the current object.
RTCIceParameters getLocalParameters()
Obtain the ICE parameters of the local
RTCIceTransport
RTCIceParameters? getRemoteParameters()
Obtain the current ICE parameters of the remote
RTCIceTransport
RTCIceTransport createAssociatedTransport ()
Create an associated
RTCIceTransport
for RTCP,
and implicitly create a newly associated
RTCIceListener
object as well for RTCP, or reusing an existing
RTCP
RTCIceListener
if one is already associated with the
existing RTP
RTCIceListener
If called more than once for the same
component, throw an
InvalidStateError
exception. If called when
component
is "RTCP",
throw a
SyntaxError
exception.
void addRemoteCandidate(RTCIceCandidate remoteCandidate)
Add remote candidate associated with remote
RTCIceTransport
void setRemoteCandidates(sequence remoteCandidates)
Set the sequence of candidates associated with the remote
RTCIceTransport
attribute EventHandler? onlocalcandidate
This event handler, of event handler event type
icecandidate
MUST
be supported by all objects implementing the
RTCIceTransport
interface.
It receives events when a new local ICE candidate is available.
attribute EventHandler? onicestatechange
This event handler, of event handler type
icestatechange
uses the
RTCIceTransportStateChangedEvent
interface.
It
MUST
be supported by
all objects implementing the
RTCIceTransport
interface.
It is called any time the
RTCIceTransportState
changes.
attribute EventHandler? onicegatheringstatechange
This event handler, of event handler type
icegatheringstatechange
uses the
RTCIceGatheringStateChangedEvent
interface.
It
MUST
be supported by
all objects implementing the
RTCIceTransport
interface.
It is called any time the
RTCIceGatheringState
changes.
enum RTCIceComponent
RTCIceComponent
contains the component-id of the
RTCIceTransport
, which will be "RTP" unless RTP and RTCP are not multiplexed and
the
RTCIceTransport
object was returned by
createAssociatedTransport()
RTP
The RTP component ID, defined (as '1') in [[!RFC5245]] Section 4.1.1.1.
RTCP
The RTCP component ID, defined (as '2') in [[!RFC5245]] Section 4.1.1.1.
enum RTCIceGatheringState
The
RTCIceGatheringState
represents the state of ICE gathering.
new
No networking has occurred yet.
gathering
The ICE transport is in the process of gathering candidates.
complete
The ICE transport has completed gathering and is currently idle.
Events such as adding a new interface or a new TURN server will cause the state to go back to gathering.
RTCIceGatheringStateChangedEvent
The
icegatheringstatechange
event of the
RTCIceTransport
object uses
the
RTCIceGatheringStateChangedEvent
interface.
Firing an
RTCIceGatheringStateChangedEvent
event named
with an
RTCIceGatheringState
state
means that an event with the name
which does not bubble (except where otherwise stated) and is not
cancelable (except where otherwise stated), and which uses the
RTCIceGatheringStateChangedEvent
interface with the
state
attribute set to the new
RTCIceGatheringState
MUST
be
created and dispatched at the given target.
readonly attribute RTCIceGatheringState state
The
state
attribute is the new
RTCIceGatheringState
that caused the event.
RTCIceGatheringState? state
The
state
attribute is the new
RTCIceGatheringState
that caused the event.
The RTCIceParameters Object
The
RTCIceParameters
object includes the ICE username and password.
DOMString usernameFragment
ICE username.
DOMString password
ICE password.
enum RTCIceRole
RTCIceRole
contains the current role of the ICE transport.
controlling
controlling state
controlled
controlled state
enum RTCIceGatherPolicy
RTCIceGatherPolicy
denotes the policy relating to the gathering of ICE candidates.
all
The ICE transport gathers all types of candidates when this value is specified.
nohost
The ICE transport gathers all ICE candidate types except for host candidates.
relay
The ICE transport
MUST
only gather media relay candidates such as candidates passing through a TURN server.
This can be used to reduce leakage of IP addresses in certain use cases.
enum RTCIceTransportState
RTCIceTransportState
represents the current state of the ICE transport.
new
The ICE Transport is gathering addresses and/or waiting for remote candidates to be supplied.
checking
The ICE Transport has received at least one remote candidate, and is checking candidate pairs but has not yet found a connection. In addition to checking, it may also still be gathering.
connected
The ICE Transport 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.
completed
The ICE Transport has finished gathering and checking and found a connection.
There is no such state for the ICE non-controlling peer.
disconnected
Liveness checks have failed. This may trigger intermittently (and resolve itself without action).
closed
The ICE Transport has shut down and is no longer responding to STUN requests.
The non-normative ICE state transitions are:
RTCIceTransportStateChangedEvent
The
icestatechange
event of the
RTCIceTransport
object uses
the
RTCIceTransportStateChangedEvent
interface.
Firing an
RTCIceTransportStateChangedEvent
event named
with an
RTCIceTransportState
state
means that an event with the name
which does not bubble (except where otherwise stated) and is not
cancelable (except where otherwise stated), and which uses the
RTCIceTransportStateChangedEvent
interface with the
state
attribute set to the new
RTCIceTransportState
MUST
be
created and dispatched at the given target.
readonly attribute RTCIceTransportState state
The
state
attribute is the new
RTCIceTransportState
that caused the event.
RTCIceTransportState? state
The
state
attribute is the new
RTCIceTransportState
that caused the event.
The RTCIceOptions Object
The
RTCIceOptions
object includes information relating to ICE configuration.
sequence iceServers
An array containing STUN and TURN servers available to be used by ICE.
The RTCIceServer Object
The
RTCIceServer
is used to provide STUN or TURN server configuration.
In network topologies with multiple layers of NATs, it is desirable to have a STUN server
between every layer of NATs in addition to the TURN servers to minimize the peer to peer network latency.
An example of an array of
RTCIceServer
objects:
[ { urls: "stun:stun1.example.net } , { urls:"turn:turn.example.org", username: "user", credential:"myPassword"} ]
(DOMString or sequence) urls
STUN or TURN URI(s) as defined in [[!RFC7064]] and [[!RFC7065]]
DOMString username
If this
RTCIceServer
object represents a TURN server, then this attribute specifies
the username to use with that TURN server.
DOMString credential
If the uri element is a TURN URI, then this is the credential to use with that TURN server.
The RTCIceCandidate Object
The
RTCIceCandidate
object includes information relating to an ICE candidate.
foundation: "abcd1234",
priority: 1694498815,
ip: "192.0.2.33",
protocol: "udp",
port: 10000,
type: "host"
};
DOMString foundation
A unique identifier that allows ICE to correlate candidates that appear on multiple
RTCIceTransport
s.
unsigned long priority
The assigned priority of the candidate. This is automatically populated by the browser.
DOMString ip
The IP address of the candidate.
RTCIceProtocol protocol
The protocol of the candidate (UDP/TCP).
unsigned short port
The port for the candidate.
RTCIceCandidateType type
The type of candidate.
RTCIceTcpCandidateType tcpType
The type of TCP candidate.
DOMString relatedAddress=""
For candidates that are derived from others, such as relay or reflexive candidates, the
relatedAddress
refers to the host candidate that these are derived from. For host candidates, the
relatedAddress
is set to the empty string.
unsigned short relatedPort
For candidates that are derived from others, such as relay or reflexive candidates, the
relatedPort
refers to the host candidate that these are derived from. For host candidates, the
relatedPort
is null.
The RTCIceProtocol
The
RTCIceProtocol
includes the protocol of the ICE candidate.
udp
A UDP candidate, as described in [[!RFC5245]].
tcp
A TCP candidate, as described in [[RFC6544]].
The RTCIceTcpCandidateType
The
RTCIceTcpCandidateType
includes the type of the ICE TCP candidate, as described in [[RFC6544]].
active
An active TCP candidate is one for which the agent will
attempt to open an outbound connection but will not receive incoming
connection requests.
passive
A passive TCP candidate is one for which the agent
will receive incoming connection attempts but not attempt a
connection.
so
An so candidate is one for which the agent will attempt
to open a connection simultaneously with its peer.
The RTCIceCandidateType
The
RTCIceCandidateType
includes the type of the ICE candidate.
host
A host candidate.
srflx
A server reflexive candidate.
prflx
A peer reflexive candidate.
relay
A relay candidate.
RTCIceTransportEvent
The
icecandidate
event of the RTCIceTransport uses
the
RTCIceTransportEvent
interface.
Firing an
RTCIceTransportEvent
event named
with an
RTCIceCandidate
candidate
means that an event with the name
which does not bubble (except where otherwise stated) and is not
cancelable (except where otherwise stated), and which uses the
RTCIceTransportEvent
interface with the
candidate
attribute set to the new ICE candidate,
MUST
be
created and dispatched at the given target.
readonly attribute RTCIceCandidate candidate
The
candidate
attribute is the
RTCIceCandidate
object with the new ICE
candidate that caused the event.
If
candidate
is null,
there are no additional candidates for now.
RTCIceCandidate? candidate
The ICE candidate that caused the event.
Examples
// Assume we already have a way to signal. This is an example
// of how to offer ICE and DTLS parameters and ICE candidates and
// get back ICE and DTLS parameters and ICE candidates, and start
// both ICE and DTLS, assuming that RTP and RTCP are multiplexed.

function initiate(mySignaller) {
var iceOptions = ...;
var ice = new RTCIceTransport(iceOptions);
var dtls = new RTCDtlsTransport(ice);
// ... get tracks and RTP objects from other example

mySignaller.mySendInitiate({
"ice": ice.getLocalParameters(),
"dtls": dtls.getLocalParameters(),
// ... include RTP info from other example
}, function(remote) {
ice.start(remote.ice, RTCIceRole.controlling);
dtls.start(remote.dtls);
// ... start RTP senders and receivers from other example
});

ice.onlocalcandidate = function(candidate) {
mySignaller.mySendLocalCandidate(candidate);

mySignaller.onRemoteCandidate = function(candidate) {
ice.addRemoteCandidate(candidate);
// Assume we already have a way to signal and remote info is
// signalled to us. This is an example of how to answer with ICE and DTLS
// and DTLS parameters and ICE candidates and start both ICE and DTLS,
// assuming that RTP and RTCP are multiplexed.
//
function accept(mySignaller, remote) {
var iceOptions = ...;
var ice = new RTCIceTransport(iceOptions);
var dtls = new RTCDtlsTransport(ice);
// ... get tracks and RTP objects from other example
ice.onlocalcandidate = function(candidate) {
mySignaller.mySendLocalCandidate(candidate);

mySignaller.onRemoteCandidate = function(candidate) {
ice.addRemoteCandidate(candidate);

mySignaller.mySendAccept({
"ice": ice.getLocalParameters(),
"dtls": dtls.getLocalParameters()
// ... include RTP info from other example
});

ice.start(remote.ice, RTCIceRole.controlled);
dtls.start(remote.dtls);

// ... start RTP senders and receivers from other example
The RTCIceTransportController Object
The
RTCIceTransportController
object assists in the managing of ICE freezing and bandwidth estimation.
Overview
An
RTCIceTransportController
object provides methods to add and retrieve
RTCIceTransport
objects with a
component
of "RTP".
Operation
An
RTCIceTransportController
instance is automatically constructed.
Interface Definition
sequence getTransports()
Returns the
RTCIceTransport
objects with a
component
of "RTP".
void addTransport(RTCIceTransport transport, optional unsigned long index)
Adds
transport
to the
RTCIceTransportController
object for the purposes of managing
ICE freezing and sharing bandwidth estimation.
RTCIceTransport
objects will be unfrozen
according to their
index
transport
is inserted at
index
or at the end if
index
is not specified.
If
index
is greater than the current number of
RTCIceTransport
with a
component
of "RTP",
throw an
InvalidParameter
exception.
If
transport
has already been added to another
RTCIceTransportController
object, or if
the
component
is "RTCP", throw an
InvalidStateError
exception.
TODO: Specify the behavior when ICE is in progress and addTransport is called.
Examples
// This is an example of how to utilize distinct ICE transports for Audio and Video
// As well as for RTP and RTCP. If both sides can multiplex audio/video
// and/or RTP/RTCP then the multiplexing will occur.
//
// Assume we have an audioTrack and a videoTrack to send.
//
//create the RTP and RTCP ICE transports for audio and video
var audioRtpIceTransport = new RTCIceTransport(...);
var audioRtcpIceTransport = audioRtpIceTransport.createAssociatedTransport();
var videoRtpIceTransport = new RTCIceTransport(...);
var videoRtcpIceTransport = audioRtpIceTransport.createAssociatedTransport();
//
//create the DTLS transports
var audioRtpDtlsTransport = new RTCDtlsTransport(audioRtpIceTransport);
var audioRtcpDtlsTransport = new RTCDtlsTransport(audioRtcpIceTransport);
var videoRtpDtlsTransport = new RTCDtlsTransport(videoRtpIceTransport);
var videoRtcpDtlsTransport = new RTCDtlsTransport(videoRtcpIceTransport);
//
// Create the sender and receiver objects
var audioSender = new RtpSender(audioTrack, audioRtpDtlsTransport, audioRtcpDtlsTransport);
var videoSender = new RtpSender(videoTrack, videoRtpDtlsTransport, videoRtcpDtlsTransport);
var audioReceiver = new RtpReceiver(audioRtpDtlsTransport, audioRtcpDtlsTransport);
var videoReceiver = new RtpReceiver(videoRtpDtlsTransport, videoRtcpDtlsTransport);
//
// Retrieve the receiver and sender capabilities
var recvAudioCaps = RTCRtpReceiver.getCapabilities("audio");
var recvVideoCaps = RTCRtpReceiver.getCapabiltiies("video");
var sendAudioCaps = RTCRtpSender.getCapabilities("audio");
var recvVideoCaps = RTCRtpSender.getCapabilities("video");
//
// At this point, ICE/DTLS parameters and Send/Receive capabilities can be exchanged.
mySignaller.myOfferTracks({
// Indicate that the initiator would prefer to multiplex both A/V and RTP/RTCP
"bundle": true,
// Indicate that the initiator is willing to multiplex RTP/RTCP without A/V mux
"rtcpMux": true,
// Offer the ICE parameters
"audioRtpIce": audioRtpIceTransport.getLocalParameters(),
"audioRtcpIce": audioRtcpIceTransport.getLocalParameters(),
"videoRtpIce": videoRtpIceTransport.getLocalParameters(),
"videoRtcpIce": videoRtcpIceTransport.getLocalParameters(),
// Offer the DTLS parameters
"audioRtpDtls": audioRtpDtlsTransport.getLocalParameters(),
"audioRtcpDtls": audioRtcpDtlsTransport.getLocalParameters(),
"videoRtpDtls": videoRtpDtlsTransport.getLocalParameters(),
"videoRtcpDtls": videoRtcpDtlsTransport.getLocalParameters(),
// Offer the receiver and sender audio and video capabilities.
"recvAudioCaps": recvAudioCaps,
"recvVideoCaps": recvVideoCaps,
"sendAudioCaps": sendAudioCaps,
"sendVideoCaps": sendVideoCaps
}, function(answer) {
// The responder answers with its preferences, parameters and capabilities
//
// Derive the send and receive parameters, assuming that RTP/RTCP mux will be enabled.
var audioSendParams = myCapsToSendParams(sendAudioCaps, answer.recvAudioCaps);
var videoSendParams = myCapsToSendParams(sendVideoCaps, answer.recvVideoCaps);
var audioRecvParams = myCapsToRecvParams(recvAudioCaps, answer.sendAudioCaps);
var videoRecvParams = myCapsToRecvParams(recvVideoCaps, answer.sendVideoCaps);
//
// If the responder wishes to enable bundle, we will enable it
if (answer.bundle) {
// Only start the single transport that is needed
// No need for the ICE Transport Controller.
audioRtpIceTransport.start(answer.audioRtpIce, RTCIceRole.controlling);
audioRtpDtlsTransport.start(remote.audioRtpDtls);
//
// Replace the transport on the Sender and Receiver objects
//
audioSender.setTransport(audioRtpDtlsTransport);
videoSender.setTransport(audioRtpDtlsTransport);
audioReceiver.setTransport(audioRtpDtlsTransport);
videoReceiver.setTransport(audioRtpDtlsTransoprt);
// If bundle was requested, then implicitly RTP/RTCP mux must be done
answer.rtcpMux = true;
} else {
if (answer.rtcpMux){
// We don't want to bundle, but we do want to multiplex RTP/RTCP
// Create the ICE Transport Controller object
var controller = new RTCIceTransportController();
controller.addTransport(audioRtpIceTransport);
controller.addTransport(videoRtpIceTransport);
// Start the audio and video ICE transports
audioRtpIceTransport.start(answer.audioRtpIce, RTCIceRole.controlling);
videoRtpIceTransport.start(answer.videoRtpIce, RTCIceRole.controlling);
// Start the audio and video DTLS transports
audioRtpDtlsTransport.start(answer.audioRtpDtls);
videoRtpDtlsTransport.start(answer.videoRtpDtls);
// Replace the transport on the Sender and Receiver objects
//
audioSender.setTransport(audioRtpDtlsTransport);
videoSender.setTransport(videoRtpDtlsTransport);
audioReceiver.setTransport(audioRtpDtlsTransport);
videoReceiver.setTransport(videoRtpDtlsTransoprt);
};
// Check if the responder does not want to bundle
// and does not want RTP/RTCP multiplexing
if (!answer.rtcpMux) {
// Create the ICE Transport Controller object
var controller = new RTCIceTransportController();
controller.addTransport(audioRtpIceTransport);
controller.addTransport(videoRtpIceTransport);
// Start the ICE transports that are needed
audioRtpIceTransport.start(answer.audioRtpIce, RTCIceRole.controlling);
audioRtcpIceTransport.start(answer.audioRtcpIce, RTCIceRole.controlling);
videoRtpIceTransport.start(answer.videoRtpIce, RTCIceRole.controlling);
videoRtcpIceTransport.start(answer.videoRtcpIce, RTCIceRole.controlling);
// Start the DTLS transports that are needed
audioRtpDtlsTransport.start(answer.audioRtpDtls);
audioRtcpDtlsTransport.start(answer.audioRtcpDtls);
videoRtpDtlsTransport.start(answer.videoRtpDtls);
videoRtcpDtlsTransport.start(answer.videoRtcpDtls);
// Disable RTP/RTCP multiplexing
audioSendParams.rtcp.mux = false;
videoSendParams.rtcp.mux = false;
audioRecvParams.rtcp.mux = false;
videoRecvParams.rtcp.mux = false;
};
//
// Set the audio and video send and receive parameters.
audioSender.send(audioSendParams);
videoSender.send(videoSendParams);
audioReceiver.receive(audioRecvParams);
videoReceiver.receive(videoRecvParams);
});
// Now we can render/play
// audioReceiver.track and videoReceiver.track.
The RTCIceListener Object
The
RTCIceListener
enables an endpoint to construct multiple
RTCIceTransport
objects from a set of local ICE parameters,
enabling usage scenarios such as parallel forking.
Overview
An
RTCIceListener
instance is associated to an
RTCIceTransport
Operation
An
RTCIceListener
instance is optionally constructed from an
RTCIceOptions
object,
or an
RTCIceListener
is automatically constructed.
Interface Definition
RTCIceOptions getOptions()
Retrieves the ICE options.
void setOptions(RTCIceOptions options)
Sets the ICE options, if they weren't present in the constructor.
attribute EventHandler? onerror
This event handler, of event handler type
error
MUST
be supported by all objects implementing the
RTCIceListener
interface.
If TURN credentials are invalid, then this event
MUST
be fired.
Example
// Example to demonstrate forking when RTP and RTCP are not multiplexed.

var iceOptions = ...;
var iceRtpListener = new RTCIceListener(iceOptions);
var iceBaseRtpTransport = new RTCIceTransport(iceRtpListener);
//create the RTCP ICE transport
var iceBaseRtcpTransport = iceBaseRtpTransport.createAssociatedTransport();

mySendInitiate(
"icertp": iceBaseRtpTransport.getLocalParameters(),
"icertcp": iceBaseRtcpTransport.getLocalParameters()
},
function(response) {
// We may get N responses
var iceRtpTransport = new RTCIceTransport(iceRtpListener);
// Create new ice RTCP transport based on the (implicitly created) iceListener
var iceRtcpTransport = iceRtpTransport.createAssociatedTransport();

// check to make sure the RTCRtpIceListener objects are set up as expected.
assert(iceRtpTransport.iceListener == iceBaseRtpTransport.iceListener);
assert(iceRtcpTransport.iceListener == iceBaseRtcpTransport.iceListener);

iceRtpTransport.start(response.icertp, RTCIceRole.controlling);
iceRtcpTransport.start(response.icertcp, RTCIceRole.controlling);
// ... setup DTLS, RTP, SCTP, etc.
});

iceBaseRtpTransport.onlocalcandidate = mySendLocalRtpCandidate;
iceBaseRtcpTransport.onlocalcandidate = mySendLocalRtcpCandidate;
The RTCRtpSender Object
The
RTCRtpSender
includes information relating to the RTP sender.
Overview
An
RTCRtpSender
instance is associated to a sending
MediaStreamTrack
and provides RTC related methods to it.
Operation
RTCRtpSender
instance is constructed from an
MediaStreamTrack
object and
associated to an
RTCDtlsTransport
Interface Definition
readonly attribute MediaStreamTrack track
The associated
MediaStreamTrack
instance.
readonly attribute RTCDtlsTransport transport
The associated RTP
RTCDtlsTransport
instance.
readonly attribute RTCDtlsTransport rtcpTransport
The associated RTCP
RTCDtlsTransport
instance.
RTCRtpParameters getParameters()
Retrieve the
RTCRtpParameters
currently configured for use by this object.
void setTransport(RTCDtlsTransport transport, optional RTCDtlsTransport rtcpTransport)
Set the RTP
RTCDtlsTransport
(and if used) RTCP
RTCDtlsTransport
void setTrack(MediaStreamTrack track)
Set the
MediaStreamTrack
static RTCRtpCapabilities getCapabilities(optional DOMString kind)
Obtain the sender capabilities, based on
kind
. If
kind
is
omitted, then all capabilities are returned.
Promise send(RTCRtpParameters parameters)
Media to be sent is controlled by parameters.
The sender starts sending when
send()
is called and stops sending when
stop()
is called.
If
send
is called with invalid parameters,
throw an
InvalidParameters
exception.
By the time the Promise is fulfilled,
RTCRtcpParameters.ssrc
RTCRtpParameters.cname
and
RTCRtpEncodingParameters.ssrc
MUST
be set,
so that the values (which may be needed for signaling) can be
retrieved by calling
getParameters()
void stop()
Stops sending the track on the wire.
Stop is final as in
MediaStreamTrack.stop()
attribute EventHandler? onerror
This event handler, of event handler type
error
MUST
be supported by all objects implementing the
RTCRtpSender
interface.
The RTCRtpReceiver Object
The
RTCRtpReceiver
includes information relating to the RTP receiver.
Overview
An
RTCRtpReceiver
instance is associated to a receiving
MediaStreamTrack
and provides RTC related methods to it.
Operation
RTCRtpReceiver
instance is constructed from an
RTCDtlsTransport
object.
Interface Definition
readonly attribute MediaStreamTrack? track
The associated
MediaStreamTrack
instance.
readonly attribute RTCDtlsTransport transport
The associated RTP
RTCDtlsTransport
instance.
readonly attribute RTCDtlsTransport rtcpTransport
The associated RTCP
RTCDtlsTransport
instance.
RTCRtpParameters getParameters()
Retrieve the
RTCRtpParameters
currently configured for use by this object.
void setTransport(RTCDtlsTransport transport, optional RTCDtlsTransport rtcpTransport)
Set the RTP
RTCDtlsTransport
(and if used) RTCP
RTCDtlsTransport
static RTCRtpCapabilities getCapabilities(optional DOMString kind)
Obtain the receiver capabilities, based on
kind
. If
kind
is omitted, then
all capabilities are returned.
void receive(RTCRtpParameters parameters)
Media to be received is controlled by the receive parameters.
The receiver starts receiving when the
receive()
is called and stopped when
stop()
is called.
If
receive
is called with invalid parameters,
throw an
InvalidParameters
exception.
void stop()
Stops receiving the track on the wire. Stop is final like
MediaStreamTrack
attribute EventHandler? onerror
This event handler, of event handler type
error
MUST
be supported by all objects implementing the
RTCRtpReceiver
interface.
Examples
// Assume we already have a way to signal, a transport
// (RTCDtlsTransport), and audio and video tracks. This is an example
// of how to offer them and get back an answer with audio and
// video tracks, and begin sending and receiving them.
// The example assumes that RTP and RTCP are multiplexed.
function myInitiate(mySignaller, transport, audioTrack, videoTrack) {
var audioSender = new RTCRtpSender(audioTrack, transport);
var videoSender = new RTCRtpSender(videoTrack, transport);
var audioReceiver = new RTCRtpReceiver(transport);
var videoReceiver = new RTCRtpReceiver(transport);

// Retrieve the audio and video receiver capabilities
var recvAudioCaps = RTCRtpReceiver.getCapabilities("audio");
var recvVideoCaps = RTCRtpReceiver.getCapabilities("video");
// Retrieve the audio and video sender capabilities
var sendAudioCaps = RTCRtpSender.getCapabilities("audio");
var sendVideoCaps = RTCRtpSender.getCapabilities("video");

mySignaller.myOfferTracks({
// The initiator offers its receiver and sender capabilities.
"recvAudioCaps": recvAudioCaps,
"recvVideoCaps": recvVideoCaps,
"sendAudioCaps": sendAudioCaps,
"sendVideoCaps": sendVideoCaps
}, function(answer) {
// The responder answers with its receiver capabilities

// Derive the send and receive parameters
var audioSendParams = myCapsToSendParams(sendAudioCaps, answer.recvAudioCaps);
var videoSendParams = myCapsToSendParams(sendVideoCaps, answer.recvVideoCaps);
var audioRecvParams = myCapsToRecvParams(recvAudioCaps, answer.sendAudioCaps);
var videoRecvParams = myCapsToRecvParams(recvVideoCaps, answer.sendVideoCaps);
audioSender.send(audioSendParams);
videoSender.send(videoSendParams);
audioReceiver.receive(audioRecvParams);
videoReceiver.receive(videoRecvParams);

// Now we can render/play
// audioReceiver.track and videoReceiver.track.
});
// Assume we already have a way to signal, a transport (RTCDtlsTransport)
// and audio and video tracks. This is an example of how to answer an
// offer with audio and video tracks, and begin sending and receiving them.
// The example assumes that RTP and RTCP are multiplexed.
function myAccept(
mySignaller, remote, transport, audioTrack, videoTrack) {
var audioSender = new RTCRtpSender(audioTrack, transport);
var videoSender = new RTCRtpSender(videoTrack, transport);
var audioReceiver = new RTCRtpReceiver(transport);
var videoReceiver = new RTCRtpReceiver(transport);

// Retrieve the send and receive capabilities
var recvAudioCaps = RTCRtpReceiver.getCapabilities("audio");
var recvVideoCaps = RTCRtpReceiver.getCapabilities("video");
var sendAudioCaps = RTCRtpSender.getCapabilities("audio");
var sendVideoCaps = RTCRtpSender.getCapabilities("video");

mySignaller.myAnswerTracks({
"recvAudioCaps": recvAudioCaps,
"recvVideoCaps": recvVideoCaps,
"sendAudioCaps": sendAudioCaps,
"sendVideoCaps": sendVideoCaps
});

// Derive the send and receive parameters using Javascript functions defined in Section 15.2.
var audioSendParams = myCapsToSendParams(sendAudioCaps, remote.recvAudioCaps);
var videoSendParams = myCapsToSendParams(sendVideoCaps, remote.recvVideoCaps);
var audioRecvParams = myCapsToRecvParams(recvAudioCaps, remote.sendAudioCaps);
var videoRecvParams = myCapsToRecvParams(recvVideoCaps, remote.sendVideoCaps);
audioSender.send(audioSendParams);
videoSender.send(videoSendParams);
audioReceiver.receive(audioRecvParams);
videoReceiver.receive(videoRecvParams);

// Now we can render/play
// audioReceiver.track and videoReceiver.track.
The RTCRtpListener Object
The
RTCRtpListener
listens to RTP packets received from the DTLS transport.
Overview
An
RTCRtpListener
instance is associated to an
RTCDtlsTransport
Operation
An
RTCRtpListener
instance is constructed from an
RTCDtlsTransport
object.
Interface Definition
readonly attribute RTCDtlsTransport transport
The RTP
RTCDtlsTransport
instance.
attribute EventHandler? onunhandledrtp
The event handler which handles the
RTCRtpUnhandledRtpEvent
RTCRtpUnhandledEvent
An
unhandledrtp
event is fired if the
RTCRtpListener
detects an
RTP stream that is not configured to be processed by an
existing
RTCRtpReceiver
object. The amount of buffering to be provided for unhandled
RTP streams is recommended to be strictly limited to protect against denial of service attacks.
To determine whether an RTP stream is configured to be processed by an existing
RTCRtpReceiver
object,
the
RTCRtpListener
attempts to match the values of an incoming RTP packet's
Payload Type and SSRC fields as well as the value of its receiverId RTP extension (if present) against the
RTCRtpReceiver.RTCRtpParameters.RTCRtpCodecParameters.payLoadType
RTCRtpReceiver.RTCRtpParameters.RTCRtpEncodingParameters.ssrc
and
RTCRtpReceiver.RTCRtpParameters.receiverId
attributes of configured
RTCRtpReceiver
objects.
TODO: provide details of matching behavior, along with examples.
The
unhandledrtp
event of the
RTCRtpListener
object uses
the
RTCRtpUnhandledEvent
interface.
Firing an
unhandledrtp
event named
with an
RTCRtpUnhandled
stream
means that an event with the name
which does not bubble (except where otherwise stated) and is not
cancelable (except where otherwise stated), and which uses the
RTCRtpUnhandledEvent
interface with the
stream
attribute set to an
RTCRtpUnhandled
object,
MUST
be
created and dispatched at the given target.
readonly attribute RTCRtpUnhandled stream
The
stream
attribute is the
RTCRtpUnhandled
object with the characteristics of the RTP stream
that caused the event.
RTCRtpUnhandled? stream
The characteristics of the RTP stream that caused the event.
Example
Dictionaries related to Rtp
RTCRtpUnhandled
RTCRtpUnhandled
provides information on the RTP packet that caused the
RTCRtpUnhandled
event.
unsigned long ssrc
The SSRC in the RTP stream triggering the
unhandledrtp
event.
payloadtype payloadType
The Payload Type value in the RTP stream triggering the
unhandledrtp
event.
DOMString receiverId
The value of the AppId header extension in the RTP stream triggering the
unhandledrtp
event, if present.
dictionary RTCRtpCapabilities
The
RTCRtpCapabilities
object expresses the capabilities of
RTCRtpSender
and
RTCRtpReceiver
objects. Features which are mandatory to implement in [[!RTP-USAGE]], such as RTP/RTCP multiplexing [[!RFC5761]] and
reduced-size RTCP [[!RFC5506]] are assumed to be available and are therefore not included in
RTCRtpCapabilities
although these features can be set via
RTCRtpParameters
sequence codecs
Supported codecs.
sequence headerExtensions
Supported RTP header extensions.
sequence fecMechanisms
Supported FEC mechanisms; valid values include "xor" [[RFC5109]] and "raptor" [[RFC6681]].
dictionary RTCRtcpFeedback
RTCRtcpFeedback
provides information on RTCP feedback messages.
DOMString type
Valid values for
type
are the "RTCP Feedback" Attribute Values enumerated in [[!IANA-SDP-14]] ("ack", "ccm", "nack", etc.).
DOMString parameter
For a
type
of "ack" or "nack", valid values for
parameters
are the "ack" and "nack" Attribute Values enumerated in [[!IANA-SDP-15]] ("sli", "rpsi", etc.).
For a
type
of "ccm", valid values for
parameters
are the "Codec Control Messages" enumerated in [[!IANA-SDP-19]] ("fir", "tmmbr" (includes "tmmbn"), etc.).
dictionary RTCRtpCodecCapability
RTCRtpCodecCapability
provides information on the capabilities of a codec.
DOMString name=""
The MIME media type, if set, empty string otherwise.
Valid MIME media types are listed in [[!IANA-RTP-2]].
DOMString kind
The media supported by the codec: "audio", "video" or "" for both.
unsigned long clockRate
Codec clock rate expressed in Hertz, null if unset.
payloadtype preferredPayloadType
Added to make it possible for the sender and receiver to pick a
matching payload type when creating sender and receiver parameters.
unsigned long numChannels=1
The number of channels supported (e.g. stereo); one by default.
For video, this will be null.
sequence rtcpFeedback
Transport layer and codec-specific feedback messages for this codec.
Dictionary parameters
Codec-specific parameters that must be signaled to the remote party.
Dictionary options
Codec-specific parameters available for signaling.
unsigned short maxTemporalLayers = 0
Maximum number of temporal layer extensions supported by this codec (e.g. a value of 1 indicates support for up to 2 temporal layers). A value of 0 indicates no support for temporal scalability.
unsigned short maxSpatialLayers = 0
Maximum number of spatial layer extensions supported by this codec (e.g. a value of 1 indicates support for up to 2 spatial layers). A value of 0 indicates no support for spatial scalability.
boolean svcMultiStreamSupport=false
Whether the implementation can send SVC layers utilizing distinct SSRCs. The default is false.
Only set if the codec supports temporal, spatial or quality scalability.
If the codec does not support temporal,
Codec capability parameters
The capability parameters for commonly implemented codecs are provided below.
Opus
The following capabilities are defined for Opus, as noted in [[!OPUS-RTP]] Section 6.1:
Property Name
Values
Notes
maxplaybackrate
unsigned long
A hint about the maximum output sampling rate that
the receiver is capable of rendering in Hz.
minptime
unsigned long
The decoder's minimum length of time in milliseconds rounded up to the next full integer value.
stereo
boolean
Specifies whether the decoder prefers receiving stereo (if true) or mono signals (if false).
VP8
The following receiver capabilities are defined for VP8, as noted in [[VP8-RTP]] Section 6.1:
Property Name
Values
Notes
max-fr
unsigned long
This capability indicates the maximum frame rate in frames per second that the decoder is capable of decoding.
max-fs
unsigned long long
This capability indicates the maximum frame size in macroblocks that the decoder is capable of decoding.
H.264
The following capabilities are defined for H.264, as noted in [[RFC6184]] Section 8.1.
NOTE: Update this section as [[!RTCWEB-VIDEO]] evolves.
Property Name
Values
Notes
max-recv-level
unsigned long
Indicates the highest level a receiver supports.
packetization-mode
unsigned short
An integer in the range of 0 to 2 which indicates which packetization-mode this implementation supports.
dictionary RTCRtpParameters
RTCRtpParameters
contains the RTP stack settings.
DOMString receiverId=""
The receiverId assigned to the RTP stream, if any, empty string if unset.
In an
RTCRtpReceiver
object, this corresponds to
recv-appId
defined in [[APPID]]. In an
RTCRtpSender
object, it corresponds to the
appId
This is a stable identifier that can be defined and assign to any RTP stream rather than relying on an SSRC.
An SSRC is randomly generated and can change arbitrarily due to conflicts with other SSRCs, whereas
the
receiverId
has a value
whose meaning can be defined in advance between RTP
sender and receiver, assisting in RTP demultiplexing.
sequence codecs
The codecs to send or receive (could include RED, RTX and CN as well).
sequence headerExtensions
Configured header extensions.
sequence encodings
The "encodings" or "layers" to be used for things like simulcast, Scalable Video Coding, RTX, FEC, etc.
RTCRtcpParameters rtcp
Parameters to configure RTCP.
dictionary RTCRtcpParameters
RTCRtcpParameters
provides information on RTCP settings.
unsigned long ssrc
The SSRC to be used in the Receiver Report "SSRC of packet sender" field as defined in [[!RFC3550]] Section 6.4.2.
If unset, the SSRC is chosen by the browser.
Note that the browser may change the SSRC in event of a collision, as described in [[!RFC3550]].
DOMString cname
The Canonical Name (CNAME) used by RTCP (e.g. in SDES messages). Guidelines for CNAME generation are provided in [[!RTP-USAGE]] Section 4.9.
By default, ORTC implementations SHOULD set the CNAME to be the same within all
RTCRtcpParameter
objects created within the
same Javascript sandbox. However, for backward compatibility with WebRTC 1.0, applications MAY set the cname to an alternative value.
boolean reducedSize=false
Whether reduced size RTCP [[!RFC5506]] is configured (if true) or compound RTCP as specified in [[!RFC3550]] (if false).
The default is
false
boolean mux=true
Whether RTP and RTCP are multiplexed, as specified in [[!RFC5761]].
The default is
true
. If set to
false
, the
RTCIceTransport
MUST
have an associated
RTCIceTransport
object with
component
of "RTCP",
in which case RTCP will be sent on the associated
RTCIceTransport
dictionary RTCRtpCodecParameters
RTCRtpCodecParameters
provides information on codec settings.
DOMString name=""
The MIME media type, if set, empty string otherwise.
Valid MIME media types are listed in [[!IANA-RTP-2]].
payloadtype payloadType
The value that goes in the RTP Payload Type Field [[!RFC3550]]. The
payloadType
must always be provided both within
RTCRtpSender
and
RTCRtpReceiver
objects.
unsigned long clockRate
Codec clock rate expressed in Hertz, null if unset.
unsigned long numChannels=1
The number of channels supported (e.g. stereo); one by default. For video, this will be null.
sequence rtcpFeedback
Transport layer and codec-specific feedback messages for this codec.
Dictionary parameters
Codec-specific parameters available for signaling.
Codec parameters
The capabilities for commonly implemented codecs described in Section 9.4.2, are also
used as codec parameters, with
RTCRtpCodecCapability.parameters
of the receiver used as
RTCRtpCodecParameters.parameters
of the sender, and
RTCRtpCodecCapability.parameters
of the sender used as
RTCRtpCodecParameters.parameters
of the receiver, with the Property Name
and Values unchanged.
dictionary RTCRtpEncodingParameters
unsigned long ssrc
The SSRC for this layering/encoding, null if unset.
If an SSRC is not present in a receive object, any
SSRC will match. If an SSRC is not present in a sender object, the browser will choose.
payloadtype codecPayloadType
For per-encoding codec specifications, give the codec Payload Type here.
If null, the browser will choose.
RTCRtpFecParameters fec
Specifies the FEC mechanism if present.
RTCRtpRtxParameters rtx
Specifies the RTX mechanism if present.
double priority=1.0
The higher the value, the more the bits will be given to each as available bandwidth goes up. Default is 1.0.
This parameter is ignored in scalable video coding.
double maxBitrate
Ramp up resolution/quality/framerate until this bitrate, if present. Summed when using dependent layers.
This parameter is ignored in scalable video coding, or in an
RTCRtpReceiver
object.
double minQuality=0
Never send less than this quality. 1.0 = maximum attainable quality.
This parameter is ignored in scalable video encoding, or in an
RTCRtpReceiver
object.
double framerateBias=0.5
What to give more bits to, if available, null if unset. 0.0 = strongly favor resolution or 1.0 = strongly favor
framerate. 0.5 = neither (default). This parameter is ignored in a scalable video encoding,
or in an
RTCRtpReceiver
object.
double resolutionScale
Fraction of the input resolution to be encoded, or die trying. 1.0 = full resolution.
For scalable video coding,
resolutionScale
refers to
the aggregate fraction of the input resolution achieved by this
layer when combined with all
dependent layers.
double framerateScale
Fraction of the input framerate to be encoded. 1.0 = full framerate.
For scalable video coding,
framerateScale
refers to the aggregate fraction of the input framerate achieved by this layer
when combined with all dependent layers.
boolean active=true
Whether the sender or receiver is active. If false, don't send any media right now.
Disable is different than omitting the encoding; it can keep resources
available to re-enable more quickly than re-adding. Plus, it still sends RTCP. Default is active.
DOMString encodingId
An identifier for the encoding object. This identifier should be unique within the scope of the
localized sequence of
RTCRtpEncodingParameters
for any given
RTCRtpParameters
object.
If encodings contained within sequences of other
RTCRtpParameters
objects are dependent
upon this encoding identifier, the identifier should be globally unique (unless two or more
encodings are intentionally referencing the same dependency
RTCRtpEncodingParameters
such
as described in [[RFC5583]] Section 6.5a).
sequence dependencyEncodingIds
Just the IDs. Within this specification it resolves to
encodingIds
within the same sequence only.
In the future if MST were to be supported, then if searching the same sequence did not produce a match, then a global search
would be carried out.
Examples
Basic Example
//Send a thumbnail along with regular size
var encodings = [{ ssrc: 1, priority: 1.0 }]

// Control the resolution and framerate with a different track and RtpSender.
var encodings = [{ ssrc: 2,
// Prioritize the thumbnail over the main video.
priority: 10.0 }];

// Sign Language (need high framerate, but don't get too bad quality)
var encodings = [{ minQuality: 0.2, framerateBias: 1.0 }];

// Screencast (High quality, framerate can be low)
var encodings = [{ framerateBias: 0.0 }];

//Remote Desktop (High framerate, must not downscale)
var encodings = [{ framerateBias: 1.0 }];

// Audio more important than video
var audioEncodings = [{ priority: 10.0 }];
var videoEncodings = [{ priority: 0.1 }];

//Video more important than audio
var audioEncodings = [{ priority: 0.1 }];
var videoEncodings = [{ priority: 10.0 }];

//Crank up the quality
var encodings = [{ maxBitrate: 10000000 }];

//Keep the bandwidth low
var encodings = [{ maxBitrate: 100000 }];
Temporal Scalability
Example of 3-layer temporal scalability encoding
var encodings =[{
// Base framerate is one quarter of the input framerate
encodingId: "0",
framerateScale: 0.25
}, {
// Temporal enhancement (half the input framerate when combined with the base layer)
encodingId: "1",
dependencyEncodingIds: ["0"]
framerateScale: 0.5
}, {
// Another temporal enhancement layer (full input framerate when all layers combined)
encodingId: "2",
dependencyEncodingIds: ["0", "1"]
framerateScale: 1.0
}]

Example of 3-layer temporal scalability with all but bottom layer disabled
var encodings =[{
encodingId: "0",
framerateScale: 0.25
}, {
encodingId: "1",
dependencyEncodingIds: ["0"],
framerateScale: 0.5,
active: false
}, {
encodingId: "2",
dependencyEncodingIds: ["0", "1"],
framerateScale: 1.0,
active: false
}];
Below is a representation of a 3-layer temporal scalability encoding. In the diagram, I0 is the base layer I-frame,
and P0 represents base-layer P-frames. P1 represents the first temporal enhancement layer, and P2 represents the
second temporal enhancement layer.
Spatial Simulcast
Example of 3-layer spatial simulcast
var encodings =[{
// Simulcast layer at one quarter scale
encodingId: "0",
resolutionScale: 0.25
}, {
// Simulcast layer at one half scale
encodingId: "1",
resolutionScale: 0.5
}, {
// Simulcast layer at full scale
encodingId: "2",
resolutionScale: 1.0
}]

Example of 3-layer spatial simulcast with all but bottom layer disabled
var encodings =[{
encodingId: "0",
resolutionScale: 0.25
}, {
encodingId: "1",
resolutionScale: 0.5,
active: false
}, {
encodingId: "2",
resolutionScale: 1.0,
active: false
}];

Example of 2-layer spatial simulcast combined with 2-layer temporal scalability
var encodings =[{
// Base layer (half the input framerate, half the input resolution)
encodingId: "0",
resolutionScale: 0.5,
framerateScale: 0.5
}, {
// Enhanced resolution Base layer (half the input framerate, full input resolution)
encodingId: "E0",
resolutionScale: 1.0,
framerateScale: 0.5
}, {
// Temporal enhancement to the base layer (full input framerate, half resolution)
encodingId: "1",
dependencyEncodingIds: ["0"],
resolutionScale: 0.5,
framerateScale: 1.0
}, {
// Temporal enhancement to enhanced resolution base layer (full input framerate and resolution)
encodingId: "E1",
dependencyEncodingIds: ["E0"],
resolutionScale: 1.0,
framerateScale: 1.0
}]
Below is a representation of 2-layer temporal scalability combined with 2-layer spatial simulcast.
Solid arrows represent temporal prediction.
In the diagram, I0 is the base-layer I-frame, and P0 represents base-layer P-frames.
EI0 is an enhanced resolution base-layer I-frame, and EP0 represents P-frames within the enhanced resolution base layer.
P1 represents the first temporal enhancement layer, and EP1 represents a temporal enhancement to the
enhanced resolution simulcast base-layer.
Spatial Scalability
Example of 3-layer spatial scalability encoding
var encodings =[{
// Base layer with one quarter input resolution
encodingId: "0",
resolutionScale: 0.25
}, {
// Spatial enhancement layer providing half input resolution when combined with the base layer
encodingId: "1",
dependencyEncodingIds: ["0"]
resolutionScale: 0.5
}, {
// Additional spatial enhancement layer providing full input resolution when combined with all layers
encodingId: "2",
dependencyEncodingIds: ["0", "1"]
resolutionScale: 1.0
}]

Example of 3-layer spatial scalability with all but bottom layer disabled
var encodings =[{
encodingId: "0",
resolutionScale: 0.25
}, {
encodingId: "1",
dependencyEncodingIds: ["0"],
resolutionScale: 0.5,
active: false
}, {
encodingId: "2",
dependencyEncodingIds: ["0", "1"],
resolutionScale: 1.0,
active: false
}];

Example of 2-layer spatial scalability combined with 2-layer temporal scalability
var encodings =[{
// Base layer (half input framerate, half resolution)
encodingId: "0",
resolutionScale: 0.5,
framerateScale: 0.5
}, {
// Temporal enhancement to base layer (full input framerate, half resolution)
encodingId: "1",
dependencyEncodingIds: ["0"],
resolutionScale: 0.5,
framerateScale: 1.0
}, {
// Spatial enhancement to base layer (half input framerate, full resolution)
encodingId: "E0",
dependencyEncodingIds: ["0"],
resolutionScale: 1.0,
framerateScale: 0.5
}, {
// Spatial enhancement to temporal enhancement (full input framerate, full resolution)
encodingId: "E1",
dependencyEncodingIds: ["E0", "1"],
resolutionScale: 1.0,
framerateScale: 1.0
}]
Below is a representation of 2-layer temporal scalability combined with 2-layer spatial scalability.
Solid arrows represent temporal prediction and dashed arrows represent inter-layer prediction.
In the diagram, I0 is the base-layer I-frame, and EI0 is an intra spatial enhancement.
P0 represents base-layer P-frames, and P1 represents the first temporal enhancement layer.
EP0 represents a resolution enhancement to the base-layer P frames, and EP1 represents a resolution enhancement to the
second temporal layer P-frames.
dictionary RTCRtpFecParameters
unsigned long ssrc
The SSRC to use for FEC.
If unset in an
RTCRtpSender
object, the browser will choose.
DOMString mechanism
The Forward Error Correction (FEC) mechanism to use.
dictionary RTCRtpRtxParameters
unsigned long ssrc
The SSRC to use for RTX.
If unset in an
RTCRtpSender
object, the browser will choose.
dictionary RTCRtpHeaderExtension
DOMString kind
The media supported by the header extension: "audio" for an audio codec,
"video" for a video codec, and "" for both.
DOMString uri
The URI of the RTP header extension, as defined in [[!RFC5285]].
unsigned short preferredId
The preferred ID value that goes in the packet.
boolean preferredEncrypt=false
If true, it is preferred that the value in the header be
encrypted as per [[!RFC6904]]. Default is to prefer unencrypted.
dictionary RTCRtpHeaderExtensionParameters
DOMString uri
The URI of the RTP header extension, as defined in [[!RFC5285]].
unsigned short id
The value that goes in the packet.
boolean encrypt=false
If true, the value in the header is encrypted as per [[!RFC6904]]. Default is unencrypted.
RTP header extensions
Registered RTP header extensions are listed in [[!IANA-RTP-10]]. Header extensions mentioned in [[!RTP-USAGE]] include:
Header Extension
Reference
Notes
Rapid Synchronization
[[RFC6051]]
This extension enables carriage of an NTP-format timestamp, as defined in [[!RFC6051]] Section 3.3.
Client-to-Mixer Audio Level
[[!RFC6464]]
This extension indicates the audio level of the audio sample carried in an RTP packet.
Mixer-to-Client Audio Level
[[RFC6465]]
This extension indicates the audio level of individual conference participants.
Application ID
[[APPID]]
This extension defines an Application ID which can be used to identify an RTP stream.
The RTCDtmfSender Object
This section of the ORTC API specification depends on the WebRTC 1.0 DtmfSender API, and needs to be synchronized once it is updated.
Overview
An
RTCDtmfSender
instance allows sending DTMF tones to/from the remote peer, as per [[!RFC4733]].
Operation
An
RTCDtmfSender
object is constructed from an
RTCRtpSender
object.
Interface Definition
readonly attribute boolean
canInsertDTMF
Whether the
RTCDtmfSender
is capable of sending DTMF.
void insertDTMF(in DOMString tones, optional long duration, long
interToneGap)
readonly attribute
RTCRtpSender
sender
The
RTCRtpSender
instance
attribute EventHandler ontonechange
The
ontonechange
event handler uses the
RTCDTMFToneChangeEvent
interface to return the
character for each tone as it is played out.
readonly attribute DOMString toneBuffer
The
toneBuffer
attribute returns a list of the
tones remaining to be played out.
readonly attribute long duration
The
duration
attribute returns the current tone duration
value in milliseconds. This value will be the value last set via the
insertDTMF()
method, or the default value of 70 ms if
insertDTMF()
was called without specifying the duration.
readonly attribute long interToneGap
The
interToneGap
attribute returns the current value of
the between-tone gap. This value will be the value last set via the
insertDTMF()
method, or the default value of 70
ms if
insertDTMF()
was called without specifying
the
interToneGap.
RTCDTMFToneChangeEvent
The tonechange event uses the
RTCDTMFToneChangeEvent
interface.
Firing an tonechange event named
with a DOMString
tone
means
that an event with the name
, which does not bubble (except
where otherwise stated) and is not cancelable (except where otherwise
stated), and which uses the
RTCDTMFToneChangeEvent
interface with the
tone
attribute set to
tone
, MUST be created and dispatched at the given target.
readonly attribute DOMString tone
The
tone
attribute contains the character for the tone that has just begun
playout (see
insertDTMF()
). If the value is the
empty string, it indicates that the previous tone has completed
playback.
DOMString tone=""
The
tone
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 are equivalent to A to D.
The character ',' indicates a delay of 2 seconds before processing the next character in the tones parameter.
Unrecognized characters are ignored.
DTMF Example
Examples assume that
sendObject
is an
RTCRtpSender
object.
Sending the DTMF signal "1234" with 500 ms duration per tone:
var sender = new RTCDtmfSender(sendObject);
if (sender.canInsertDTMF) {
var duration = 500;
sender.insertDTMF("1234", duration);
} else
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):
var sender = new RTCDtmfSender(sendObject);
sender.ontonechange = function (e) {
if (!e.tone)
return;
// light up the key when playout starts
lightKey(e.tone);
// turn off the light after tone duration
setTimeout(lightKey, sender.duration, "");
};
sender.insertDTMF("1234");
Send a 1-second "1" tone followed by a 2-second "2" tone:
var sender = new RTCDtmfSender(sendObject);
sender.ontonechange = function (e) {
if (e.tone == "1")
sender.insertDTMF("2", 2000);
};
sender.insertDTMF("1", 1000);
It is always safe to append to the tone buffer. This example appends
before any tone playout has started as well as during playout.
var sender = new RTCDtmfSender(sendObject);
sender.insertDTMF("123");
// append more tones to the tone buffer before playout has begun
sender.insertDTMF(sender.toneBuffer + "456");

sender.ontonechange = function (e) {
if (e.tone == "1")
// append more tones when playout has begun
sender.insertDTMF(sender.toneBuffer + "789");
};
Send the DTMF signal "123" and abort after sending "2".
var sender = new RTCDtmfSender(sendObject);
sender.ontonechange = function (e) {
if (e.tone == "2")
// empty the buffer to not play any tone after "2"
sender.insertDTMF("");
};
sender.insertDTMF("123");
The RTCDataChannel Object
Overview
An
RTCDataChannel
class instance allows sending data messages to/from the remote peer.
Operation
An
RTCDataChannel
object is constructed from an
RTCDataTransport
object and
an
RTCDataChannelParameters
object.
Interface Definition
The
RTCDataChannel
interface represents a bi-directional data channel between
two peers.
There are two ways to establish a connection with
RTCDataChannel
The first way is to construct an
RTCDataChannel
at one of the peers with the
RTCDataChannelParameters
negotiated
attribute unset or set to its default value false.
This will announce the new channel in-band and trigger an
ondatachannel
event 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
RTCDataChannelParameters
negotiated
dictionary member set to true, and signal out-of-band (e.g. via a web server) to the other
side that it should create a corresponding
RTCDataChannel
with the
RTCDataChannelParameters
negotiated
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.
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.
readonly attribute RTCDataTransport transport
The readonly attribute referring to the related transport object.
readonly attribute RTCDataChannelParameters parameters
The parameters applying to this data channel.
readonly attribute RTCDataChannelState readyState
The
readyState
attribute represents the state of the
RTCDataChannel
object.
It
MUST
return the value to which the user agent last set it (as defined by the processing model algorithms).
readonly attribute unsigned long bufferedAmount
The
bufferedAmount
attribute
MUST
return the number of bytes of application data
(UTF-8 text and binary data) that have been queued using send() but that, as of the last time
the event loop started executing a task, had not yet been transmitted to the network.
This includes any text sent during the execution of the current task, regardless of whether the
user agent is able to transmit text asynchronously with script execution.
This does not include framing overhead incurred by the protocol, or buffering done by the
operating system or network hardware.
If the channel is closed, this attribute's value will only increase with each call to the
send() method (the attribute does not reset to zero once the channel closes).
attribute DOMString binaryType
The
binaryType
attribute
MUST
, on getting, return the value to which it was last set.
On setting, the user agent
MUST
set the IDL attribute to the new value.
When an
RTCDataChannel
object is constructed, the
binaryType
attribute
MUST
be initialized to the string 'blob'.
This attribute controls how binary data is exposed to scripts.
See the [[WEBSOCKETS-API]] for more information.
void 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:
1. Let channel be the
RTCDataChannel
object which is about to be closed.
2. If channel's
readyState
is closing or closed, then abort these steps.
3. Set channel's
readyState
attribute to closing.
4. If the closing procedure has not started yet, start it.
attribute EventHandler onopen
This event handler, of event handler type
open
MUST
be supported by all objects implementing the RTCDataChannel interface.
attribute EventHandler onerror
This event handler, of event handler type
error
MUST
be supported by all objects implementing the RTCDataChannel interface.
attribute EventHandler onclose
This event handler, of event handler type
close
MUST
be supported by all objects implementing the RTCDataChannel interface.
attribute EventHandler onmessage
This event handler, of event handler event type
message
MUST
be fired to
allow a developer's JavaScript to receive data from a remote peer.
Event Argument
Description
Object data
The received remote data.
void send (DOMString data)
Run the steps described by the
send()
algorithm with argument type
string
object.
void send (Blob data)
Run the steps described by the
send()
algorithm with argument type
Blob
object.
void send (ArrayBuffer data)
Run the steps described by the
send()
algorithm with argument type
ArrayBuffer
object.
void send (ArrayBufferView data)
Run the steps described by the
send()
algorithm with argument type
ArrayBufferView
object.
Interface Definition
enum RTCDataChannelState
connecting
The user agent is attempting to establish the underlying data transport.
This is the initial state of an
RTCDataChannel
object.
open
The underlying data transport is established and communication is possible.
This is the initial state of an
RTCDataChannel
object dispatched as a
part of an RTCDataChannelEvent.
closing
The procedure to close down the underlying data transport has started.
closed
The underlying data transport has been closed or could not be established.
dictionary RTCDataChannelParameters
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.
DOMString label=""
The
label
attribute represents a label that can be used to distinguish this
RTCDataChannel
object from other
RTCDataChannel
objects.
The attribute
MUST
return the value to which it was set when the
RTCDataChannel
object was constructed.
For an SCTP data channel, the label is carried in the DATA_CHANNEL_OPEN message defined in
[[!DATA-PROT]] Section 5.1.
boolean ordered=true
The
ordered
attribute returns true if the
RTCDataChannel
is ordered, and
false if out of order delivery is allowed. Default is true.
The attribute
MUST
return the value to which it was set when the
RTCDataChannel
was constructed.
unsigned short maxPacketLifetime
The
maxPacketLifetime
attribute represents the length of the time window (in milliseconds) during which
retransmissions may occur in unreliable mode, or null if unset.
The attribute
MUST
return the value to which it was set when the
RTCDataChannel
was constructed.
unsigned short maxRetransmits
The
maxRetransmits
attribute returns the maximum number of
retransmissions that are attempted in unreliable mode, or null if unset.
The attribute
MUST
be initialized to null by default and
MUST
return the
value to which it was set when the
RTCDataChannel
was constructed.
DOMString protocol=""
The name of the sub-protocol used with this
RTCDataChannel
if any,
or the empty string otherwise (in which case the protocol is unspecified).
The attribute
MUST
return the value to which it was set when the
RTCDataChannel
was constucted. Sub-protocols are registered in the
'Websocket Subprotocol Name Registry' created in [[RFC6455]] Section 11.5.
boolean negotiated=false
The
negotiated
attribute returns true if this
RTCDataChannel
was negotiated by the application, or false otherwise. The attribute
MUST
be initialized to
false
by default and
MUST
return the value to which it
was set when the
RTCDataChannel
was constructed.
If set to true, the application developer
MUST
signal to the remote peer to
construct an
RTCDataChannel
object with the same id for the data channel
to be open.
If set to false, the remote party will receive an ondatachannel event with
a system constructed
RTCDataChannel
object.
unsigned short id
The id attribute returns the id for this
RTCDataChannel
, or null if unset.
The id was either assigned by the user agent at channel creation time or was selected by the script.
For SCTP, the id represents a stream identifier, as discussed in [[!DATA]] Section 6.5.
The attribute
MUST
return the value to which it was set when the
RTCDataChannel
was constructed.
The RTCSctpTransport Object
The
RTCSctpTransport
includes information relating to Stream Control Transmission Protocol (SCTP) transport.
Overview
An
RTCSctpTransport
inherits from an
RTCDataTransport
object, which is associated to an
RTCDataChannel
object.
Operation
An
RTCSctpTransport
is constructed from an
RTCDtlsTransport
object.
Interface Definition
readonly attribute
RTCDtlsTransport
transport
The
RTCDtlsTransport
instance the
RTCSctpTransport
object is sending over.
static
RTCSctpCapabilities
getCapabilities()
Retrieves the
RTCSctpCapabilities
of the
RTCSctpTransport
instance.
void start(
RTCSctpCapabilities
remoteCaps)
void stop()
Stops the
RTCSctpTransport
instance.
attribute EventHandler ondatachannel
The
ondatachannel
event handler, of type
datachannel
MUST
be
supported by all objects implementing the
RTCSctpTransport
interface.
If the remote peers sets
RTCDataChannelParameters
negotiated
to false,
then the event will fire indicating a new
RTCDataChannel
object has been
constructed to connect with the
RTCDataChannel
constructed by the remote peer.
dictionary RTCSctpCapabilities
unsigned short maxMessageSize
Maximum message size.
RTCDataChannelEvent
The
datachannel
event
uses the
RTCDataChannelEvent
interface.
Firing a datachannel event named
with a
RTCDataChannel
channel
means that an event with the name
, which
does not bubble (except where otherwise stated) and is not cancelable
(except where otherwise stated), and which uses the
RTCDataChannelEvent
interface with the
channel
attribute set to
channel
, MUST be created and dispatched at the given
target.
readonly attribute RTCDataChannel channel
The
channel
attribute
represents the
RTCDataChannel
object associated
with the event.
RTCDataChannel channel
TODO
Example
function initiate(signaller) {
var dtls = ...; // See ICE/DTLS example.
var sctp = new RTCSctpTransport(dtls);
var parameters = ...; // Construct RTCDataChannelParameters object

signaller.sendInitiate({
// ... include ICE/DTLS info from other example.
"sctpCapabilities": RTCSctpTransport.getCapabilities()
}, function(remote) {
sctp.start(remote.sctpCapabilities);
});

var channel = new RTCDataChannel (sctp, parameters);
channel.send("foo");

function accept(signaller, remote) {
var dtls = ...; // See ICE/DTLS example.
signaller.sendAccept({
// ... include ICE/DTLS info from other example.
"sctpCapabilities": RTCSctpTransport.getCapabilities()
});

var sctp = new RTCSctpTransport(dtls);
sctp.start(remote.sctpCapabilties);

// Assume in-band signalling. We could also easily add
// RTCDataChannelParameters into the out-of-band signalling
// And construct the data channel with with negotiated: true.

sctp.ondatachannel = function(channel) {
channel.onmessage = function(message) {
if (message == "foo") {
channel.send("bar");
Statistics API
The Statistics API enables retrieval of statistics relating to
RTCRtpSender
RTCRtpReceiver
RTCDtlsTransport
RTCIceTransport
and
RTCSctpTransport
objects.
This section of the ORTC API specification depends on the WebRTC 1.0 Statistics API, and needs to be synchronized once it is updated.
Promise getStats()
Gathers stats for the given object
and reports the result asynchronously.
When the
getStats()
method is
invoked, the user agent MUST queue a task to run the following
steps:
If the object's
RTCRtpParameters.RTCRtpEncodingParameters.active
state is
false
, throw an
InvalidStateError
exception.
Return, but continue the following steps in the
background.
Start gathering the stats.
When the relevant stats have been gathered, return a
new
RTCStatsReport
object, representing the
gathered stats.
RTCStatsReport Object
The
getStats()
method delivers a successful result in the form of a
RTCStatsReport
object. A
RTCStatsReport
object represents a map between
strings, identifying the inspected objects (
RTCStats.id
), and their corresponding
RTCStats
objects.
An
RTCStatsReport
may be composed of several
RTCStats
objects, each reporting stats for one
underlying object.
One achieves the total for the object by summing over all stats of a
certain type; for instance, if an
RTCRtpSender
object is sending
RTP streams involving multiple SSRCs over the network, the
RTCStatsReport
may contain one
RTCStats
object per SSRC (which can be distinguished by the value of the "ssrc"
stats attribute).
getter RTCStats (DOMString id)
Getter to retrieve the
RTCStats
objects that
this stats report is composed of.
The set of supported property names [[!WEBIDL]] is defined as the
ids of all the
RTCStats
objects that has been
generated for this stats report. The order of the property names is
left to the user agent.
RTCStats Dictionary
An
RTCStats
dictionary represents the stats
gathered by inspecting a specific 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.
OPEN ISSUE: Need to define an IANA registry for this and populate with
pointers to existing things such as the RTCP statistics.
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 a
RTCStats
object.
DOMHiResTimeStamp timestamp
The
timestamp
of type
DOMHiResTimeStamp
[[!HIGHRES-TIME]], associated
with this object. The time is relative to the UNIX epoch (Jan 1,
1970, UTC).
RTCStatsType type
The type of this object.
The
type
attribute
MUST be initialized to the name of the most specific type this
RTCStats
dictionary represents.
DOMString id
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. User
agents are free to pick any format for the id as long as it meets the
requirements above.
Consider naming id something that indicates that the id refers to
the underlying object that was inspected to produce the stats,
instead of being an id for the JavaScript object. Suggestions:
statsObjectId, reporterId, srcId.
enum RTCStatsType
inboundrtp
Inbound RTP. Relevant to
RTCRtpReceiver
objects.
outboundrtp
Outbound RTP. Relevant to
RTCRtpSender
objects.
session
track
transport
Transport statistics. Relevant to
RTCDtlsTransport
objects.
candidatepair
ICE candidate pair statistics. Relevant to
RTCIceTransport
objects.
localcandidate
ICE local candidate statistics. Relevant to
RTCIceTransport
objects.
remotecandidate
ICE remote candidate statistics. Relevant to
RTCIceTransport
objects.
Derived Stats Dictionaries
RTCRtpStreamStats
DOMString ssrc
...
DOMString remoteId
The
remoteId
can be used to look up the corresponding
RTCStats
object that represents stats reported by
the other peer.
boolean isRemote = false
DOMString mediaTrackId
DOMString transportId
DOMString codecId
unsigned long firCount
Count of FIR packets, defined in [[!RFC5104]] Section 4.3.1.
We are not counting the FIR defined in RFC 2032 Section 5.2.1, which was deprecated in [[RFC4587]].
unsigned long pliCount
Count of PLI packets, defined in [[!RFC4585]] Section 6.3.1.
unsigned long nackCount
Count of NACK packets, defined in [[!RFC4585]] Section 6.2.1.
unsigned long sliCount
Count of SLI packets, defined in [[!RFC4585]] Section 6.3.2.
RTCInboundRTPStreamStats
RTCInboundRTPStreamStats
are relevant to
RTCRtpReceiver
objects.
unsigned long packetsReceived
Packets received.
unsigned long long bytesReceived
Bytes received.
unsigned long packetsLost
Packets lost.
double jitter
Jitter, as calculated in [[!RFC3550]] Section 6.4.1, but given in seconds.
RTCOutboundRTPStreamStats
RTCOutboundRTPStreamStats
are relevant to
RTCRtpSender
objects.
unsigned long packetsSent
Packets sent.
unsigned long long bytesSent
Bytes sent.
double targetBitrate
Presently configured bitrate target of this SSRC, in bits per second.
Typically this is a configuration parameter of the codec's encoder.
double roundTripTime
Estimated round trip time (seconds) based on the RTCP timestamp, as described
in [[!RFC3550]] Section 6.4.1.
RTCMediaStreamTrackStats
RTCMediaStreamTrackStats
are relevant to
MediaStreamTrack
s.
DOMString trackIdentifier
track.id property
boolean remoteSource
sequence ssrcIds
SSRCs.
unsigned long frameWidth
unsigned long frameHeight
double framesPerSecond
The nominal FPS value.
unsigned long framesSent
unsigned long framesReceived
Only makes sense for remoteSource=true.
unsigned long framesDecoded
unsigned long framesDropped
Same definition as droppedVideoFrames in media-source VideoPlaybackQuality
unsigned long framesCorrupted
double audioLevel
Values 0..1, linear, with 1.0 = 0dBov as defined in [[!RFC6464]].
double echoReturnLoss
As defined in G.168 (2012) Section 3.14, in decibels.
double echoReturnLossEnhancement
As above, Section 3.15.
RTCMediaStreamStats
RTCMediaStreamStats
are relevant to
MediaStream
s.
DOMString streamIdentifier
stream.id property
sequence trackIds
This is the id of the stats object, not the track.id
RTCDataChannelStats
RTCDataChannelStats
are relevant to
RTCDataChannel
s.
DOMString label
DOMString protocol
long datachannelid
The 'id' attribute of the
RTCDataChannel
object.
RTCDataChannelState state
The state of the
RTCDataChannel
object.
unsigned long long bytesSent
unsigned long long bytesReceived
unsigned long messagesSent
Number of API 'message' events.
unsigned long messagesReceived
RTCTransportStats
RTCTransportStats
are relevant to
RTCDtlsTransport
objects.
unsigned long long bytesSent
unsigned long long bytesReceived
DOMString rtcpTransportStatsId
If RTP and RTCP are not multiplexed, this is the ID of the transoprt
that gives stats for the RTCP component, and this record has only the RTP component stats.
boolean activeConnection
DOMString selectedCandidatePairId
DOMString localCertificateId
DOMString remoteCertificateId
enum RTCStatsIceCandidatePairState
RTCStatsIceCandidatePairState
provides the state of an ICE candidate pair.
frozen
The candidate pair is frozen.
waiting
The candidate pair is waiting for a connectivity check to go out.
inprogress
A connectivity check is in progress.
failed
The connectivity check has failed.
succeeded
The connectivity check has succeeded.
cancelled
The connectivity check has been cancelled.
dictionary RTCIceCandidatePairStats
RTCStatsIceCandidatePairStats
provides statistics on ICE candidate pairs.
DOMString transportId
DOMString localCandidateId
DOMString remoteCandidateId
RTCStatsIceCandidatePairState state
unsigned long long priority
boolean nominated
boolean writable
boolean readable
unsigned long long bytesSent
unsigned long long bytesReceived
double roundTripTime
double availableOutgoingBitrate
Bits per second, implementation dependent computation.
double availableIncomingBitrate
Bits per second, implementation dependent computation.
dictionary RTCCertificateStats
RTCCertificateStats
provides information on certificates.
DOMString fingerprint
As defined in RFC 4572 section 5 - the fingerprint value only
DOMString fingerprintAlgorithm
For instance 'sha-256'.
DOMString base64Certificate
DER-encoded, base-64 representation of the certificate.
DOMString issuerCertificateId
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:
var mySender = new RTCRtpSender(myTrack);
var myPreviousReport = null;

// ... wait a bit
setTimeout(function () {
mySender.getStats().then(function (report) {
processStats(report);
myPreviousReport = report;
});
}, aBit);

function processStats(currentReport) {
if (myPreviousReport === null) return;
// currentReport + myPreviousReport are an RTCStatsReport interface
// compare the elements from the current report with the baseline
for (var now in currentReport) {
if (now.type != "outbound-rtp")
continue;
// get the corresponding stats from the previous report
base = myPreviousReport[now.id];
// base + now will be of RTCRtpStreamStats dictionary type
if (base) {
remoteNow = currentReport[now.remoteId];
remoteBase = myPreviousReport[base.remoteId];
var packetsSent = now.packetsSent - base.packetsSent;
var packetsReceived = remoteNow.packetsReceived - remoteBase.packetsReceived;
// if fractionLost is > 0.3, we have probably found the culprit
var fractionLost = (packetsSent - packetsReceived) / packetsSent;
Identity
This section of the ORTC API specification depends on the WebRTC 1.0 Identity API,
and needs to be synchronized once it is updated.
Overview
An
RTCIdentity
instance enables authentication of a DTLS transport using a
web-based identity provider (IdP).
The idea is that the initiator acts as the Authenticating Party
(AP) and obtains an identity assertion from the IdP which is then conveyed in signaling.
The responder acts as the Relying Party (RP) and verifies the assertion.
The interaction with the IdP is designed to decouple the browser from any particular
identity provider, so that the browser need only know how to load the IdP's
Javascript (which is deterministic from the IdP's identity), and the generic protocol
for requesting and verifying assertions. The IdP provides whatever logic
is necessary to bridge the generic protocol to the IdP's specific requirements.
Thus, a single browser can support any number of identity protocols, including being
forward compatible with IdPs which did not exist at the time the Identity Provider API was implemented.
The generic protocol details are described in [[!RTCWEB-SECURITY-ARCH]].
This section specifices the procedures required to instantiate the IdP proxy,
request identity assertions, and consume the results.
Operation
RTCIdentity
instance is constructed from an
RTCDtlsTransport
object.
Identity Provider Selection
In order to communicate with the IdP, the browser instantiates an
isolated interpreted context, effectively an invisible IFRAME. The
initial contents of the context are loaded from a URI derived from the
IdP's domain name, as described in [[!RTCWEB-SECURITY-ARCH]].
For purposes of generating assertions, the IdP shall be chosen as
follows:
If the
getIdentityAssertion()
method has been called,
the IdP provided shall be used.
If the
getIdentityAssertion()
method has not been
called, then the browser can use an IdP configured into the
browser.
In order to verify assertions, the IdP domain name and protocol are
taken from the
domain
and
protocol
fields of
the identity assertion.
Instantiating an IdP Proxy
The browser creates an IdP proxy by loading an isolated, invisible
IFRAME with HTML content from the IdP URI. The URI for the IdP is a
well-known URI formed from the
domain
and
protocol
fields, as specified in [[!RTCWEB-SECURITY-ARCH]].
When an IdP proxy is required, the browser performs the following
steps:
An invisible, sandboxed IFRAME is created within the browser
context. The IFRAME
sandbox
attribute is set to
"allow-forms allow-scripts allow-same-origin" to limit the
capabilities available to the IdP. The browser MUST prevent the IdP
proxy from navigating the browsing context to a different location.
The browser MUST prevent the IdP proxy from interacting with the user
(this includes, in particular, popup windows and user dialogs).
Once the IdP proxy is created, the browser creates a
MessageChannel
[[!webmessaging]] within the context of
the IdP proxy and assigns one port from the channel to a variable
named
rtcwebIdentityPort
on the
window
. This
message channel forms the basis of communication between the browser
and the IdP proxy. Since it is an essential security property of the
web sandbox that a page is unable to insert objects into content from
another origin, this ensures that the IdP proxy can trust that
messages originating from
window.rtcwebIdentityPort
are
from
RTCIdentity
and not some other page. This
protection ensures that pages from other origins are unable to
instantiate IdP proxies and obtain identity assertions.
The IdP proxy completes loading and informs the
RTCIdentity
object that it is ready by sending a "READY"
message to the message channel port [[!RTCWEB-SECURITY-ARCH]]. Once
this message is received by the
RTCIdentity
object, the
IdP is considered ready to receive requests to generate or verify
identity assertions.
[TODO: This is not sufficient unless we expect the IdP to protect
this information. Otherwise, the identity information can be copied
from a session with "good" properties to any other session with the same
fingerprint information. Since we want to reuse credentials, that would
be bad.] The identity mechanism MUST provide an indication to the remote
side of whether it requires the stream contents to be
protected. Implementations MUST have an user interface that indicates
the different cases and identity for these.
Requesting Identity Assertions
The identity assertion request process involves the following steps:
The
RTCIdentity
instantiates an IdP proxy as
described in
Identity
Provider Selection section
and waits
for the IdP to signal that it is ready.
The IdP sends a "SIGN" message to the IdP proxy. This message
includes the material the
RTCIdentity
object desires to be bound to the user's
identity.
If the user has been authenticated by the IdP, and the IdP is
willing to generate an identity assertion, the IdP generates an identity
assertion. This step depends entirely on the IdP. The methods by which
an IdP authenticates users or generates assertions is not specified,
though this could involve interacting with the IdP server or other
servers.
The IdP proxy sends a response containing the identity assertion to
the
RTCIdentity
object over the message channel.
The
RTCIdentity
object MAY store the identity assertion.
The format and contents of the messages that are exchanged are
described in detail in [[!RTCWEB-SECURITY-ARCH]].
The IdP proxy can return an "ERROR" response. If an error is
encountered, the
getIdentityAssertion
Promise MUST
be rejected.
The browser SHOULD limit the time that it will allow for this process.
This includes both the loading of
the
IdP proxy
and the
identity assertion generation. Failure to do so potentially causes the
corresponding operation to take an indefinite amount of time. This timer
can be cancelled when the IdP produces a response. The timer running to
completion can be treated as equivalent to an error from the IdP.
NOTE: Where RTP and RTCP are not multiplexed, distinct
RTCRtpIceTransport
RTCRtpDtlsTransport
and
RTCIdentity
objects can be constructed for RTP and RTCP.
However, while it is possible for
getIdentityAssertion()
to be called with different values of
provider
protocol
and
username
for the RTP and RTCP
RTCIdentity
objects, application developers desiring backward compatibility with WebRTC 1.0 are strongly
discouraged from doing so, since this is likely to result in an error.
User Login Procedure
An IdP could respond to a request to generate an identity assertion
with a "LOGINNEEDED" error. This indicates that the site does not have
the necessary information available to it (such as cookies) to authorize
the creation of an identity assertion.
The "LOGINNEEDED" response includes a URL for a page where the
authorization process can be completed. This URL is exposed to the
application through the
loginUrl
attribute
of the
RTCIdentityError
object.
This URL might be to a page where a user is able to enter their (IdP)
username and password, or otherwise provide any information the IdP
needs to authorize a assertion request.
An application can load the login URL in an IFRAME or popup; the
resulting page then provides the user with an opportunity to provide
information necessary to complete the authorization process.
Once the authorization process is complete, the page loaded in the
IFRAME or popup sends a message using
postMessage
[[!webmessaging]] to the page that loaded it (through the
window.opener
attribute for popups, or through
window.parent
for pages loaded in an IFRAME). The message MUST be the
DOMString
"LOGINDONE". This message informs the application
that another attempt at generating an identity assertion is likely to be
successful.
Verifying Identity Assertions
Identity assertion validation happens
when
setIdentityAssertion()
is invoked. The process runs
asynchronously.
The identity assertion validation process involves the following
steps:
The
RTCIdentity
instantiates an IdP proxy as
described in
Identity
Provider Selection section
and waits
for the IdP to signal that it is ready.
The IdP sends a "VERIFY" message to the IdP proxy. This message
includes the assertion which is to be
verified.
The IdP proxy verifies the identity assertion (depending on the
authentication protocol this could involve interacting with the IDP
server).
Once the assertion is verified, the IdP proxy sends a response
containing the verified assertion results to the
RTCIdentity
object over the message channel.
The
RTCIdentity
object validates that the fingerprint
provided by the IdP in the validation response matches the certificate
fingerprint that is, or will be, used for communications. This is done by
waiting for the DTLS connection to be established and checking
that the certificate fingerprint on the connection matches the one
provided by the IdP.
The
RTCIdentity
validates that the domain portion
of the identity matches the domain of the IdP as described in [[!RTCWEB-SECURITY-ARCH]].
The
RTCIdentity
stores the assertion in the
peerIdentity
, and returns an
RTCIdentityAssertion
object
when the Promise from
setIdentityAssertion()
is fulfilled.
The assertion
information to be displayed MUST contain the domain name of the IdP as
provided in the assertion.
The browser MAY display identity information to a user in browser
UI. Any user identity information that is displayed in this fashion
MUST use a mechanism that cannot be spoofed by content.
The IdP might fail to validate the identity assertion by providing an
"ERROR" response to the validation request. Validation can also fail due
to the additional checks performed by the browser. In both cases, the
process terminates and no identity information is exposed to the
application or the user.
The browser MUST cause the Promise of
setIdentityAssertion()
to be rejected if
validation of an identity assertion fails for any reason.
The browser SHOULD limit the time that it will allow for this process.
This includes both the loading of
the
IdP proxy
and the
identity assertion validation. Failure to do so potentially causes the
corresponding operation to take an indefinite amount of time. This timer
can be cancelled when the IdP produces a response. The timer running to
completion can be treated as equivalent to an error from the IdP.
The format and contents of the messages that are exchanged are
described in detail in [[!RTCWEB-SECURITY-ARCH]].
NOTE: Where RTP and RTCP are not multiplexed, it is possible that the assertions for both the RTP and RTCP will be validated,
but that the identities will not be equivalent. For applications requiring backward compatibility with WebRTC 1.0,
this MUST be considered an error. However, if backward compatibility with WebRTC 1.0 is not required the application MAY consider
an alternative, such as ignoring the RTCP identity assertion.
RTCIdentity Interface
The Identity API is described below.
readonly attribute RTCIdentityAssertion? peerIdentity
peerIdentity
contains the peer identity assertion information if an identity
assertion was provided and verified. Once this value is set to a
non-
null
value, it cannot change.
readonly attribute RTCDtlsTransport transport
The
RTCDtlsTransport
to be authenticated.
Promise getIdentityAssertion (DOMString provider, optional DOMString protocol = "default", optional DOMString username)
Sets the identity provider to be used for a given
RTCIdentity
object, and initiates the process of obtaining an identity assertion.
When
getIdentityAssertion()
is invoked, the user agent MUST
run the following steps:
If
transport.state
is
closed
, throw an
InvalidStateError
exception and abort these
steps.
Set the current identity provider values to the triplet
provider
protocol
username
).
If any identity provider value has changed, discard any stored
identity assertion.
Request an
identity assertion
from the IdP.
If the IdP proxy provides an assertion over the message channel,
the Promise is fulfilled, and the assertion is returned (equivalent to
onidentityresult
in the
WebRTC 1.0 API). If the IdP proxy returns an "ERROR" response, the Promise is rejected, and an
RTCIdentityError
object is returned,
(equivalent to
onidpassertionerror
in the WebRTC 1.0 API).
Promise setIdentityAssertion (DOMString assertion)
Validates the identity assertion. If the Promise is fulfilled,
an
RTCIdentityAssertion
is returned.
If the Promise is rejected, an
RTCIdentityError
object is returned, (equivalent to
onidpvalidationerror
in the WebRTC 1.0 API).
dictionary RTCIdentityError
DOMString idp
The domain name of the identity provider that is providing the error response.
DOMString protocol
The IdP protocol that is in use.
DOMString? loginUrl
An IdP that is unable to generate an identity assertion due to a
lack of sufficient user authentication information can provide a URL
to a page where the user can complete authentication.
If the IdP provides this URL, this attribute includes the value provided
by the IdP.
dictionary RTCIdentityAssertion
DOMString idp
A domain name representing the identity provider.
DOMString name
A representation of the verified peer identity conforming to [[RFC5322]].
This identity will have been verified via the
procedures described in [[!RTCWEB-SECURITY-ARCH]].
Example
The identity system is designed so that applications need not take any
special action in order for users to generate and verify identity
assertions; if a user has configured an IdP into their browser, then the
browser will automatically request/generate assertions and the other side
will automatically verify them and display the results. However,
applications may wish to exercise tighter control over the identity
system as shown by the following examples.
This example shows how to configure the identity provider and
protocol, and consume identity assertions.
var iceOptions = ...;
var ice = new RTCIceTransport(iceOptions);
var dtls = new RTCDtlsTransport(ice);
var identity = new RTCIdentity(dtls);
identity.getIdentityAssertion("example.com", "default", "alice@example.com").then(signalAssertion(assertion)
,function (e) {
console.log("Could not obtain an Identity Assertion. idp: ",e.idp,"Protocol: ",e.protocol,"loginUrl: ",e.loginUrl);
});

function signalAssertion(assertion){
mySignalInitiate(
{ "myAssertion": assertion
}, function (response) {
identity.setIdentityAssertion(response.myAssertion).then(function (peerAssertion) {
console.log("Peer identity assertion validated. idp: ",peerAssertion.idp, "name: ", peerAssertion.name);
}, function (e) {
console.log("Could not validate peer assertion. idp: ", e.idp, "Protocol: ",e.protocol);
});
});
Event summary
The following events fire on
RTCDtlsTransport
objects:
Event name
Interface
Fired when...
error
Event
The
RTCDtlsTransport
object has
received a DTLS Alert.
dtlsstatechange
RTCDtlsTransportStateChangedEvent
The
RTCDtlsTransportState
changes.
The following events fire on
RTCIceTransport
objects:
Event name
Interface
Fired when...
icestatechange
RTCIceTransportStateChangedEvent
The
RTCIceTransportState
changes.
icegatheringstatechange
RTCIceGatheringStateChangedEvent
The
RTCIceGatheringState
changes.
icecandidate
RTCIceTransport
A new
RTCIceCandidate
is made available to the script.
error
Event
The
RTCIceTransport
object has
experienced an ICE gathering failure (such as an authentication failure with TURN credentials).
The following event fires on
RTCRtpSender
objects:
Event name
Interface
Fired when...
error
Event
An error has been detected within the
RTCRtpSender
object. This is not used for programmatic exceptions.
The following event fires on
RTCRtpReceiver
objects:
Event name
Interface
Fired when...
error
Event
An error has been detected within the
RTCRtpReceiver
object. This is not used for programmatic exceptions.
The following events fire on
RTCRtpListener
objects:
Event name
Interface
Fired when...
unhandledrtp
RTCRtpUnhandledEvent
The
RTCRtpListener
object has received an
RTP packet that it cannot deliver to an
RTCRtpReceiver
object.
The following events fire on
RTCDTMFSender
objects:
Event name
Interface
Fired when...
tonechange
Event
The
RTCDTMFSender
object has either just
begun playout of a tone (returned as the
tone
attribute) or just ended playout of a tone (returned as an empty
value in the
tone
attribute).
The following events fire on
RTCDataChannel
objects:
Event name
Interface
Fired when...
open
Event
The
RTCDataChannel
object's
underlying data transport
has been established (or re-established).
MessageEvent
Event
A message was successfully received. TODO: Ref where MessageEvent
is defined?
error
Event
TODO.
close
Event
The
RTCDataChannel
object's
underlying data transport
has been closed.
The following events fire on
RTCSctpTransport
objects:
Event name
Interface
Fired when...
datachannel
RTCDataChannelEvent
A new
RTCDataChannel
is dispatched to the script in response to the
other peer creating a channel.
WebRTC 1.0 Compatibility
It is a goal of the ORTC API to provide the functionality of the WebRTC 1.0 API [[WEBRTC10]], as well as to enable the
WebRTC 1.0 API to be implemented on top of the ORTC API, utilizing a Javascript "shim" library. This section
discusses WebRTC 1.0 compatibility issues that have been encountered by ORTC API implementers.
Voice Activity Detection
[[WEBRTC10]] Section 4.2.4 defines the
RTCOfferOptions
dictionary, which includes the
voiceActivityDetection
attribute,
which determines whether Voice Activity Detection (VAD) is enabled within the Offer produced by
createOffer()
The effect of setting
voiceActivityDetection
to
TRUE
is to include the Comfort Noice (CN) codec defined in
[[RFC3389]] within the Offer.
Within ORTC API, equivalent behavior can be obtained by configuring the Comfort Noise codec for use within the
RTCRtpParameters
object,
or configuring a codec with built-in support for Comfort Noise (such as G.729) to enable comfort noise.
Examples
Simple Peer-to-peer Example
This example code provides a basic audio and video session between two browsers.
myCapsToSendParams Example
RTCRtpParameters function myCapsToSendParams (RTCRtpCapabilities sendCaps, RTCRtpCapabilities remoteRecvCaps) {
// Function returning the sender RTCRtpParameters, based on the local sender and remote receiver capabilities.
// The goal is to enable a single stream audio and video call with minimum fuss.
//
// Steps to be followed:
// 1. Determine the RTP features that the receiver and sender have in common.
// 2. Determine the codecs that the sender and receiver have in common.
// 3. Within each common codec, determine the common formats, header extensions and rtcpFeedback mechanisms.
// 4. Determine the payloadType to be used, based on the receiver preferredPayloadType.
// 5. Set RTCRtcpParameters such as mux to their default values.
// 6. Return RTCRtpParameters enablig the jointly supported features and codecs.

RTCRtpParameters function myCapsToRecvParams (RTCRtpCapabilities recvCaps, RTCRtpCapabilities remoteSendCaps) {
// Function returning the receiver RTCRtpParameters, based on the local receiver and remote sender capabilities.
return myCapsToSendParams(remoteSendCaps, recvCaps);
Change Log
This section will be removed before publication.
Changes since 16 June 2014
Added section on WebRTC 1.0 compatibility issues, responding to
Issue 66
Added Identity support, as described in
Issue 78
Reworked getStats method, as described in
Issue 85
Removed ICE restart method described in
Issue 93
Addressed CNAME and synchronization context issues described in
Issue 94
Fixed WebIDL issues noted in
Issue 97
Addressed NITs described in
Issue 99
DTLS transport issues fixed as described in
Issue 100
ICE transport issues fixed as described in
Issue 101
ICE transport controller fixes made as described in
Issue 102
Sender and Receiver object fixes made as described in
Issue 103
Fixed RTCRtpEncodingParameter default issues described in
Issue 104
Fixed 'Big Picture' issues descibed in
Issue 105
Fixed RTCRtpParameter default issues described in
Issue 106
Added a multi-stream capability, as noted in
Issue 108
Removed quality scalability capabilities and parameters, as described in
Issue 109
Added scalability examples as requested in
Issue 110
Addressed WebRTC 1.0 Data Channel compatibility issue described in
Issue 111
Removed header extensions from
RTCRtpCodecParameters
as described in
Issue 113
Addressed RTP/RTCP non-mux issues with IdP as described in
Issue 114
Added getParameter methods to
RTCRtpSender
and
RTCRtpReceiver
objects, as described in
Issue 116
Added layering diagrams as requested in
Issue 117
Added a typedef for payload type, as described in
Issue 118
Moved
onerror
from the
RTCIceTransport
object to the
RTCIceListener
object as described in
Issue 121
Added explanation of Voice Activity Detection (VAD), responding to
Issue 129
Clarified the meaning of maxTemporalLayers and maxSpatialLayers, as noted in
Issue 130
Added RFC 6051 to the list of header extensions and removed RFC 5450, as noted in
Issue 131
Addressed ICE terminology issues, as described in
Issue 132
Separated references into Normative and Informative, as noted in
Issue 133
Changes since 14 May 2014
Added support for non-multiplexed RTP/RTCP and ICE freezing, as described in
Issue 57
Added support for getRemoteCertificates(), as described in
Issue 67
Removed filterParameters and createParameters functions, as described in
Issue 80
Partially addressed capabilities issues, as described in
Issue 84
Addressed WebIDL type issues described in
Issue 88
Addressed Overview section issues described in
Issue 91
Addressed readonly attribute issues described in
Issue 92
Added ICE restart method to address the issue described in
Issue 93
Added onerror eventhandler to sender and receiver objects as described in
Issue 95
Changes since 29 April 2014
ICE restart explanation added, as described in
Issue 59
Fixes for error handling, as described in
Issue 75
Fixes for miscellaneous NITs, as described in
Issue 76
Enable retrieval of the SSRC to be used by RTCP, as described in
Issue 77
Support for retrieval of audio and video capabilities, as described in
Issue 81
getStats interface updated, as described in
Issue 82
Partially addressed SVC issues described in
Issue 83
Partially addressed statistics update issues described in
Issue 85
Changes since 12 April 2014
Fixes for error handling, as described in
Issue 26
Support for contributing sources removed (re-classified as a 1.2 feature), as described in
Issue 27
Cleanup of DataChannel construction, as described in
Issue 60
Separate proposal on simulcast/layering, as described in
Issue 61
Separate proposal on quality, as described in
Issue 62
Fix for TCP candidate type, as described in
Issue 63
Fix to the fingerprint attribute, as described in
Issue 64
Fix to RTCRtpFeatures, as described in
Issue 65
Support for retrieval of remote certificates, as described in
Issue 67
Support for ICE error handling, described in
Issue 68
Support for Data Channel send rate control, as described in
Issue 69
Support for capabilities and settings, as described in
Issue 70
Removal of duplicate RTCIceListener functionality, as described in
Issue 71
ICE gathering state added, as described in
Issue 72
Removed ICE role from the ICE transport constructor, as described in
Issue 73
Changes since 13 February 2014
Support for contributing source information added, as described in
Issue 27
Support for control of quality, resolution, framerate and layering added, as described in
Issue 31
RTCRtpListener object added and figure in Section 1 updated, as described in
Issue 32
More complete support for RTP and Codec Parameters added, as described in
Issue 33
Data Channel transport problem fixed, as described in
Issue 34
Various NITs fixed, as described in
Issue 37
Section 2.2 and 2.3 issues fixed, as described in
Issue 38
Default values of some dictionary attributes added, to partially address the issue described in
Issue 39
Support for ICE TCP added, as described in
Issue 41
Fixed issue with sequences as attributes, as described in
Issue 43
Fix for issues with onlocalcandidate, as described in
Issue 44
Initial stab at a Stats API, as requested in
Issue 46
Added support for ICE gather policy, as described in
Issue 47
Changes since 07 November 2013
RTCTrack split into RTCRtpSender and RTCRtpReceiver objects, as proposed on
06 January 2014.
RTCConnection split into RTCIceTransport and RTCDtlsTransport objects, as proposed on
09 January 2014.
RTCSctpTransport object added, as described in
Issue 25
RTCRtpHeaderExtensionParameters added, as described in
Issue 28
RTCIceListener added, in order to support parallel forking, as described in
Issue 29
DTMF support added, as described in
Issue 30