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 current WebRTC 1.0 APIs, ORTC does not mandate a media signaling protocol or format. As a result, ORTC does not utilize Session Description Protocol (SDP) within its APIs, nor does it mandate support for the Offer/Answer state machine. Instead, ORTC focuses on "connections" and "tracks" being carried over those connections.

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), RTCIceListener (Section 4), RTCRtpSender (Section 5), RTCRtpReceiver (Section 6), RTCRtpListener (Section 7), RTCDtmfSender (Section 9), RTCDataChannel (Section 10), and RTCSctpTransport (Section 11). RTP dictionaries are described in Section 8, and the Statistics API is described in Section 12.

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.

The non-normative ORTC Big Picture Diagram

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, MediaStreamTrack, Constraints, and Consumer are defined in [[!GETUSERMEDIA]].

The terms single-session transmission (SST) and multi-session transmission (MST) are defined in [[!RFC6190]]. While this specification supports SST but not MST, multiple streams may be utilized within a single RTP session.

The RTCDtlsTransport Object

The RTCDtlsTransport includes information relating to Datagram Transport Layer Security (DTLS) transport.

Overview

An RTCDtlsTransport instance is associated to an RTCRtpSender or an RTCRtpReceiver.

Operation

A RTCDtlsTransport instance is optionally constructed from an RTCIceTransport object or an RTCDtlsTransport is automatically constructed.

Interface Definition

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<ArrayBuffer> 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.

void stop()

Stops and closes the DTLS transport object.

attribute EventHandler? onstatechange

This event handler, of type trust, with the name statechanged, uses the RTCDtlsStateChangedEvent interface. It MUST be supported by all objects implementing the RTCDtlsTransport interface. This event is fired when the RTCDtlsTransportState changes, and provides the new RTCDtlsTransportState that caused the event.

attribute EventHandler? onerror

This event handler, of type error, MUST be supported by all objects implementing the RTCDtlsTransport interface.

The RTCDtlsParameters Object

The RTCDtlsParameters object includes information relating to DTLS configuration.

RTCDtlsRole role="auto"

The DTLS role, with a default of auto.

sequence<RTCDtlsFingerprint> 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

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 provides information about 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

closed state

RTCDtlsTransportStateChangedEvent

The statechanged event of the RTCDtlsTransport object uses the RTCDtlsTransportStateChangedEvent interface.

Firing an RTCDtlsTransportStateChangedEvent event named e with an RTCDtlsTransportState state means that an event with the name e, 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.

Constructor(DOMString type, RTCDtlsTransportStateChangedEventInit eventInitDict)
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

RTCIceRole contains the current role of the ICE transport.

readonly attribute RTCIceTransportState state

The current state of the ICE transport.

readonly attribute RTCIceGatheringState iceGatheringState

The current state of ICE candidate gathering.

sequence<RTCIceCandidate> 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<RTCIceCandidate> getRemoteCandidates()

Retrieve the sequence of candidates associated with the remote RTCIceTransport.

void gather(RTCIceGatherPolicy gatherPolicy)

Start gathering RTCIceCandidate objects, based on the gather policy (set on the local system, not negotiated).

void start(RTCIceParameters remoteParameters, optional RTCIceRole role)

Starts candidate connectivity checks and attempts to connect to the remote RTCIceTransport.

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.

void addRemoteCandidate(RTCIceCandidate remoteCandidate)

Add remote candidate associated with remote RTCIceTransport.

void setRemoteCandidates(sequence<RTCIceCandidate> 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? onstatechange

This event handler, of type trust, with the name statechanged, uses the RTCIceTransportStateChangedEvent interface. It MUST be supported by all objects implementing the RTCIceTransport interface. This event is fired when the RTCIceTransportState changes, and provides the new RTCIceTransportState that caused the event.

attribute EventHandler? ongatheringstatechange

This event handler, of type trust, with the name gatherstatechanged, uses the RTCIceGatheringStateChangedEvent interface. It MUST be supported by all objects implementing the RTCIceTransport interface. This event is fired when the RTCIceGatheringState changes, and provides the new RTCIceGatheringState that caused the event.

attribute EventHandler? onerror

This event handler, of type error, MUST be supported by all objects implementing the RTCIceTransport interface.

enum RTCIceGatheringState

new

No networking has occurred yet.

gathering

The ICE engine is in the process of gathering candidates.

complete

The ICE engine 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 gatherstatechanged event of the RTCIceTransport object uses the RTCIceGatheringStateChangedEvent interface.

Firing an RTCIceGatheringStateChangedEvent event named e with an RTCIceGatheringState state means that an event with the name e, 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.

Constructor(DOMString type, RTCIceGatheringStateChangedEventInit eventInitDict)
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

controlling

controlling state

controlled

controlled state

enum RTCIceGatherPolicy

all

Gather all ICE candidate types.

nohost

Gather all ICE candidate types except for host candidates.

relayonly

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

new

new state

checking

checking state

connected

connected state

completed

completed state

disconnected

disconnected state

closed

closed state

The non-normative ICE state transitions are:

The non-normative ICE State Transition Diagram

RTCIceTransportStateChangedEvent

The statechanged event of the RTCIceTransport object uses the RTCIceTransportStateChangedEvent interface.

Firing an RTCIceTransportStateChangedEvent event named e with an RTCIceTransportState state means that an event with the name e, 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.

Constructor(DOMString type, RTCIceTransportStateChangedEventInit eventInitDict)
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<RTCIceServer>? 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<DOMString>) 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 RTCIceTransports.

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=null

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=null

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 e with an RTCIceCandidate candidate means that an event with the name e, 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.

Constructor(DOMString type, RTCIceTransportEventInit eventInitDict)
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.

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

 signaller.sendInitiate({
   "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) {
   signaller.sendLocalCandidate(candidate);
 }

 signaller.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.
//
function accept(signaller, 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) {
   signaller.sendLocalCandidate(candidate);
 }

 signaller.onRemoteCandidate = function(candidate) {
   ice.addRemoteCandidate(candidate);
 }

 signaller.sendAccept({
   "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 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

attribute RTCIceOptions? options

The RTCIceOptions instance.

Example



var iceOptions = ...;
var iceListener = new RTCIceListener(iceOptions);
var iceBase = new RTCIceTransport(iceListener);
sendInitiate(iceBase.getLocalParameters(), function(response) {
  // We may get N responses
  var ice = new RTCIceTransport(iceListener);
  ice.start(response.iceParameters, RTCIceRole.controlling);
  // ... setup DTLS, RTP, SCTP, etc.
});

iceBase.onlocalcandidate = sendLocalCandidate;

    

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

A RTCRtpSender instance is constructed from an MediaStreamTrack object and associated to an RTCDtlsTransport.

Interface Definition

attribute MediaStreamTrack track

The associated MediaStreamTrack instance.

attribute RTCDtlsTransport transport

The associated RTCDtlsTransport instance.

static RTCRtpCapabilities getCapabilities(optional DOMString kind)

Obtain the sender capabilities, based on kind. If kind is omitted, then all capabilities are returned.

static RTCRtpParameters createParameters(MediaStreamTrack track, optional RTCRtpCapabilities remoteCapabilities)

Create parameters based on the MediaStreamTrack and the remote capabilities specified in RTCRtpCapabilities.

static RTCRtpParameters filterParameters(RTCRtpParameters parameters, optional RTCRtpCapabilities remoteCapabilities)

Filter parameters based on the remote RTCRtpCapabilities. Calling RTCRtpSender.filterParameters(RTCRtpSender.createParameters(track), remoteCapabilities) returns the same result as calling RTCRtpSender.createParameters(track, remoteCapabilities).

void send(RTCRtpParameters parameters)

Media is controlled by the given "parameters". The sender starts sending when send() is called and stops sending when stop() is called.

void stop()

Stops sending the track on the wire. Stop is final like MediaStreamTrack

Example


    

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

A RTCRtpReceiver instance is constructed from an RTCDtlsTransport object.

Interface Definition

readonly attribute MediaStreamTrack? track

The associated MediaStreamTrack instance.

attribute RTCDtlsTransport transport

The associated RTCDtlsTransport instance.

static RTCRtpCapabilities getCapabilities(optional DOMString kind)

Obtain the receiver capabilities, based on kind. If kind is omitted, then all capabilities are returned.

static RTCRtpParameters createParameters (DOMString kind, optional RTCRtpCapabilities remoteCapabilities)
Create parameters based on kind and the remote capabilities specified in RTCRtpCapabilities.
DOMString kind
Specifies kind, either "audio" or "video".
optional RTCRtpCapabilities remoteCapabilities
static RTCRtpParameters filterParameters(RTCRtpParameters parameters, optional RTCRtpCapabilities remoteCapabilities)

Filter parameters based on the remote RTCRtpCapabilities. Calling RTCRtpReceiver.filterParameters(RTCRtpReceiver.createParameters(kind), remoteCapabilities) returns the same result as calling RTCRtpReceiver.createParameters(kind, remoteCapabilities).

void receive(RTCRtpParameters parameters)

Media is controlled by the given "parameters". The receiver starts receiving when the receive() is called and stopped when the stop() is called.

void stop()

Stops receiving the track on the wire. Stop is final like MediaStreamTrack

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.
function initiate(signaller, 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);

  var sendAudioParams = RTCRtpSender.createParameters(audioTrack);
  var sendVideoParams = RTCRtpSender.createParameters(videoTrack);
  signaller.offerTracks({
    // The initiator offers parameters it wants to send with,
    // and the capabilities it has for receiving.
    "rtpAudioCaps": RTCRtpReceiver.getCapabilities("audio"), 
    "rtpVideoCaps": RTCRtpReceiver.getCapabilities("video"),
    "audio": sendAudioParams,
    "video": sendVideoParams
  }, function(answer) {
    // The responder answers with parameters it wants to send with
    // and the capabilities it has for receiving.
    var audioSendParams = RTCRtpSender.filterParameters(
       sendAudioParams, answer.rtpAudioCaps);
    var videoSendParams = RTCRtpSender.filterParameters(
       sendVideoParams, answer.rtpVideoCaps);
    var audioRecvParams = RTCRtpReceiver.filterParameters(
       answer.audio);
    var videoRecvParams = RTCRtpReceiver.filterParameters(
       answer.video);

    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.
function accept(
  signaller, 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);

  var audioSendParams = RTCRtpSender.createParameters(
    audioTrack, remote.rtpAudioCaps);
  var videoSendParams = RTCRtpSender.createParameters(
    videoTrack, remote.rtpVideoCaps);
  var audioRecvParams = RTCRtpReceiver.filterParameters(
     remote.audio);
  var videoRecvParams = RTCRtpReceiver.filterParameters(
     remote.video);

  audioSender.send(audioSendParams);
  videoSender.send(videoSendParams)
  audioReceiver.receive(audioRecvParams);
  videoReceiver.receive(videoRecvParams);

  signaller.answerTracks({
    "rtpAudioCaps": RTCRtpReceiver.getCapabilities("audio"),
    "rtpVideoCaps": RTCRtpReceiver.getCapabilities("video"),
    "audio": audioSendParams,
    "video": videoSendParams
  });

  // 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 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 uses the RTCRtpUnhandledEvent interface.

Firing an unhandledrtp event named e with an RTCRtpUnhandled stream means that an event with the name e, 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.

Constructor(DOMString type, RTCRtpUnhandledEventInit eventInitDict)
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

unsigned int ssrc

The SSRC in the RTP stream triggering the unhandledrtp event.

unsigned byte 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

sequence<RTCRtpCodec> audioCodecs

Supported audio codecs.

sequence<RTCRtpCodec> videoCodecs

Supported video codecs.

sequence<DOMString> headerExtensions

URIs of supported RTP header extensions.

Capabilities features

Features supported by the RTP engine.

Capabilities rtcpFeedback

Supported RTCP capabilities, such as "nack" from [[!RFC4585]].

dictionary RTCRtpCodec

DOMString name=""

The MIME media type, if set, empty string otherwise.

unsigned int? hzRate=null

Codec clock rate expressed in Hertz, null if unset.

unsigned byte preferredPayloadType

Added to make it possible for the sender and receiver to pick a matching payload type when creating sender and receiver parameters.

unsigned int? numChannels=1

The number of channels supported (e.g. stereo); one by default. For video, this will be null.

Capabilities formats

Codec-specific capabilities available for signaling.

dictionary RTCRtpParameters

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<RTCRtpCodecParameters> codecs

The codecs to send or receive (could include RTX and CN as well).

sequence<RTCRtpHeaderExtensionParameters> headerExtensions

The RTP header extensions to send or receive.

sequence<RTCRtpEncodingParameters> encodings

The "encodings" or "layers" to be used for things like simulcast, Scalable Video Coding, RTX, FEC, etc.

unsigned int rtcpSsrc

The SSRC used by RTCP (e.g. in Receiver Reports)t. 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]].

dictionary RTCRtpCodecParameters

unsigned byte payloadType

The value that goes in the RTP Payload Type Field [[!RFC3550]]. The payloadType must always be provided both within RTCRtpSender and RTCRtpReceiver objects.

RTCRtpCodec codec

The codec to be used for sending or receiving.

Settings formats

Codec settings that control what is sent or received. For example, with Opus [[!RFC6716]], stereo=1.

Settings rtcpFeedback

RTCP feedback settings, separated out because they are so different.

dictionary RTCRtpEncodingParameters

unsigned int? ssrc=null

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.

DOMString codecName=""

For per-encoding codec specifications, give the codec name here. If the empty string, the browser will choose.

RTCRtpFecParameters? fec=null

Specifies the FEC mechanism if used, null if unset.

RTCRtpRtxParameters? rtx=null

Specifies the RTX mechanism if used, null if unset.

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=null

Ramp up resolution/quality/framerate until this bitrate, null if unset. 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 scale=null

Do this scale of the input resolution, or die trying. 1.0 = full resolution. Default is unconstrained (null). For scalable video coding, scale refers to the aggregate scale achieved by this layer when combined with all dependent layers.

double framerate=null

Fraction of the framerate to be encoded. 1.0 = full framerate. Default is unconstrained (null). For scalable video coding, framerate refers to the aggregate framerate achieved by this layer when combined with all dependent layers.

double quality=null

Fraction of the quality to be encoded. 1.0 = full quality. Default is unconstrained (null). For scalable video coding, quality refers to the aggregate quality 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<DOMString> 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


//Send a thumbnail along with regular size
var encodings1 = [{ ssrc: 1,  priority: 1.0 }] 

// Control the resolution and framerate with a different track and RtpSender. 
var encodings2 = [{ 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 }]; 

Example of 3-layer spatial scalability encoding
var encodings =[{
  layerId: "0",
  scale: 0.25
}, {
  layerId: "1",
  layerDependencies: ["0"]
  scale: 0.5
}, {
  layerId: "2",
  layerDependencies: ["0", "1"]
  scale: 1.0
}]

Example of 3-layer spatial scalability with all but bottom layer disabled
var encodings1 =[{
  layerId: "0",
  scale: 0.25
}, {
  layerId: "1",
  layerDependencies: ["0"],
  scale: 0.5,
  active: false
}, {
  layerId: "2",
  layerDependencies: ["0", "1"],
  scale: 1.0,
  active: false
}];

Example of 3-layer spatial simulcast
var encodings =[{
  layerId: "0",
  scale: 0.25
}, {
  layerId: "1",
  scale: 0.5
}, {
  layerId: "2",
  scale: 1.0
}]

Example of 3-layer spatial simulcast with all but bottom layer disabled
var encodings1 =[{
  layerId: "0",
  scale: 0.25
}, {
  layerId: "1",
  scale: 0.5,
  active: false
}, {
  layerId: "2",
  scale: 1.0,
  active: false
}];

Example of 2-layer spatial simulcast combined with 2-layer temporal scalability
var encodings =[{
  layerId: "halfScaleBase",
  scale: 0.5,
  framerate: 0.5
}, {
  layerId: "fullScaleBase",
  scale: 1.0,
  framerate: 0.5
}, {
  layerId: "temporalEnhancemenToHalfScaleBase",
  layerDependencies: ["halfScaleBase"],
  scale: 0.5,
  framerate: 1.0
}, {
  layerId: "temporalEnhancementToFullScaleBase",
  layerDependencies: ["fullScaleBase"],
  scale: 1.0,
  framerate:1.0
}]

dictionary RTCRtpFecParameters

unsigned int? ssrc=null

The SSRC to use for FEC, null if unset. If null in an RTCRtpSender object, the browser will choose.

DOMString mechanism

The Forward Error Correction (FEC) mechanism to use.

dictionary RTCRtpRtxParameters

unsigned int? ssrc=null

The SSRC to use for RTX, null if unset. If null in an RTCRtpSender object, the browser will choose.

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.

Properties

PropertyValueSet

MultiPropertyValueSet

PropertyValueDoubleRange

double max

The maximum legal value of this property.

double min

The minimum legal value of this property.

PropertyValueLongRange

long max

The maximum legal value of this property.

long min

The minimum legal value of this property.

ConstraintValue

ConstraintValues

Capabilities

Capabilities are a dictionary containing one or more key-value pairs, where each key MUST be a property defined in the registry, and each value MUST be a subset of the set of values defined for that property in the registry. The exact syntax of the value expression depends on the type of the property but is of type ConstraintValues. The Capabilities dictionary specifies the subset of the properties and values from the registry that the UA supports. Note that a UA may support only a subset of the properties that are defined in the registry, and may support a subset of the set values for those properties that it does support. Note that Capabilities are returned from the UA to the application, and cannot be specified by the application. However, the application can set the Parameters. An example of a Capabilities dictionary is shown below. This example is not very realistic in that a browser would actually be required to support more settings that just these.

Examples


Settings

A Settings is a dictionary containing one or more key-value pairs. It MUST contain each key returned in Capabilities. There MUST be a single value for each key and the value MUST be a member of the set defined for that property by Capabilities. The exact syntax of the value expression depends on the type of the property. It will be a DOMString[] for properties of type MultiPropertyValueSet. It will be a DOMString[] for properties of type PropertyValueSet, it will be a long for properties of type PropertyValueLongRange , it will be a double for properties of type PropertyValueDoubleRange. Thus the Settings dictionary contains the actual values that the UA has chosen for the object's Capabilities. An example of a Settings dictionary is shown below. This example is not very realistic in that a browser would actually be required to support more settings that just these.

Examples


The RTCDtmfSender Object

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 e with a DOMString tone means that an event with the name e, 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.

Constructor(DOMString type, RTCDTMFToneChangeEventInit eventInitDict)
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 type open, MUST be supported by all objects implementing the RTCDataChannel interface.

attribute EventHandler onerror

This event handler, of type error, MUST be supported by all objects implementing the RTCDataChannel interface.

attribute EventHandler onclose

This event handler, of 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.
Promise send (DOMString data)

Run the steps described by the send() algorithm with argument type string object.

Promise send (Blob data)

Run the steps described by the send() algorithm with argument type Blob object.

Promise send (ArrayBuffer data)

Run the steps described by the send() algorithm with argument type ArrayBuffer object.

Promise 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=null

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=null

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=null

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

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 int maxMessageSize=null

Maximum message size or null if unset.

RTCDataChannelEvent

The datachannel event uses the RTCDataChannelEvent interface.

Firing a datachannel event named e with a RTCDataChannel channel means that an event with the name e, 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.

Constructor(DOMString type, RTCDataChannelEventInit eventInitDict)
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.

void getStats(RTCStatsObject object, RTCStatsCallback successCallback, RTCErrorCallback failureCallback)

Gathers stats for the given object and reports the result asynchronously. TODO: Rework this as a Promise.

When the getStats() method is invoked, the user agent MUST queue a task to run the following steps:

  1. If the object's RTCRtpEncodingParameters.active state is false, throw an InvalidStateError exception.

  2. Return, but continue the following steps in the background.

  3. Start gathering the stats.

  4. When the relevant stats have been gathered, queue a task to invoke the success callback (the method's second argument) with a new RTCStatsReport object, representing the gathered stats, as its argument.

RTCStatsCallback

RTCStatsReport report

A RTCStatsReport representing the gathered stats.

RTCErrorCallback

DOMError error

An error object encapsulating information about what went wrong.

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
R

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.
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 default false

DOMString mediaTrackId

DOMString transportId

DOMString codecId

unsigned long firCount

Count of FIR packets, defined in [[!RFC5104]] Section 4.3.1.

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.

double targetBitrate

Present bitrate target of this SSRC, in bits per second.

RTCOutboundRTPStreamStats

RTCOutboundRTPStreamStats are relevant to RTCRtpSender objects.

unsigned long packetsSent

Packets sent.

unsigned long long bytesSent

Bytes sent.

double roundTripTime

Estimated round trip time (seconds) based on the RTCP timestamp, as described in [[!RFC3550]] Section 6.4.1.

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. Assume that receiver is an RTCRtpReceiver object. The following example code might be used:

var baselineReport, currentReport;

getStats(receiver, function (report) { 
    baselineReport = report;
});

// ... wait a bit
setTimeout(function () {
        getStats(receiver, function (report) {
        currentReport = report;
        processStats();
    });
}, aBit);

function processStats() {
    // compare the elements from the current report with the baseline
    for each (var now in currentReport) {
        if (now.type != "outbound-rtp")
            continue;

        // get the corresponding stats from the baseline report
        base = baselineReport[now.id];

        if (base) {
            remoteNow = currentReport[now.remoteId];
            remoteBase = baselineReport[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;
        }
    }
}

RTCP Protocol

This specification requires that RTCP packets must be multiplexed with the RTP packets as defined by [[!RFC5761]].

Event summary

The following events fire on RTCDtlsTransport objects:

Event name Interface Fired when...
error Event The RTCDtlsTransport object has received a DTLS Alert.
statechanged RTCDtlsTransportStateChangedEvent The RTCDtlsTransportState changes.

The following events fire on RTCIceTransport objects:

Event name Interface Fired when...
statechanged RTCIceTransportStateChangedEvent The RTCIceTransportState changes.
gatherstatechanged 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 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.

Examples

Simple Peer-to-peer Example

This example code provides a basic audio and video session between two browsers.


Change Log

This section will be removed before publication.

Changes since 29 April 2014

  1. ICE restart explanation added, as described in Issue 59
  2. Fixes for error handling, as described in Issue 75
  3. Fixes for miscellaneous NITs, as described in Issue 76
  4. Enable retrieval of the SSRC to be used by RTCP, as described in Issue 77
  5. Support for retrieval of audio and video capabilities, as described in Issue 81
  6. getStats interface updated, as described in Issue 82
  7. Partially addressed SVC issues described in Issue 83
  8. Partially addressed statistics update issues described in Issue 85

Changes since 12 April 2014

  1. Fixes for error handling, as described in Issue 26
  2. Support for contributing sources removed (re-classified as a 1.2 feature), as described in Issue 27
  3. Cleanup of DataChannel construction, as described in Issue 60
  4. Separate proposal on simulcast/layering, as described in Issue 61
  5. Separate proposal on quality, as described in Issue 62
  6. Fix for TCP candidate type, as described in Issue 63
  7. Fix to the fingerprint attribute, as described in Issue 64
  8. Fix to RTCRtpFeatures, as described in Issue 65
  9. Support for retrieval of remote certificates, as described in Issue 67
  10. Support for ICE error handling, described in Issue 68
  11. Support for Data Channel send rate control, as described in Issue 69
  12. Support for capabilities and settings, as described in Issue 70
  13. Removal of duplicate RTCIceListener functionality, as described in Issue 71
  14. ICE gathering state added, as described in Issue 72
  15. Removed ICE role from the ICE transport constructor, as described in Issue 73

Changes since 13 February 2014

  1. Support for contributing source information added, as described in Issue 27
  2. Support for control of quality, resolution, framerate and layering added, as described in Issue 31
  3. RTCRtpListener object added and figure in Section 1 updated, as described in Issue 32
  4. More complete support for RTP and Codec Parameters added, as described in Issue 33
  5. Data Channel transport problem fixed, as described in Issue 34
  6. Various NITs fixed, as described in Issue 37
  7. Section 2.2 and 2.3 issues fixed, as described in Issue 38
  8. Default values of some dictionary attributes added, to partially address the issue described in Issue 39
  9. Support for ICE TCP added, as described in Issue 41
  10. Fixed issue with sequences as attributes, as described in Issue 43
  11. Fix for issues with onlocalcandidate, as described in Issue 44
  12. Initial stab at a Stats API, as requested in Issue 46
  13. Added support for ICE gather policy, as described in Issue 47

Changes since 07 November 2013

  1. RTCTrack split into RTCRtpSender and RTCRtpReceiver objects, as proposed on 06 January 2014.
  2. RTCConnection split into RTCIceTransport and RTCDtlsTransport objects, as proposed on 09 January 2014.
  3. RTCSctpTransport object added, as described in Issue 25
  4. RTCRtpHeaderExtensionParameters added, as described in Issue 28
  5. RTCIceListener added, in order to support parallel forking, as described in Issue 29
  6. DTMF support added, as described in Issue 30