Abstract

This document defines a set of ECMAScript APIs in WebIDL to allow media to be sent to and received from another browser or device implementing the appropriate set of real-time protocols. This specification is being developed in conjunction with a protocol specification developed by the IETF RTCWEB group and an API specification to get access to local media devices developed by the Media Capture Task Force.

Status of This Document

This section describes the status of this document at the time of its publication. Other documents may supersede this document. A list of current W3C publications and the latest revision of this technical report can be found in the W3C technical reports index at http://www.w3.org/TR/.

The Editors and active contributors of WebRTC 1.0 intend to publish a Candidate Recommendation soon. Consequently, this is a Request for Comments by the WebRTC Working Group to seek wide review of this document.

The API is based on preliminary work done in the WHATWG.

This document was published by the Web Real-Time Communications Working Group as an Editor's Draft. If you wish to make comments regarding this document, please send them to public-webrtc@w3.org (subscribe, archives). All comments are welcome.

Publication as an Editor's Draft does not imply endorsement by the W3C Membership. This is a draft document and may be updated, replaced or obsoleted by other documents at any time. It is inappropriate to cite this document as other than work in progress.

This document was produced by a group operating under the 5 February 2004 W3C Patent Policy. W3C maintains a public list of any patent disclosures made in connection with the deliverables of the group; that page also includes instructions for disclosing a patent. An individual who has actual knowledge of a patent which the individual believes contains Essential Claim(s) must disclose the information in accordance with section 6 of the W3C Patent Policy.

This document is governed by the 1 September 2015 W3C Process Document.

1. Introduction

This section is non-normative.

There are a number of facets to video-conferencing in HTML covered by this specification:

This document defines the APIs used for these features. This specification is being developed in conjunction with a protocol specification developed by the IETF RTCWEB group and an API specification to get access to local media devices [GETUSERMEDIA] developed by the Media Capture Task Force. An overview of the system can be found in [RTCWEB-OVERVIEW] and [RTCWEB-SECURITY].

2. Conformance

As well as sections marked as non-normative, all authoring guidelines, diagrams, examples, and notes in this specification are non-normative. Everything else in this specification is normative.

The key words MAY, MUST, MUST NOT, SHALL, and SHOULD are to be interpreted as described in [RFC2119].

This specification defines conformance criteria that apply to a single product: the user agent that implements the interfaces that it contains.

Conformance requirements phrased as algorithms or specific steps may be implemented in any manner, so long as the end result is equivalent. (In particular, the algorithms defined in this specification are intended to be easy to follow, and not intended to be performant.)

Implementations that use ECMAScript to implement the APIs defined in this specification MUST implement them in a manner consistent with the ECMAScript Bindings defined in the Web IDL specification [WEBIDL-1], as this specification uses that specification and terminology.

3. Terminology

The EventHandler interface, representing a callback used for event handlers, and the ErrorEvent interface are defined in [HTML5].

The concepts queue a task, fire a simple event and networking task source are defined in [HTML5].

The terms event, event handlers and event handler event types are defined in [HTML5].

The terms MediaStream, MediaStreamTrack, and MediaStreamConstraints are defined in [GETUSERMEDIA].

The term media description is defined in [RFC4566].

4. Peer-to-peer connections

4.1 Introduction

An RTCPeerConnection instance allows to establish peer to peer communications. Communications are coordinated via a signaling channel which is provided by unspecified means, but generally by a script in the page via the server, e.g. using XMLHttpRequest [XMLHttpRequest] or Web Sockets [WEBSOCKETS-API].

4.2 Configuration

4.2.1 RTCConfiguration Dictionary

The RTCConfiguration defines a set of parameters to configure how the peer to peer communication established via RTCPeerConnection is established or re-established.

dictionary RTCConfiguration {
    sequence<RTCIceServer>   iceServers;
    RTCIceTransportPolicy    iceTransportPolicy = "all";
    RTCBundlePolicy          bundlePolicy = "balanced";
    RTCRtcpMuxPolicy         rtcpMuxPolicy = "require";
    DOMString                peerIdentity;
    sequence<RTCCertificate> certificates;
    unsigned short           iceCandidatePoolSize = 0;
};
Dictionary RTCConfiguration Members
iceServers of type sequence<RTCIceServer>

An array of objects describing servers available to be used by ICE, such as STUN and TURN server.

iceTransportPolicy of type RTCIceTransportPolicy, defaulting to "all"

Indicates which candidates the ICE agent is allowed to use.

bundlePolicy of type RTCBundlePolicy, defaulting to "balanced"

Indicates which media-bundling policy to use when gathering ICE candidates.

rtcpMuxPolicy of type RTCRtcpMuxPolicy, defaulting to "require"

Indicates which rtcp-mux policy to use when gathering ICE candidates.

peerIdentity of type DOMString

Sets the target peer identity for the RTCPeerConnection. The RTCPeerConnection will not establish a connection to a remote peer unless it can be successfully authenticated with the provided name.

certificates of type sequence<RTCCertificate>

A set of certificates that the RTCPeerConnection uses to authenticate.

Valid values for this parameter are created through calls to the generateCertificate function.

Although any given DTLS connection will use only one certificate, this attribute allows the caller to provide multiple certificates that support different algorithms. The final certificate will be selected based on the DTLS handshake, which establishes which certificates are allowed. The RTCPeerConnection implementation selects which of the certificates is used for a given connection; how certificates are selected is outside the scope of this specification.

If this value is absent, then a set of certificates are generated for each RTCPeerConnection instance.

This option allows applications to establish key continuity. An RTCCertificate can be persisted in [INDEXEDDB] and reused. Persistence and reuse also avoids the cost of key generation.

The value for this configuration option cannot change after its value is initially selected.

iceCandidatePoolSize of type unsigned short, defaulting to 0

Size of the prefetched ICE pool as defined in [JSEP] (section 3.5.4. and section 4.1.1.)

4.2.2 RTCIceCredentialType Enum

enum RTCIceCredentialType {
    "password",
    "token"
};
Enumeration description
password The credential is a long-term authentication password, as described in [RFC5389], Section 10.2.
token The credential is an access token, as described in [TRAM-TURN-THIRD-PARTY-AUTHZ], Section 6.2.

4.2.3 RTCIceServer Dictionary

The RTCIceServer dictionary is used to describe the STUN and TURN servers that can be used by the ICE agent to establish a connection with a peer.

dictionary RTCIceServer {
    required (DOMString or sequence<DOMString>) urls;
             DOMString                          username;
             DOMString                          credential;
             RTCIceCredentialType               credentialType = "password";
};
Dictionary RTCIceServer Members
urls of type (DOMString or sequence<DOMString>), required

STUN or TURN URI(s) as defined in [RFC7064] and [RFC7065] or other URI types.

username of type DOMString

If this RTCIceServer object represents a TURN server, then this attribute specifies the username to use with that TURN server.

credential of type DOMString

If this RTCIceServer object represents a TURN server, then this attribute specifies the credential to use with that TURN server.

credentialType of type RTCIceCredentialType, defaulting to "password"

If this RTCIceServer object represents a TURN server, then this attribute specifies how credential should be used when that TURN server requests authorization.

An example array of RTCIceServer objects is:

Example 1
[
     { "urls": "stun:stun1.example.net" },
     { "urls": ["turns:turn.example.org", "turn:turn.example.net"],
       "username": "user",
       "credential": "myPassword",
       "credentialType": "password" }
]

4.2.4 RTCIceTransportPolicy Enum

As noted in [JSEP] (section 4.1.1.), if RTCIceTransportPolicy is specified, it causes the browser to only surface the permitted candidates to the application, and only use those candidates for connectivity checks.

enum RTCIceTransportPolicy {
    "relay",
    "all"
};
Enumeration description
relay The ICE agent MUST only use 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.
all The ICE agent may use any type of candidates when this value is specified. This will not include addresses that have been filtered by the browser.

4.2.5 RTCBundlePolicy Enum

As described in [JSEP] (section 4.1.1.), BUNDLE policy affects which media tracks are negotiated if the remote endpoint is not BUNDLE-aware, and what ICE candidates are gathered. If the remote endpoint is BUNDLE-aware, all media tracks and data channels are BUNDLEd onto the same transport.

enum RTCBundlePolicy {
    "balanced",
    "max-compat",
    "max-bundle"
};
Enumeration description
balanced Gather ICE candidates for each media type in use (audio, video, and data). If the remote endpoint is not BUNDLE-aware, negotiate only one audio and video track on separate transports.
max-compat Gather ICE candidates for each track. If the remote endpoint is not BUNDLE-aware, negotiate all media tracks on separate transports.
max-bundle Gather ICE candidates for only one track. If the remote endpoint is not BUNDLE-aware, negotiate only one media track.

4.2.6 RTCRtcpMuxPolicy Enum

Defined in [JSEP] (section 4.1.1.). The following is a non-normative summary for convenience.

The RtcpMuxPolicy affects what ICE candidates are gathered to support non-multiplexed RTCP.

enum RTCRtcpMuxPolicy {
    "negotiate",
    "require"
};
Enumeration description
negotiate Gather ICE candidates for both RTP and RTCP candidates. If the remote-endpoint is capable of multiplexing RTCP, multiplex RTCP on the RTP candidates. If it is not, use both the RTP and RTCP candidates separately.
require Gather ICE candidates only for RTP and multiplex RTCP on the RTP candidates. If the remote endpoint is not capable of rtcp-mux, session negotiation will fail.

4.2.7 Offer/Answer Options

These dictionaries describe the options that can be used to control the offer/answer creation process.

dictionary RTCOfferAnswerOptions {
    boolean voiceActivityDetection = true;
};
Dictionary RTCOfferAnswerOptions Members
voiceActivityDetection of type boolean, defaulting to true

Many codecs and systems are capable of detecting "silence" and changing their behavior in this case by doing things such as not transmitting any media. In many cases, such as when dealing with emergency calling or sounds other than spoken voice, it is desirable to be able to turn off this behavior. This option allows the application to provide information about whether it wishes this type of processing enabled or disabled.

dictionary RTCOfferOptions : RTCOfferAnswerOptions {
    boolean iceRestart = false;
};
Dictionary RTCOfferOptions Members
iceRestart of type boolean, defaulting to false

When the value of this dictionary member is true, the generated description will have ICE credentials that are different from the current credentials (as visible in the localDescription attribute's SDP). Applying the generated description will restart ICE.

When the value of this dictionary member is false, and the localDescription attribute has valid ICE credentials, the generated description will have the same ICE credentials as the current value from the localDescription attribute.

dictionary RTCAnswerOptions : RTCOfferAnswerOptions {
};

4.3 RTCPeerConnection Interface

The general operation of the RTCPeerConnection is described in [JSEP].

4.3.1 Operation

Calling new RTCPeerConnection(configuration ) creates an RTCPeerConnection object.

The configuration has the information to find and access the servers used by ICE. There may be multiple servers of each type and any TURN server also acts as a STUN server.

An RTCPeerConnection object has a signaling state, an ICE gathering state, and an ICE connection state. These are initialized when the object is created.

The ICE protocol implementation of an RTCPeerConnection is represented by an ICE agent [ICE]. The User Agent MUST respond to the following events triggered by the ICE Agent:

When the RTCPeerConnection() constructor is invoked, the user agent MUST run the following steps:

  1. Let connection be a newly created RTCPeerConnection object.

  2. Initialize connection's ICE Agent.

  3. Set the configuration specified by the constructor's first argument.

  4. Let connection have an [[isClosed]] internal slot, initialized to false.

  5. Set connection's signaling state to stable.

  6. Set connection's ICE connection state to new.

  7. Set connection's ICE gathering state to new.

  8. Set connection's pendingLocalDescription, currentLocalDescription, pendingRemoteDescription and currentRemoteDescription to null.

  9. Initialize an internal variable operations to represent a queue of operations with an empty array.

  10. If the certificates value in the RTCConfiguration structure is non-empty, check that the expires on each value is in the future. If a certificate has expired, throw an InvalidAccessError exception and abort these steps; otherwise, store the certificates. If no certificates value was specified, one or more new RTCCertificate instances are generated for use with this RTCPeerConnection instance.

  11. Return connection.

Once the RTCPeerConnection object has been initialized, for every call to createOffer, setLocalDescription, createAnswer, setRemoteDescription, and addIceCandidate, execute the following steps:

  1. Let p be a new promise.

  2. Append an object representing the current call being handled (i.e. function name and corresponding arguments) to the operations array.

  3. If the length of the operations array is exactly 1, execute the object from the front of the queue.

  4. Upon fulfillment or rejection of the promise returned by the function, fulfill or reject p with the corresponding value or reason. Upon fulfillment or rejection of p, execute the following steps:

    1. Remove the corresponding object from the operations array.

    2. If the array is non-empty, execute the first object queued.

  5. Return p.

The general idea is to have only one among createOffer, setLocalDescription, createAnswer and setRemoteDescription and addIceCandidate executing at any given time. If subsequent calls are made while the returned promise of a previous call is still unsettled, they are added to a queue and executed when all the previous calls are executed and their promises are settled.

When a new ICE candidate is available or when the ICE gathering process is done , the user agent MUST queue a task to run the following steps:

  1. Let connection be the RTCPeerConnection object associated with this ICE Agent.

  2. If connection's [[isClosed]] slot is true, abort these steps.

  3. If the intent of the ICE Agent is to notify the script that:

  4. Fire an event named icecandidate with newCandidate at connection.

To update the ICE gathering state of an RTCPeerConnection instance connection to newState, the User Agent MUST queue a task that runs the following steps:

  1. If connection's [[isClosed]] slot is true or connection's ice gathering state has the same value as newState, abort these steps.

  2. Set connection's ice gathering state to newState.

  3. Fire a simple event named icegatheringstatechange at connection.

To update the ICE connection state of an RTCPeerConnection instance connection to newState, the User Agent MUST queue a task that runs the following steps:

  1. If connection's [[isClosed]] slot is true or connection's ice connection state has the same value as newState, abort these steps.

  2. Set connection's ice connection state to newState.

  3. Fire a simple event named iceconnectionstatechange at connection.

To set an RTCSessionDescription description on an RTCPeerConnection object connection, run the following steps:

  1. If connection's [[isClosed]] slot is true, the user agent MUST return a promise rejected with an InvalidStateError.

  2. Let p be a new promise.

  3. In parallel, start the process to apply description as described in [JSEP] (section 5.4. and section 5.5.).

    1. If the process to apply description fails for any reason, then user agent MUST queue a task runs the following steps:

      1. If connection's [[isClosed]] slot is true, then abort these steps.

      2. If elements of the SDP were modified in an invalid way as specified in [JSEP] (section 6.), then reject p with an InvalidModificationError and abort these steps.

      3. If the description's type is wrong for the current signaling state of connection, then reject p with a InvalidStateError and abort these steps.

      4. If the content of description is invalid, then reject p with an InvalidAccessError and abort these steps.

      5. For all other errors, for example if description cannot be applied at the media layer, reject p with OperationError.

    2. If description is applied successfully, the user agent MUST queue a task that runs the following steps:

      1. If connection's [[isClosed]] slot is true, then abort these steps.

      2. If description is set as a local description, and its content matches the state of all tracks and data channels, as defined below, clear the negotiation-needed flag.

        Issue 1

        NOTE: The principles of pending and current SDP were agreed by the WG but the details in the next steps have not yet been fully reviewed. TODO - review this.

      3. If description is set as a local description, then run one of the following steps:

      4. Otherwise, if description is set as a remote description, then run one of the following steps:

      5. If connection's signaling state changed above, fire a simple event named signalingstatechange at connection.

      6. If description is set as a local description, connection's ICE gathering state is new, and description contains media, then update connection's ICE gathering state to gathering.

      7. If the process to apply description resulted in an ICE restart [JSEP] (section 5.7. and section 5.8.), then run the following steps:

        1. If connection is not already gathering, update connection's ICE gathering state to gathering.

        2. If connection's ICE connection state is completed, update connection's ICE connection state to connected.

      8. If description is set as a remote description with new media descriptions [JSEP], the User Agent MUST dispatch a receiver for all new media descriptions.

      9. If connection's signaling state is now stable, and the negotiation-needed flag is set, the User Agent MUST queue a task to fire a simple event named negotiationneeded at connection and clear the negotiation-needed flag.

      10. Resolve p with undefined.

  4. Return p.

The task source for the tasks listed in this section is the networking task source.

4.3.2 Interface Definition

The RTCPeerConnection interface presented in this section is extended by several partial interfaces throughout this specification. Notably, the RTP Media API section, that adds the APIs to send and receive MediaStreamTrack objects.

[Constructor(optional RTCConfiguration configuration)]
interface RTCPeerConnection : EventTarget {
    Promise<RTCSessionDescriptionInit> createOffer(optional RTCOfferOptions options);
    Promise<RTCSessionDescriptionInit> createAnswer(optional RTCAnswerOptions options);
    Promise<void>                      setLocalDescription(RTCSessionDescriptionInit description);
    readonly        attribute RTCSessionDescription?    localDescription;
    readonly        attribute RTCSessionDescription?    currentLocalDescription;
    readonly        attribute RTCSessionDescription?    pendingLocalDescription;
    Promise<void>                      setRemoteDescription(RTCSessionDescriptionInit description);
    readonly        attribute RTCSessionDescription?    remoteDescription;
    readonly        attribute RTCSessionDescription?    currentRemoteDescription;
    readonly        attribute RTCSessionDescription?    pendingRemoteDescription;
    Promise<void>                      addIceCandidate((RTCIceCandidateInit or RTCIceCandidate)? candidate);
    readonly        attribute RTCSignalingState         signalingState;
    readonly        attribute RTCIceGatheringState      iceGatheringState;
    readonly        attribute RTCIceConnectionState     iceConnectionState;
    readonly        attribute RTCPeerConnectionState    connectionState;
    readonly        attribute boolean?                  canTrickleIceCandidates;
    static readonly attribute FrozenArray<RTCIceServer> defaultIceServers;
    RTCConfiguration                   getConfiguration();
    void                               setConfiguration(RTCConfiguration configuration);
    void                               close();
                    attribute EventHandler              onnegotiationneeded;
                    attribute EventHandler              onicecandidate;
                    attribute EventHandler              onicecandidateerror;
                    attribute EventHandler              onsignalingstatechange;
                    attribute EventHandler              oniceconnectionstatechange;
                    attribute EventHandler              onicegatheringstatechange;
                    attribute EventHandler              onconnectionstatechange;
};
Constructors
RTCPeerConnection
See the RTCPeerConnection constructor algorithm.
Parameter Type Nullable Optional Description
configuration RTCConfiguration ✘ ✔
Attributes
localDescription of type RTCSessionDescription, readonly , nullable

The localDescription attribute MUST return pendingLocalDescription if it is not null and otherwise it MUST return currentLocalDescription.

currentLocalDescription of type RTCSessionDescription, readonly , nullable

The currentLocalDescription attribute represents the local RTCSessionDescription that was successfully negotiated the last time the RTCPeerConnection transitioned into the stable state plus any local candidates that have been generated by the ICE Agent since the offer or answer was created.

The currentLocalDescription attribute MUST return the last value that algorithms in this specification set it to, completed with any local candidates that have been generated by the ICE Agent since the offer or answer was created. Prior to being set, it returns null.

pendingLocalDescription of type RTCSessionDescription, readonly , nullable

The pendingLocalDescription attribute represents a local RTCSessionDescription that is in the process of being negotiated plus any local candidates that have been generated by the ICE Agent since the offer or answer was created. If the RTCPeerConnection is in the stable state, the value is null. This attribute is updated by setLocalDescription.

The pendingLocalDescription attribute MUST return the last value that algorithms in this specification set it to, completed with any local candidates that have been generated by the ICE Agent since the offer or answer was created. Prior to being set, it returns null.

remoteDescription of type RTCSessionDescription, readonly , nullable

The remoteDescription attribute MUST return pendingRemoteDescription if it is not null and otherwise it MUST return currentRemoteDescription.

currentRemoteDescription of type RTCSessionDescription, readonly , nullable

The currentRemoteDescription attribute represents the last remote RTCSessionDescription that was successfully negotiated the last time the RTCPeerConnection transitioned into the stable state plus any remote candidates that have been supplied via addIceCandidate() since the offer or answer was created.

The currentRemoteDescription attribute MUST return the value that algorithms in this specification set it to, completed with any remote candidates that have been supplied via addIceCandidate() since the offer or answer was created. Prior to being set, it returns null.

pendingRemoteDescription of type RTCSessionDescription, readonly , nullable

The pendingRemoteDescription attribute represents a remote RTCSessionDescription that is in the process of being negotiated, completed with any remote candidates that have been supplied via addIceCandidate() since the offer or answer was created. If the RTCPeerConnection is in the stable state, the value is null. This attribute is updated by setLocalDescription.

The pendingRemoteDescription attribute MUST return the value that algorithms in this specification set it to, completed with any remote candidates that have been supplied via addIceCandidate() since the offer or answer was created. Prior to being set, it returns null.

signalingState of type RTCSignalingState, readonly

The signalingState attribute MUST return the RTCPeerConnection object's signaling state.

iceGatheringState of type RTCIceGatheringState, readonly

The iceGatheringState attribute MUST return the ICE gathering state of the RTCPeerConnection instance.

iceConnectionState of type RTCIceConnectionState, readonly

The iceConnectionState attribute MUST return the ICE connection state of the RTCPeerConnection instance.

connectionState of type RTCPeerConnectionState, readonly

The connectionState attribute MUST return the aggregate of the states of the DtlsTransports and IceTransports of the RTCPeerConnection, as describe in the values of the RTCPeerConnectionState enum.

canTrickleIceCandidates of type boolean, readonly , nullable

The canTrickleIceCandidates attribute indicates whether the remote peer is able to accept trickled ICE candidates [TRICKLE-ICE]. The value is determined based on whether a remote description indicates support for trickle ICE, as defined in [JSEP] (section 4.1.11.). Prior to the completion of setRemoteDescription, this value is null.

defaultIceServers of type FrozenArray<RTCIceServer>, static readonly

The defaultIceServers attribute provides a list of ICE servers that are configured into the browser. A browser might be configured to use local or private STUN or TURN servers. This method allows an application to learn about these servers and optionally use them.

onnegotiationneeded of type EventHandler
The event type of this event handler is negotiationneeded.
onicecandidate of type EventHandler
The event type of this event handler is icecandidate.
onicecandidateerror of type EventHandler
The event type of this event handler is icecandidateerror.
onsignalingstatechange of type EventHandler
The event type of this event handler is signalingstatechange.
oniceconnectionstatechange of type EventHandler
The event type of this event handler is iceconnectionstatechange
onicegatheringstatechange of type EventHandler
The event type of this event handler is icegatheringstatechange.
onconnectionstatechange of type EventHandler
The event type of this event handler is connectionstatechange.
Methods
createOffer

The createOffer method generates a blob of SDP that contains an RFC 3264 offer with the supported configurations for the session, including descriptions of the local MediaStreamTracks attached to this RTCPeerConnection, the codec/RTP/RTCP options supported by this implementation, and any candidates that have been gathered by the ICE Agent. The options parameter may be supplied to provide additional control over the offer generated.

As an offer, the generated SDP will contain the full set of capabilities supported by the session (as opposed to an answer, which will include only a specific negotiated subset to use); for each SDP line, the generation of the SDP MUST follow the appropriate process for generating an offer. In the event createOffer is called after the session is established, createOffer will generate an offer that is compatible with the current session, incorporating any changes that have been made to the session since the last complete offer-answer exchange, such as addition or removal of tracks. If no changes have been made, the offer will include the capabilities of the current local description as well as any additional capabilities that could be negotiated in an updated offer.

Session descriptions generated by createOffer MUST be immediately usable by setLocalDescription without causing an error as long as setLocalDescription is called reasonably soon. If a system has limited resources (e.g. a finite number of decoders), createOffer needs to return an offer that reflects the current state of the system, so that setLocalDescription will succeed when it attempts to acquire those resources. The session descriptions MUST remain usable by setLocalDescription without causing an error until at least the end of the fulfillment callback of the returned promise. Calling this method is needed to get the ICE user name fragment and password.

The value for certificates in the RTCConfiguration for the RTCPeerConnection is used to produce a set of certificate fingerprints. These certificate fingerprints are used in the construction of SDP and as input to requests for identity assertions.

If the RTCPeerConnection is configured to generate Identity assertions by calling setIdentityProvider, then the session description SHALL contain an appropriate assertion. If the identity provider is unable to produce an identity assertion, the call to createOffer MUST be rejected with a DOMException that has a name of NotReadableError.

If this RTCPeerConnection object is closed before the SDP generation process completes, the user agent MUST suppress the result and not resolve or reject the returned promise.

If the SDP generation process completed successfully, the user agent MUST resolve the returned promise with a newly created RTCSessionDescription object, representing the generated offer.

The SDP generation process exposes a subset of the media capabilities of the underlying system, which provides generally persistent cross-origin information on the device. It thus increases the fingerprinting surface of the application. In privacy-sensitive contexts, browsers can consider mitigations such as generating SDP matching only a common subset of the capabilities.(This is a fingerprinting vector.)

If the SDP generation process failed for any other reason, the user agent MUST reject the returned promise with an DOMException object of type OperationError as its argument.

Parameter Type Nullable Optional Description
options RTCOfferOptions ✘ ✔
Return type: Promise<RTCSessionDescriptionInit>
createAnswer

The createAnswer method generates an [SDP] answer with the supported configuration for the session that is compatible with the parameters in the remote configuration. Like createOffer, the returned blob contains descriptions of the local MediaStreamTracks attached to this RTCPeerConnection, the codec/RTP/RTCP options negotiated for this session, and any candidates that have been gathered by the ICE Agent. The options parameter may be supplied to provide additional control over the generated answer.

As an answer, the generated SDP will contain a specific configuration that, along with the corresponding offer, specifies how the media plane should be established. The generation of the SDP MUST follow the appropriate process for generating an answer.

Session descriptions generated by createAnswer MUST be immediately usable by setLocalDescription without causing an error as long as setLocalDescription is called reasonably soon. Like createOffer, the returned description SHOULD reflect the current state of the system. The session descriptions MUST remain usable by setLocalDescription without causing an error until at least the end of the fulfillment callback of the returned promise. Calling this method is needed to get the ICE user name fragment and password.

An answer can be marked as provisional, as described in [JSEP] (section 4.1.4.1.), by setting the type to pranswer.

If the RTCPeerConnection is configured to generate Identity assertions by calling setIdentityProvider, then the session description SHALL contain an appropriate assertion. If the identity provider is unable to produce an identity assertion, the call to createAnswer MUST be rejected with a DOMException that has a name of NotReadableError.

If this RTCPeerConnection object is closed before the SDP generation process completes, the user agent MUST suppress the result and not resolve or reject the returned promise.

If the SDP generation process completed successfully, the user agent MUST resolve the returned promise with a newly created RTCSessionDescription object, representing the generated answer.

If the SDP generation process failed for any reason, the user agent MUST reject the returned promise with a DOMException object of type OperationError.

Parameter Type Nullable Optional Description
options RTCAnswerOptions ✘ ✔
Return type: Promise<RTCSessionDescriptionInit>
setLocalDescription

The setLocalDescription method instructs the RTCPeerConnection to apply the supplied RTCSessionDescriptionInit as the local description.

This API changes the local media state. In order to successfully handle scenarios where the application wants to offer to change from one media format to a different, incompatible format, the RTCPeerConnection MUST be able to simultaneously support use of both the current and pending local descriptions (e.g. support codecs that exist in both descriptions) until a final answer is received, at which point the RTCPeerConnection can fully adopt the pending local description, or rollback to the current description if the remote side rejected the change.

When the method is invoked, the user agent must set the RTCSessionDescription indicated by the method's first argument.

[JSEP] (section 6.) specifies what elements of the SDP returned by createOffer can be changed before passing it to setLocalDescription.

Parameter Type Nullable Optional Description
description RTCSessionDescriptionInit ✘ ✘
Return type: Promise<void>
setRemoteDescription

The setRemoteDescription method instructs the RTCPeerConnection to apply the supplied RTCSessionDescriptionInit as the remote offer or answer. This API changes the local media state.

When the method is invoked, the user agent must set the RTCSessionDescription indicated by the method's first argument. In addition, a remote description is processed to determine and verify the identity of the peer.

If an a=identity attribute is present in the session description, the browser validates the identity assertion..

If the "peerIdentity" configuration is applied to the RTCPeerConnection, this establishes a target peer identity of the provided value. Alternatively, if the RTCPeerConnection has previously authenticated the identity of the peer (that is, there is a current value for peerIdentity ), then this also establishes a target peer identity.

The target peer identity cannot be changed once set. Once set, if a different value is provided, the user agent MUST reject the returned promise with InvalidModificationError and abort this operation. The RTCPeerConnection MUST be closed if the validated peer identity does not match the target peer identity.

If there is no target peer identity, then setRemoteDescription does not await the completion of identity validation.

Parameter Type Nullable Optional Description
description RTCSessionDescriptionInit ✘ ✘
Return type: Promise<void>
addIceCandidate

The addIceCandidate() method provides a remote candidate to the ICE Agent. This method can also be used to indicate the end of remote candidates when called with a null value for candidate. The only members of the argument used by this method are candidate, sdpMid and sdpMLineIndex; the rest are ignored.

  1. Let connection be the RTCPeerConnection object on which the method was invoked.

  2. If connection's [[isClosed]] slot is true, return a promise rejected with an InvalidStateError.

  3. Let candidate be the methods argument.

  4. If candidate is not null but is missing values for both sdpMid and sdpMLineIndex, return a promise rejected with a TypeError.

  5. Let p be a new promise.

  6. In parallel, start the process to apply candidate.

    Note
    We probably want to define the process to 'apply an ICE candidate' in JSEP and link to it above.

    1. If candidate is null, the User Agent MUST queue a task that runs the following steps:

      1. For each media description in the last successfully applied remote description, perform the processing for an end-of-candidates indication for said media description as defined in [TRICKLE-ICE].

      2. Resolve p with undefined.

    2. If candidate could not be successfully, applied the User Agent MUST queue a task that runs the following steps:

      1. If connection's [[isClosed]] slot is true, then abort these steps.

      2. Reject p with a DOMException object whose name attribute has the value OperationError and abort these steps.

    3. If candidate is applied successfully, the User Agent MUST queue a task that runs the following steps:

      1. If connection's [[isClosed]] slot is true, then abort these steps.

      2. Let remoteDescription be connection's pendingRemoteDescription if not null, otherwise connection's currentRemoteDescription.

      3. Add candidate to remoteDescription.

      4. If the ICE Agent is not currently checking candidate pairs, the ICE Agent MUST start checking candidate pairs and update connection's ICE connection state to checking.

      5. Resolve p with undefined.

  7. Return p.

Parameter Type Nullable Optional Description
candidate (RTCIceCandidateInit or RTCIceCandidate) ✔ ✘
Return type: Promise<void>
getConfiguration

Returns a RTCConfiguration object representing the current configuration of this RTCPeerConnection object.

When this method is call, the user agent MUST a construct new RTCConfiguration object to be returned, and initialize it using the ICE Agent's ICE transports setting and ICE servers list.

The returned configuration MUST include a certificates attribute containing the candidate set of certificates used for connecting to peers. This attribute contains the certificates chosen by the application, or the certificates generated by the user agent for use with this RTCPeerConnection instance.

No parameters.
Return type: RTCConfiguration
setConfiguration

The setConfiguration method updates the ICE Agent process of gathering local candidates and pinging remote candidates.

This call may result in a change to the state of the ICE Agent, and may result in a change to media state if it results in connectivity being established.

When the setConfiguration method is invoked, the user agent MUST run the following steps:

  1. Let connection be the RTCPeerConnection on which the method was invoked.

  2. If connection's [[isClosed]] slot is true, throw an InvalidStateError exception and abort these steps.

  3. Set the configuration specified by the methods argument on connection.

To set a configuration, run the following steps:

  1. Let configuration be the RTCConfiguration dictionary to be processed.
  2. Let connection be the target RTCPeerConnection object.
  3. If configuration.peerIdentity is set and its value differs from the target peer identity, throw an InvalidModificationError.
  4. If configuration.certificates is set and the set of certificates differs from the ones used when connection was constructed, throw an InvalidModificationError.

  5. Let the value of configuration.iceTransportPolicy be the ICE Agent's ICE transports setting.

  6. Let the value of configuration.bundlePolicy be connection's bundle policy.

  7. Let the value of configuration.iceCandidatePoolSize be the ICE Agent's prefetched ICE candidate pool size as defined in [JSEP] (section 4.1.12.).

  8. Let validatedServers be an empty list.

  9. If configuration.iceServers is defined, then run the following steps for each element:

    1. Let server be the current list element.

    2. If server.urls is a string, let server.urls be a list consisting of just that string.

    3. For each url in server.urls parse url and obtain scheme name. If the scheme name is not implemented by the browser, or if parsing based on the syntax defined in [RFC7064] and [RFC7065] fails, throw a SyntaxError and abort these steps.

    4. If scheme name is turn or turns, and either of server.username or server.credential are omitted, then throw an InvalidAccessError and abort these steps.

    5. Appendserver to validatedServers.

    Let validatedServers be the ICE Agent's ICE servers list.

    If a new list of servers replaces the ICE Agent's existing ICE servers list, no action will be taken until the RTCPeerConnection's ICE gathering state transitions to gathering. If a script wants this to happen immediately, it should do an ICE restart.

Parameter Type Nullable Optional Description
configuration RTCConfiguration ✘ ✘
Return type: void
close

When the close method is invoked, the user agent MUST run the following steps:

  1. Let connection be the RTCPeerConnection object on which the method was invoked.

  2. If connection's [[isClosed]] slot is true, abort these steps.

  3. Destroy connection's ICE Agent, abruptly ending any active ICE processing and any active streaming, and releasing any relevant resources (e.g. TURN permissions).

  4. All RTCRtpSenders in connection's set of senders are now considered stopped.

  5. Set connection's [[isClosed]] slot to true.

No parameters.
Return type: void

4.3.3 Legacy Interface Extensions

Note
These methods are kept on RTCPeerConnection for legacy purposes. 
partial interface RTCPeerConnection {
    Promise<void> createOffer(RTCSessionDescriptionCallback successCallback,
                              RTCPeerConnectionErrorCallback failureCallback,
                              optional RTCOfferOptions options);
    Promise<void> setLocalDescription(RTCSessionDescriptionInit description,
                                      VoidFunction successCallback,
                                      RTCPeerConnectionErrorCallback failureCallback);
    Promise<void> createAnswer(RTCSessionDescriptionCallback successCallback,
                               RTCPeerConnectionErrorCallback failureCallback);
    Promise<void> setRemoteDescription(RTCSessionDescriptionInit description,
                                       VoidFunction successCallback,
                                       RTCPeerConnectionErrorCallback failureCallback);
    Promise<void> addIceCandidate((RTCIceCandidateInit or RTCIceCandidate) candidate,
                                  VoidFunction successCallback,
                                  RTCPeerConnectionErrorCallback failureCallback);
    Promise<void> getStats(MediaStreamTrack? selector,
                           RTCStatsCallback successCallback,
                           RTCPeerConnectionErrorCallback failureCallback);
};
Methods
createOffer

When the createOffer method is called, the user agent MUST run the following steps:

  1. Let successCallback be the method's first argument.

  2. Let failureCallback be the callback indicated by the method's second argument.

  3. Let options be the callback indicated by the method's third argument.

  4. Run the steps specified by RTCPeerConnection's createOffer() method with options as the sole argument, and let p be the resulting promise.

  5. Upon fulfillment of p with value offer, invoke successCallback with offer as the argument.

  6. Upon rejection of p with reason r, invoke failureCallback with r as the argument.

  7. Return a promise resolved with undefined.

Parameter Type Nullable Optional Description
successCallback RTCSessionDescriptionCallback ✘ ✘
failureCallback RTCPeerConnectionErrorCallback ✘ ✘
options RTCOfferOptions ✘ ✔
Return type: Promise<void>
setLocalDescription

When the setLocalDescription method is called, the user agent MUST run the following steps:

  1. Let description be the method's first argument.

  2. Let successCallback be the callback indicated by the method's second argument.

  3. Let failureCallback be the callback indicated by the method's third argument.

  4. Run the steps specified by RTCPeerConnection's setLocalDescription method with description as the sole argument, and let p be the resulting promise.

  5. Upon fulfillment of p, invoke successCallback with undefined as the argument.

  6. Upon rejection of p with reason r, invoke failureCallback with r as the argument.

  7. Return a promise resolved with undefined.

Parameter Type Nullable Optional Description
description RTCSessionDescriptionInit ✘ ✘
successCallback VoidFunction ✘ ✘
failureCallback RTCPeerConnectionErrorCallback ✘ ✘
Return type: Promise<void>
createAnswer

When the createAnswer method is called, the user agent MUST run the following steps:

  1. Let successCallback be the method's first argument.

  2. Let failureCallback be the callback indicated by the method's second argument.

  3. Run the steps specified by RTCPeerConnection's createAnswer() method with no arguments, and let p be the resulting promise.

  4. Upon fulfillment of p with value answer, invoke successCallback with answer as the argument.

  5. Upon rejection of p with reason r, invoke failureCallback with r as the argument.

  6. Return a promise resolved with undefined.

Parameter Type Nullable Optional Description
successCallback RTCSessionDescriptionCallback ✘ ✘
failureCallback RTCPeerConnectionErrorCallback ✘ ✘
Return type: Promise<void>
setRemoteDescription

When the setRemoteDescription method is called, the user agent MUST run the following steps:

  1. Let description be the method's first argument.

  2. Let successCallback be the callback indicated by the method's second argument.

  3. Let failureCallback be the callback indicated by the method's third argument.

  4. Run the steps specified by RTCPeerConnection's setRemoteDescription method with description as the sole argument, and let p be the resulting promise.

  5. Upon fulfillment of p, invoke successCallback with undefined as the argument.

  6. Upon rejection of p with reason r, invoke failureCallback with r as the argument.

  7. Return a promise resolved with undefined.

Parameter Type Nullable Optional Description
description RTCSessionDescriptionInit ✘ ✘
successCallback VoidFunction ✘ ✘
failureCallback RTCPeerConnectionErrorCallback ✘ ✘
Return type: Promise<void>
addIceCandidate

When the addIceCandidate method is called, the user agent MUST run the following steps:

  1. Let candidate be the method's first argument.

  2. Let successCallback be the callback indicated by the method's second argument.

  3. Let failureCallback be the callback indicated by the method's third argument.

  4. Run the steps specified by RTCPeerConnection's addIceCandiddate() method with candidate as the sole argument, and let p be the resulting promise.

  5. Upon fulfillment of p, invoke successCallback with undefined as the argument.

  6. Upon rejection of p with reason r, invoke failureCallback with r as the argument.

  7. Return a promise resolved with undefined.

Parameter Type Nullable Optional Description
candidate (RTCIceCandidateInit or RTCIceCandidate) ✘ ✘
successCallback VoidFunction ✘ ✘
failureCallback RTCPeerConnectionErrorCallback ✘ ✘
Return type: Promise<void>
getStats

When the getStats method is called, the user agent MUST run the following steps:

  1. Let selector be the method's first argument.

  2. Let successCallback be the callback indicated by the method's second argument.

  3. Let failureCallback be the callback indicated by the method's third argument.

  4. Run the steps specified by RTCPeerConnection's getStats() method with selector as the sole argument, and let p be the resulting promise.

  5. Upon fulfillment of p with value report, invoke successCallback with report as the argument.

  6. Upon rejection of p with reason r, invoke failureCallback with r as the argument.

  7. Return a promise resolved with undefined.

Parameter Type Nullable Optional Description
selector MediaStreamTrack ✔ ✘
successCallback RTCStatsCallback ✘ ✘
failureCallback RTCPeerConnectionErrorCallback ✘ ✘
Return type: Promise<void>

4.3.4 Garbage collection

An RTCPeerConnection object MUST not be garbage collected as long as any event can cause an event handler to be triggered on the object. When the object's [[isClosed]] internal slot is true, no such event handler can be triggered and it is therefore safe to garbage collect the object.

All RTCDataChannel and MediaStreamTrack objects that are connected to a RTCPeerConnection have a strong reference to the RTCPeerConnection object.

4.4 State Definitions

4.4.1 RTCSignalingState Enum

enum RTCSignalingState {
    "stable",
    "have-local-offer",
    "have-remote-offer",
    "have-local-pranswer",
    "have-remote-pranswer"
};
Enumeration description
stable There is no offer­answer exchange in progress. This is also the initial state in which case the local and remote descriptions are empty.
have-local-offer A local description, of type "offer", has been successfully applied.
have-remote-offer A remote description, of type "offer", has been successfully applied.
have-local-pranswer A remote description of type "offer" has been successfully applied and a local description of type "pranswer" has been successfully applied.
have-remote-pranswer A local description of type "offer" has been successfully applied and a remote description of type "pranswer" has been successfully applied.
signalling state transition diagram
Fig. 1 Non-normative signalling state transitions diagram

An example set of transitions might be:

Caller transition:
  • new RTCPeerConnection(): stable
  • setLocal(offer): have-local-offer
  • setRemote(pranswer): have-remote-pranswer
  • setRemote(answer): stable
Callee transition:
  • new RTCPeerConnection(): stable
  • setRemote(offer): have-remote-offer
  • setLocal(pranswer): have-local-pranswer
  • setLocal(answer): stable

4.4.2 RTCIceGatheringState Enum

enum RTCIceGatheringState {
    "new",
    "gathering",
    "complete"
};
Enumeration description
new The object was just created, and no networking has occurred yet.
gathering The ICE agent is in the process of gathering candidates for this RTCPeerConnection.
complete The ICE agent has completed gathering. Events such as adding a new interface or a new TURN server will cause the state to go back to gathering.

4.4.3 RTCPeerConnectionState Enum

enum RTCPeerConnectionState {
    "new",
    "connecting",
    "connected",
    "disconnected",
    "failed",
    "closed"
};
Enumeration description
new Any of the RTCIceTransports or RTCDtlsTransports are in the new state and none of the transports are in the connecting, checking, failed or disconnected state, or all transports are in the closed state.
connecting Any of the RTCIceTransports or RTCDtlsTransports are in the connecting or checking state and none of them is in the failed state.
connected All RTCIceTransports and RTCDtlsTransports are in the connected, completed or closed state and at least of them is in the connected or completed state.
disconnected Any of the RTCIceTransports or RTCDtlsTransports are in the disconnected state and none of them are in the failed or connecting or checking state.
failed Any of the RTCIceTransports or RTCDtlsTransports are in a failed state.
closed The RTCPeerConnection object's [[isClosed]] slot is true.

4.4.4 RTCIceConnectionState Enum

enum RTCIceConnectionState {
    "new",
    "checking",
    "connected",
    "completed",
    "failed",
    "disconnected",
    "closed"
};
Enumeration description
new Any of the RTCIceTransports are in the new state and none of them are in the checking, failed or disconnected state.
checking Any of the RTCIceTransports are in the checking state and none of them are in the failed or disconnected state.
connected All RTCIceTransports are in the connected, completed or closed state and at least one of them is in the connected state.
completed All RTCIceTransports are in the completed or closed state and at least one of them is in the completed state.
failed Any of the RTCIceTransports are in the failed state.
disconnected Any of the RTCIceTransports are in the disconnected state and none of them are in the failed state.
closed All of the RTCIceTransports are in the closed state.

Note that if an RTCIceTransport is discarded as a result of signaling (e.g. RTCP mux or BUNDLE), or created as a result of signaling (e.g. adding a new media description), the state may advance directly from one state to another.

4.5 Callback Definitions

4.5.1 RTCPeerConnectionErrorCallback

callback RTCPeerConnectionErrorCallback = void (DOMException error);
Callback RTCPeerConnectionErrorCallback Parameters
error of type DOMException
An error object encapsulating information about what went wrong.

4.5.2 RTCSessionDescriptionCallback

callback RTCSessionDescriptionCallback = void (RTCSessionDescription sdp);
Callback RTCSessionDescriptionCallback Parameters
sdp of type RTCSessionDescription
The object containing the SDP [SDP].

4.6 Error Handling

4.6.1 General Principles

All methods that return promises are governed by the standard error handling rules of promises. Methods that do not return promises may throw exceptions to indicate errors.

Legacy-methods may only throw exceptions to indicate invalid state and other programming errors. For example, when a legacy-method is called when the RTCPeerConnection is in an invalid state or a state in which that particular method is not allowed to be executed, it will throw an exception. In all other cases, legacy methods MUST provide an error object to the error callback.

4.7 Session Description Model

4.7.1 RTCSdpType

The RTCSdpType enum describes the type of an RTCSessionDescriptionInit or RTCSessionDescription instance.

enum RTCSdpType {
    "offer",
    "pranswer",
    "answer",
    "rollback"
};
Enumeration description
offer

An RTCSdpType of offer indicates that a description MUST be treated as an [SDP] offer.
pranswer

An RTCSdpType of pranswer indicates that a description MUST be treated as an [SDP] answer, but not a final answer. A description used as an SDP pranswer may be applied as a response to an SDP offer, or an update to a previously sent SDP pranswer.
answer

An RTCSdpType of answer indicates that a description MUST be treated as an [SDP] final answer, and the offer-answer exchange MUST be considered complete. A description used as an SDP answer may be applied as a response to an SDP offer or as an update to a previously sent SDP pranswer.
rollback

An RTCSdpType of rollback indicates that a description MUST be treated as canceling the current SDP negotiation and moving the SDP [SDP] offer and answer back to what it was in the previous stable state. Note the local or remote SDP descriptions in the previous stable state could be null if there has not yet been a successful offer-answer negotiation.

If the mid value of an RTCRtpTransceiver was set to a non-null value by an RTCSessionDescription that is rolled back, the mid value will be set back to null, as defined by [JSEP] (section 4.1.4.2.).

4.7.2 RTCSessionDescription Class

The RTCSessionDescription class is used by RTCPeerConnection to expose local and remote session descriptions. Attributes on this interface are mutable for legacy reasons.

[Constructor(RTCSessionDescriptionInit descriptionInitDict)]
interface RTCSessionDescription {
    readonly attribute RTCSdpType type;
    readonly attribute DOMString  sdp;
    serializer = {attribute};
};
Constructors
RTCSessionDescription
The RTCSessionDescription() constructor takes a dictionary argument, descriptionInitDict, whose content is used to initialize the new RTCSessionDescription object. This constructor is deprecated; it exists for legacy compatibility reasons only.
Parameter Type Nullable Optional Description
descriptionInitDict RTCSessionDescriptionInit ✘ ✘
Attributes
type of type RTCSdpType, readonly
The type of this RTCSessionDescription.
sdp of type DOMString, readonly
The string representation of the SDP [SDP].
Serializer

Instances of this interface are serialized as a map with entries for each of the serializable attributes.

dictionary RTCSessionDescriptionInit {
    required RTCSdpType type;
             DOMString  sdp;
};
Dictionary RTCSessionDescriptionInit Members
type of type RTCSdpType, required
DOMString sdp
sdp of type DOMString
The string representation of the SDP [SDP]; if type is rollback, this member can be left undefined.

4.8 Session Negotiation Model

Many changes to state of an RTCPeerConnection will require communication with the remote side via the signaling channel, in order to have the desired effect. The app can be kept informed as to when it needs to do signaling, by listening to the negotiationneeded event.

4.8.1 Setting Negotiation-Needed

If an operation is performed on an RTCPeerConnection that requires signaling, the connection will be marked as needing negotiation. Examples of such operations include adding or stopping a track, or adding the first data channel.

Internal changes within the implementation can also result in the connection being marked as needing negotiation. For example, if a MediaStreamTrack enters the ended state because its source device became unavailable.

4.8.2 Clearing Negotiation-Needed

The negotiation-needed flag is cleared when setLocalDescription is called (either for an offer or answer), and the supplied description matches the state of the tracks/datachannels that currenly exist on the RTCPeerConnection. Specifically, this means that all live tracks have an associated section in the local description with their MSID, all ended tracks have been removed from the local description, and, if any data channels have been created, a data section exists in the local description.

Note that setLocalDescription(answer) will clear the negotiation-needed flag only if the offer had a corresponding section for all the tracks/datachannels on the answerer side. Otherwise, a new offer by the answerer is still needed, and so the state is not cleared.

4.8.3 Firing An Event

When the RTCPeerConnection connection is marked as negotiation-needed, and it was not already marked as such:

  • If the signaling state is stable, schedule a task to check the negotiation-needed flag and, if still set, fire a negotiationneeded event on connection.
  • Otherwise, do nothing. If necessary, an event will be fired during setLocalDescription or setRemoteDescription processing, as described above.

4.9 Interfaces for Connectivity Establishment

4.9.1 RTCIceCandidate Interface

This interface describes an ICE candidate.

[Constructor(RTCIceCandidateInit candidateInitDict)]
interface RTCIceCandidate {
    readonly attribute DOMString               candidate;
    readonly attribute DOMString?              sdpMid;
    readonly attribute unsigned short?         sdpMLineIndex;
    readonly attribute DOMString               foundation;
    readonly attribute unsigned long           priority;
    readonly attribute DOMString               ip;
    readonly attribute RTCIceProtocol          protocol;
    readonly attribute unsigned short          port;
    readonly attribute RTCIceCandidateType     type;
    readonly attribute RTCIceTcpCandidateType? tcpType;
    readonly attribute DOMString?              relatedAddress;
    readonly attribute unsigned short?         relatedPort;
    serializer = {candidate, sdpMid, sdpMLineIndex};
};
Constructors
RTCIceCandidate
The RTCIceCandidate() constructor takes a dictionary argument, candidateInitDict, whose content is used to initialize the new <<<<<<< 9ec93fec8f9bf5b5964a8308785dc368d818fdd7 RTCIceCandidate object. When run, if both the sdpMid and sdpMLineIndex dictionary members are null, throw a TypeError.
Parameter Type Nullable Optional Description
candidateInitDict RTCIceCandidateInit ✘ ✘
Attributes
candidate of type DOMString, readonly
This carries the candidate-attribute as defined in section 15.1 of [ICE].
sdpMid of type DOMString, readonly , nullable
If not null, this contains the identifier of the "media stream identification" as defined in [RFC5888] for the media component this candidate is associated with.
sdpMLineIndex of type unsigned short, readonly , nullable
If not null, this indicates the index (starting at zero) of the media description in the SDP this candidate is associated with.
foundation of type DOMString, readonly
A unique identifier that allows ICE to correlate candidates that appear on multiple RTCIceTransports.
priority of type unsigned long, readonly
The assigned priority of the candidate.
ip of type DOMString, readonly

The IP address of the candidate.

Note

The IP addresses exposed in candidates gathered via ICE and made visibile to the application in RTCIceCandidate instances can reveal more information about the device and the user (e.g. location, local network topology) than the user might have expected in a non-WebRTC enabled browser.

These IP addresses are always exposed to the application, and potentially exposed to the communicating party, and can be exposed without any specific user consent (e.g. for peer connections used with data channels, or to receive media only).

These IP addresses can also be used as temporary or persistent cross-origin states, and thus contribute to the fingerprinting surface of the device.(This is a fingerprinting vector.)

Applications can avoid exposing IP addresses to the communicating party, either temporarily or permanently, by forcing the ICE Agent to report only relay candidates via the iceTransportPolicy member of RTCConfiguration, or by not signalling non-relay ICE candidates (e.g. until the user has accepted to share media).

To limit the IP addresses exposed to the application itself, browsers can offer their users different policies regarding sharing local IP addresses, as defined in [RTCWEB-IP-HANDLING].

protocol of type RTCIceProtocol, readonly
The protocol of the candidate (udp/tcp).
port of type unsigned short, readonly
The port of the candidate.
type of type RTCIceCandidateType, readonly
The type of the candidate.
tcpType of type RTCIceTcpCandidateType, readonly , nullable
If protocol is tcp, tcpType represents the type of TCP candidate. Otherwise, tcpType is null.
relatedAddress of type DOMString, readonly , nullable
For a candidate that is derived from another, such as a relay or reflexive candidate, the relatedAddress is the IP address of the candidate that it is derived from. For host candidates, the relatedAddress is null.
relatedPort of type unsigned short, readonly , nullable
For a candidate that is derived from another, such as a relay or reflexive candidate, the relatedPort is the port of the candidate that it is derived from. For host candidates, the relatedPort is null.
Serializer

Instances of this interface are serialized as a map with entries for the following attributes: candidate, sdpMid, sdpMLineIndex.

dictionary RTCIceCandidateInit {
    required DOMString       candidate;
             DOMString?      sdpMid = null;
             unsigned short? sdpMLineIndex = null;
};
Dictionary RTCIceCandidateInit Members
candidate of type DOMString, required
sdpMid of type DOMString, nullable, defaulting to null
sdpMLineIndex of type unsigned short, nullable, defaulting to null
4.9.1.1 RTCIceProtocol Enum

The RTCIceProtocol represents the protocol of the ICE candidate.

enum RTCIceProtocol {
    "udp",
    "tcp"
};
Enumeration description
udp A UDP candidate, as described in [ICE].
tcp A TCP candidate, as described in [RFC6544].
4.9.1.2 RTCIceTcpCandidateType Enum

The RTCIceTcpCandidateType represents the type of the ICE TCP candidate, as defined in [RFC6544].

enum RTCIceTcpCandidateType {
    "active",
    "passive",
    "so"
};
Enumeration description
active An active TCP candidate is one for which the transport will attempt to open an outbound connection but will not receive incoming connection requests.
passive A passive TCP candidate is one for which the transport will receive incoming connection attempts but not attempt a connection.
so An so candidate is one for which the transport will attempt to open a connection simultaneously with its peer.
4.9.1.3 RTCIceCandidateType Enum

The RTCIceCandidateType represents the type of the ICE candidate, as defined in [RFC5245] section 15.1.

enum RTCIceCandidateType {
    "host",
    "srflx",
    "prflx",
    "relay"
};
Enumeration description
host A host candidate, as defined in Section 4.1.1.1 of [RFC5245].
srflx A server reflexive candidate, as defined in Section 4.1.1.2 of [RFC5245].
prflx A peer reflexive candidate, as defined in Section 4.1.1.2 of [RFC5245].
relay A relay candidate, as defined in Section 7.1.3.2.1 of [RFC5245].

4.9.2 RTCPeerConnectionIceEvent

The icecandidate event of the RTCPeerConnection uses the RTCPeerConnectionIceEvent interface.

Firing an RTCPeerConnectionIceEvent 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 RTCPeerConnectionIceEvent interface with the candidate attribute set to the new ICE candidate, MUST be created and dispatched at the given target.

When firing an RTCPeerConnectionIceEvent event that contains a RTCIceCandidate object, it MUST include values for both sdpMid and sdpMLineIndex. If the RTCIceCandidate is of type srflx or type relay, the url property of the event MUST be set to the URL of the ICE server from which the candidate was obtained.

[Constructor(DOMString type, RTCPeerConnectionIceEventInit eventInitDict)]
interface RTCPeerConnectionIceEvent : Event {
    readonly attribute RTCIceCandidate? candidate;
    readonly attribute DOMString?       url;
};
Constructors
RTCPeerConnectionIceEvent
Parameter Type Nullable Optional Description
type DOMString ✘ ✘
eventInitDict RTCPeerConnectionIceEventInit ✘ ✘
Attributes
candidate of type RTCIceCandidate, readonly , nullable

The candidate attribute is the RTCIceCandidate object with the new ICE candidate that caused the event.

This attribute is set to null when an event is generated to indicate the end of candidate gathering.

Note

Even where there are multiple media components, only one event containing a null candidate is fired.

url of type DOMString, readonly , nullable

The url attribute is the STUN or TURN URL that identifies the STUN or TURN server used to gather this candidate. If the candidate was not gathered from a STUN or TURN server, this parameter will be set to null.

dictionary RTCPeerConnectionIceEventInit : EventInit {
    RTCIceCandidate candidate;
    DOMString       url;
};
Dictionary RTCPeerConnectionIceEventInit Members
candidate of type RTCIceCandidate

See the candidate attribute of the RTCPeerConnectionIceEvent interface.

url of type DOMString

4.9.3 RTCPeerConnectionIceErrorEvent

The icecandidateerror event of the RTCPeerConnection uses the RTCPeerConnectionIceErrorEvent interface.

[Constructor(DOMString type, RTCPeerConnectionIceErrorEventInit eventInitDict)]
interface RTCPeerConnectionIceErrorEvent : Event {
    readonly attribute DOMString      hostCandidate;
    readonly attribute DOMString      url;
    readonly attribute unsigned short errorCode;
    readonly attribute USVString      errorText;
};
Constructors
RTCPeerConnectionIceErrorEvent
readonly attribute DOMString hostCandidate
Parameter Type Nullable Optional Description
type DOMString ✘ ✘
eventInitDict RTCPeerConnectionIceErrorEventInit ✘ ✘
Attributes
hostCandidate of type DOMString, readonly

The hostCandidate attribute is the local IP address and port used to communicate with the STUN or TURN server.

On a multihomed system, multiple interfaces may be used to contact the server, and this attribute allows the application to figure out on which one the failure occurred.

If use of multiple interfaces has been prohibited for privacy reasons, this attribute will be set to 0.0.0.0:0 or [::]:0, as appropriate.

url of type DOMString, readonly

The url attribute is the STUN or TURN URL that identifies the STUN or TURN server for which the failure occurred.

errorCode of type unsigned short, readonly

The errorCode attribute is the numeric STUN error code returned by the STUN or TURN server [STUN-PARAMETERS].

If no host candidate can reach the server, errorCode will be set to a TBD value in the 7XX range, as this does not conflict with the STUN error code range. This error is only fired once per server URL while in the RTCIceGatheringState of "gathering".

Issue 2

Error code to be defined

errorText of type USVString, readonly

The errorText attribute is the STUN reason text returned by the STUN or TURN server [STUN-PARAMETERS].

If the server could not be reached, errorText will be set to an implementation-specific value providing details about the error.

dictionary RTCPeerConnectionIceErrorEventInit : EventInit {
    DOMString      hostCandidate;
    DOMString      url;
    unsigned short errorCode;
    USVString      statusText;
};
Dictionary RTCPeerConnectionIceErrorEventInit Members
hostCandidate of type DOMString
DOMString url
url of type DOMString
unsigned short errorCode
errorCode of type unsigned short
USVString statusText
statusText of type USVString

4.10 Priority and QoS Model

Many applications have multiple media flows of the same data type and often some of the flows are more important than others. WebRTC uses the priority and Quality of Service (QoS) framework described in [RTCWEB-TRANSPORT] and [TSVWG-RTCWEB-QOS] to provide priority and DSCP marking for packets that will help provide QoS in some networking environments. The priority setting can be used to indicate the relative priority of various flows. The priority API allows the JavaScript applications to tell the browser whether a particular media flow is high, medium, low or of very low importance to the application by setting the priority property of RTCRtpEncodingParameters objects to one of the following values.

4.10.1 RTCPriorityType Enum

enum RTCPriorityType {
    "very-low",
    "low",
    "medium",
    "high"
};
Enumeration description
very-low See [RTCWEB-TRANSPORT], Section 4.
low See [RTCWEB-TRANSPORT], Section 4.
medium See [RTCWEB-TRANSPORT], Section 4.
high See [RTCWEB-TRANSPORT], Section 4.

Applications that use this API should be aware that often better overall user experience is obtained by lowering the priority of things that are not as important rather than raising the priority of the things that are.

4.11 Certificate Management

The certificates that RTCPeerConnection instances use to authenticate with peers use the RTCCertificate interface. These objects can be explicitly generated by applications using the generateCertificate method on the connection and provided in the RTCConfiguration when constructing a new RTCPeerConnection instance.

The explicit certificate management functions provided here are optional. If an application does not provide the certificates configuration option when constructing an RTCPeerConnection a new set of certificates MUST be generated by the user agent. That set MUST include an ECDSA certificate with a private key on the P-256 curve and a signature with a SHA-256 hash.

partial interface RTCPeerConnection {
    static Promise<RTCCertificate> generateCertificate(AlgorithmIdentifier keygenAlgorithm);
};

Methods

generateCertificate, static

The generateCertificate function causes the user agent to create and store an X.509 certificate [X509V3] and corresponding private key. A handle to information is provided in the form of the RTCCertificate interface. The returned RTCCertificate can be used to control the certificate that is offered in the DTLS sessions established by RTCPeerConnection.

The keygenAlgorithm argument is used to control how the private key associated with the certificate is generated. The keygenAlgorithm argument uses the WebCrypto [WebCryptoAPI] AlgorithmIdentifier type. The keygenAlgorithm value MUST be a valid argument to window.crypto.subtle.generateKey; that is, the value MUST produce a non-error result when normalized according to the WebCrypto algorithm normalization process [WebCryptoAPI] with an operation name of generateKey and a [[supportedAlgorithms]] value specific to production of certificates for RTCPeerConnection. If the algorithm normalization process produces an error, the call to generateCertificate MUST be rejected with that error.

Signatures produced by the generated key are used to authenticate the DTLS connection. The identified algorithm (as identified by the name of the normalized AlgorithmIdentifier) MUST be an asymmetric algorithm that can be used to produce a signature.

The certificate produced by this process also contains a signature. The validity of this signature is only relevant for compatibility reasons. Only the public key and the resulting certificate fingerprint are used by RTCPeerConnection, but it is more likely that a certificate will be accepted if the certificate is well formed. The browser selects the algorithm used to sign the certificate; a browser SHOULD select SHA-256 [FIPS-180-4] if a hash algorithm is needed.

The resulting certificate MUST NOT include information that can be linked to a user or user agent. Randomized values for distinguished name and serial number SHOULD be used.

An optional expires attribute MAY be added to the keygenAlgorithm parameter. If this contains a DOMTimeStamp value, it indicates the maximum time that the RTCCertificate is valid for relative to the current time. A user agent sets the expires attribute of the returned RTCCertificate to the current time plus the value of the expires attribute. However, a user agent MAY choose to limit the period over which an RTCCertificate is valid.

A user agent MUST reject a call to generateCertificate() with a DOMException of type NotSupportedError if the keygenAlgorithm parameter identifies an algorithm that the user agent cannot or will not use to generate a certificate for RTCPeerConnection.

The following values MUST be supported by a user agent: { name: "RSASSA-PKCS1-v1_5", modulusLength: 2048, publicExponent: new Uint8Array([1, 0, 1]), hash: "SHA-256" }, and { name: "ECDSA", namedCurve: "P-256" }.

Note

It is expected that a user agent will have a small or even fixed set of values that it will accept.

Parameter Type Nullable Optional Description
keygenAlgorithm AlgorithmIdentifier ✘ ✘
Return type: Promise<RTCCertificate>

4.11.1 RTCCertificate Interface

The RTCCertificate interface represents a certificate used to authenticate WebRTC communications. In addition to the visible properties, internal slots contain a handle to the generated private keying materal ([[handle]]) and a certificate ([[certificate]]) that RTCPeerConnection uses to authenticate with a peer.

interface RTCCertificate {
    readonly attribute DOMTimeStamp expires;
    AlgorithmIdentifier getAlgorithm();
};
Attributes
expires of type DOMTimeStamp, readonly

The expires attribute indicates the date and time in milliseconds relative to 1970-01-01T00:00:00Z after which the certificate will be considered invalid by the browser. After this time, attempts to construct an RTCPeerConnection using this certificate fail.

Note that this value might not be reflected in a notAfter parameter in the certificate itself.

Methods
getAlgorithm

Returns the value of keygenAlgorithm passed in the call to generateCertificate().

No parameters.
Return type: AlgorithmIdentifier

For the purposes of this API, the [[certificate]] slot contains unstructured binary data.

Note that a RTCCertificate might not directly hold private keying material, this might be stored in a secure module.

The RTCCertificate object can be stored and retrieved from persistent storage by an application. When a user agent is required to obtain a structured clone [HTML] of a RTCCertificate object, it performs the following steps:

  1. Let input and memory be the corresponding inputs defined by the internal structured cloning algorithm, where input represents a RTCCertificate object to be cloned.
  2. Let output be a newly constructed RTCCertificate object.
  3. Copy the value of the expires attribute from input to output.
  4. Let the [[certificate]] internal slot of output be set to the result of invoking the internal structured clone algorithm recursively on the corresponding internal slots of input, with the slot contents as the new "input" argument and memory as the new "memory" argument.
  5. Let the [[handle]] internal slot of output refer to the same private keying material represented by the [[handle]] internal slot of input.

5. RTP Media API

The RTP media API lets a web application send and receive MediaStreamTracks over a peer-to-peer connection. Tracks, when added to a RTCPeerConnection, result in signaling; when this signaling is forwarded to a remote peer, it causes corresponding tracks to be created on the remote side.

The actual encoding and transmission of MediaStreamTracks is managed through objects called RTCRtpSenders. Similarly, the reception and decoding of MediaStreamTracks is managed through objects called RTCRtpReceivers. Each track to be sent is associated with exactly one RTCRtpSender, and each track to be received is associated with exactly one RTCRtpReceiver.

The encoding and transmission of each MediaStreamTrack SHOULD be made such that its characteristics (width, height and frameRate for video tracks; volume, sampleSize, sampleRate and channelCount for audio tracks) are to a reasonable degree retained by the track created on the remote side. There are situations when this does not apply, there may for example be resource constraints at either endpoint or in the network or there may be RTCRtpSender settings applied that instruct the implementation to act differently.

RTCRtpSenders are created when the application attaches a MediaStreamTrack to a RTCPeerConnection, via the addTrack method. RTCRtpReceivers, on the other hand, are created when remote signaling indicates new tracks are available, and each new MediaStreamTrack and its associated RTCRtpReceiver are surfaced to the application via the ontrack event. Both RTCRtpSender and RTCRtpReceiver objects are created by the addTransceiver method.

A RTCPeerConnection object contains a set of RTCRtpSenders, representing tracks to be sent, and a set of RTCRtpReceivers, representing tracks that are to be received on this RTCPeerConnection object, and a set of RTCRtpTransceivers, representing the paired senders and receiver with some shared state. All of these sets are initialized to empty sets when the RTCPeerConnection object is created.

5.1 RTCPeerConnection Interface Extensions

The RTP media API extends the RTCPeerConnection interface as described below.

partial interface RTCPeerConnection {
    sequence<RTCRtpSender>      getSenders();
    sequence<RTCRtpReceiver>    getReceivers();
    sequence<RTCRtpTransceiver> getTransceivers();
    RTCRtpSender                addTrack(MediaStreamTrack track,
                                         MediaStream... streams);
    void                        removeTrack(RTCRtpSender sender);
    RTCRtpTransceiver           addTransceiver((MediaStreamTrack or DOMString) trackOrKind,
                                               RTCRtpTransceiverInit init);
    attribute EventHandler ontrack;
};

Attributes

ontrack of type EventHandler

The event type of this event handler is track.

Methods

getSenders

Returns a sequence of RTCRtpSender objects representing the RTP senders that are currently attached to this RTCPeerConnection object.

The getSenders method MUST return a new sequence that represents a snapshot of all the RTCRtpSender objects in this RTCPeerConnection object's set of senders. The conversion from the senders set to the sequence, to be returned, is user agent defined and the order does not have to be stable between calls.

No parameters.
Return type: sequence<RTCRtpSender>
getReceivers

Returns a sequence of RTCRtpReceiver objects representing the RTP receivers that are currently attached to this RTCPeerConnection object.

The getReceivers method MUST return a new sequence that represents a snapshot of all the RTCRtpReceiver objects in this RTCPeerConnection object's set of receivers. The conversion from the receivers set to the sequence, to be returned, is user agent defined and the order does not have to be stable between calls.

No parameters.
Return type: sequence<RTCRtpReceiver>
getTransceivers

Returns a sequence of RTCRtpTransceiver objects representing the RTP transceivers that are currently attached to this RTCPeerConnection object.

The getTransceivers method MUST return a new sequence that represents a snapshot of all the RTCRtpTransceiver objects in this RTCPeerConnection object's set of transceivers. The conversion from the transceiver set to the sequence, to be returned, is user agent defined and the order does not have to be stable between calls.

No parameters.
Return type: sequence<RTCRtpTransceiver>
addTrack

Adds a new track to the RTCPeerConnection, and indicates that it is contained in the specified MediaStreams.

When the addTrack method is invoked, the user agent MUST run the following steps:

  1. Let connection be the RTCPeerConnection object on which the MediaStreamTrack track, is to be added.

  2. If connection's [[isClosed]] slot is true, throw an InvalidStateError exception and abort these steps.

  3. If an RTCRtpSender for track already exists in connection's set of senders, throw an InvalidAccessError exception and abort these steps.

  4. The steps below describe how to determine if an existing sender can be reused. Doing so will cause future calls to createOffer and createAnswer to mark the corresponding media description as sendrecv or sendonly, as defined in [JSEP] (section 5.2.2.).

    If any RTCRtpSender object, in connection's set of senders matches the following criteria, let sender be that object:

    • The sender's track is null.

    • The transceiver kind of the RTCRtpTransceiver, associated with the sender, matches track's kind.

    • The sender has never been used to send. More precisely, the RTCRtpTransceiver associated with the sender has always had a direction of either recvonly or inactive.

  5. If sender was left unset by the previous step, create a new RTCRtpTransceiver with sender.track set to track, add it to connection's set of transceivers.

    Let sender be the RTCRtpSender object associated with the newly created RTCRtpTransceiver.

    The initial value of an RTCRtpTransceiver's mid attribute is null. Setting a new RTCSessionDescription may change it to a non-null value, as defined in [JSEP] (section 5.4. and section 5.5.).

  6. A track could have contents that are inaccessible to the application. This can be due to being marked with a peerIdentity option or anything that would make a track CORS cross-origin. These tracks can be supplied to the addTrack method, and have an RTCRtpSender created for them, but content MUST NOT be transmitted, unless they are also marked with peerIdentity and they meet the requirements for sending (see isolated streams and RTCPeerConnection).

    All other tracks that are not accessible to the application MUST NOT be sent to the peer, with silence (audio), black frames (video) or equivalently absent content being sent in place of track content.

    Note that this property can change over time.

  7. Mark connection as needing negotiation.

  8. Return sender.

Parameter Type Nullable Optional Description
track MediaStreamTrack ✘ ✘
streams MediaStream ✘ ✘
Return type: RTCRtpSender
removeTrack

Stops sending media from sender. The RTCRtpSender will still appear in getSenders. Doing so will cause future calls to createOffer to mark the media description for the corresponding transceiver as recvonly or inactive, as defined in [JSEP] (section 5.2.2.).

When the other peer stops sending a track in this manner, an ended event is fired at the MediaStreamTrack object.

When the removeTrack method is invoked, the user agent MUST run the following steps:

  1. Let connection be the RTCPeerConnection object on which the RTCRtpSender, sender, is to be stopped.

  2. If connection's [[isClosed]] slot is true, throw an InvalidStateError exception and abort these steps.

  3. If sender is stopped or not in connection's set of senders, then abort these steps.

  4. Stop sender.

  5. Mark connection as needing negotiation.

Parameter Type Nullable Optional Description
sender RTCRtpSender ✘ ✘
Return type: void
addTransceiver

Create a new RTCRtpTransceiver and add it to the collection of transceivers that will be returned by getTransceivers.

Adding a transceiver will cause future calls to createOffer to add a media description for the corresponding transceiver, as defined in [JSEP] (section 5.2.2.).

The initial value of mid is null. Setting a new RTCSessionDescription may change it to a non-null value, as defined in [JSEP] (section 5.4. and section 5.5.).

If a track is passed in, the value of the sender.track will be set to that track and the MSID and media type generated by createOffer will be that of the track.

If a kind is passed in and the value is not a legal MediaStreamTrack kind, throw a TypeError.

If a kind is passed in, the value of the sender.track will be null and and media type generated by createOffer will be that of the kind. The MSID generated by createOffer (if necessary, such as when init.send == true) will be selected by the user agent and will not be related to any track. Future calls to sender.replaceTrack with a track of a different kind will fail. Future calls will not change the MSID associated with the transceiver.

If init.sendEncodings is set, then subsequent calls to createOffer will be configured to send with multiple RTP encodings as defined in [JSEP] (section 5.2.2.). When setRemoteDescription is called with a corresponding remote description that is able to receive multiple RTP encodings as defined in [JSEP], the RTCRtpSender may send multiple RTP encodings and the parameters retrieved via the transceiver’s sender.getParameters() will reflect the encodings negotiated.

RID values passed into init.sendEncodings must be composed only of case-sensitive alphanumeric characters (a-z, A-Z, 0-9) up to a maximum of 16 characters.

Parameter Type Nullable Optional Description
trackOrKind (MediaStreamTrack or DOMString) ✘ ✘
init RTCRtpTransceiverInit ✘ ✘
Return type: RTCRtpTransceiver 
dictionary RTCRtpTransceiverInit {
    RTCRtpTransceiverDirection         direction = "sendrecv";
    sequence<MediaStream>              streams;
    sequence<RTCRtpEncodingParameters> sendEncodings;
};

Dictionary RTCRtpTransceiverInit Members

direction of type RTCRtpTransceiverDirection, defaulting to "sendrecv"
The direction of the RTCRtpTransceiver.
streams of type sequence<MediaStream>

When the remote PeerConnection's ontrack event fires corresponding to the RTCRtpReceiver being added, these are the streams that will be put in the event.

sendEncodings of type sequence<RTCRtpEncodingParameters>

A sequence containing parameters for sending RTP encodings of media.

enum RTCRtpTransceiverDirection {
    "sendrecv",
    "sendonly",
    "recvonly",
    "inactive"
};
Enumeration description
sendrecv The RTCRtpTransceiver's RTCRtpSender will offer to send RTP, and will send RTP if the remote peer accepts, in which case active is set to "true". The RTCRtpTransceiver's RTCRtpReceiver will offer to receive RTP, and will receive RTP if the remote peer accepts, in which case active is set to "true".
sendonly The RTCRtpTransceiver's RTCRtpSender will offer to send RTP, and will send RTP if the remote peer accepts, in which case active is set to "true". The RTCRtpTransceiver's RTCRtpReceiver will not offer to receive RTP, and will not receive RTP (active set to "false").
recvonly The RTCRtpTransceiver's RTCRtpSender will not offer to send RTP, and will not send RTP (active set to "false"). The RTCRtpTransceiver's RTCRtpReceiver will offer to receive RTP, and will receive RTP if the remote peer accepts, in which case active is set to "true".
inactive The RTCRtpTransceiver's RTCRtpSender will not offer to send RTP, and will not send RTP (active set to "false"). The RTCRtpTransceiver's RTCRtpReceiver will not offer to receive RTP, and will not receive RTP (active set to "false").

5.1.1 Processing Remote MediaStreamTracks

Rejection of incoming MediaStreamTrack objects can be done by the application, after receiving the track, by stopping it.

To dispatch a receiver for an incoming media description [JSEP], the user agent MUST run the following steps:

  1. Let connection be the RTCPeerConnection expecting this media.

  2. If connection's [[isClosed]] slot is true, abort these steps.

  3. Let streams be a list of MediaStream objects that the sender indicated the sent MediaStreamTrack being a part of. The information needed to collect these objects is part of the media description.

  4. Run the following steps to create a track representing the incoming media description:

    1. Create a MediaStreamTrack object track to represent the media description. The source of track is a remote source provided by connection.

    2. Initialize track.kind attribute to audio or video depending on the media type of the media description.

    3. Initialize track.id attribute to the media description track id.

    4. Initialize track.label attribute to remote audio or remote video depending on the media type of the media description.

    5. Initialize track.readyState attribute to live.

    6. Initialize track.muted attribute to true. See the MediaStreamTrack section about how the muted attribute reflects if a MediaStreamTrack is receiving media data or not.

  5. Add track to all MediaStream objects in streams.

    Note

    This will result in an addtrack event being fired at each MediaStream as described in [GETUSERMEDIA].

  6. Create a new RTCRtpReceiver object receiver for track, and add it to connection's set of receivers.

  7. Fire an event named track with transceiver, track, and streams at the connection object.

When an RTCPeerConnection finds that a track from the remote peer has been removed, the user agent MUST follow these steps:

  1. Let connection be the RTCPeerConnection associated with the track being removed.

  2. Let track be the MediaStreamTrack object that represents the track being removed, if any. If there isn't one, then abort these steps.

  3. By definition, track is now ended.

    Note

    A task is thus queued to update track and fire an event.

  4. Queue a task to run the following substeps:

    1. If connection's [[isClosed]] slot is true, abort these steps.

    2. Remove the RTCRtpReceiver associated with track from connection's set of receivers.

Issue 3

Since the beginning of this specification, remote MediaStreamTracks have been created by the setRemoteDescription call, one track for each non-rejected media description in the remote description. This meant that at the caller, MediaStreamTracks were not created until the answer was received, and any media received prior to a remote description (AKA "early media") would be discarded. If any form of remote description is provided (either an answer or a pranswer), this issue does not occur.

If we want to allow early media to be played out, minor changes are necessary. Fundamentally, we would need to change when tracks are created for the offerer; this would have to happen either as a result of setLocalDescription, or when media packets are received. This ensures that these objects can be created and connected to media elements for playout.

However, there are three consequences to this potential change:

  1. Media may arrive before the remote DTLS fingerprint has been received. This means that media could be played out before the validity of the DTLS fingerprint has been established, which may be hard to explain to users. As such, it is recommended that media not be played out unless some TBD RTCConfiguration property (e.g. AllowUnverifiedMedia) has been set.
  2. The information needed to correlate MediaStreamTracks with their enclosing MediaStreams will not yet be present when the tracks are initially generated. Therefore, the implementation will need to create dummy MediaStream objects for each MediaStreamTrack, and then possibly change the associated MediaStream for each track when the remote description is received (e.g. if it indicates that an audio and video MediaStreamTrack should be combined into a single MediaStream). Since media elements act on MediaStreams, some complex reshuffling may need to occur when the remote description is received.
  3. The track events fired and their timing will change. For the offerer, ontrack will now fire during setLocalDescription, once for each track being offered, and track.onended will fire during setRemoteDescription for any offered tracks that were rejected. For the answerer, ontrack will continue to fire during setRemoteDescription, as it does today (this is necessary to allow the answerer to reject offered tracks by stopping them).

For now, we simply make note of this issue, until it can be considered fully by the WG.

5.2 RTCRtpSender Interface

The RTCRtpSender interface allows an application to control how a given MediaStreamTrack is encoded and transmitted to a remote peer. When setParameters is called on an RTCRtpSender object, the encoding is changed appropriately.

An RTCRtpSender can be stopped, which indicates that it will no longer send any media.

interface RTCRtpSender {
    readonly attribute MediaStreamTrack? track;
    readonly attribute RTCDtlsTransport  transport;
    readonly attribute RTCDtlsTransport? rtcpTransport;
    static RTCRtpCapabilities getCapabilities(DOMString kind);
    Promise<void>      setParameters(optional RTCRtpParameters parameters);
    RTCRtpParameters   getParameters();
    Promise<void>      replaceTrack(MediaStreamTrack withTrack);
};

Attributes

track of type MediaStreamTrack, readonly , nullable

The track attribute is the track that is associated with this RTCRtpSender object.

transport of type RTCDtlsTransport, readonly

The transport attribute is the transport over which media from track is sent in the form of RTP packets. When BUNDLE is used, many RTCRtpSender objects will share one transport and will all send RTP over the same transport. When RTCP mux is used, rtcpTransport will be null, and both RTP and RTCP traffic will flow over the transport described by transport.

rtcpTransport of type RTCDtlsTransport, readonly , nullable

The rtcpTransport attribute is the transport over which RTCP is sent and received. When BUNDLE is used, many RTCRtpSender objects will share one rtcpTransport and will all send and receive RTCP over the same transport. When RTCP mux is used, rtcpTransport will be null, and both RTP and RTCP traffic will flow over the transport described by transport.

Methods

getCapabilities, static

The RTCRtpSender.getCapabilities method returns the most optimist view on the capabilities of the system for sending media of the given kind. It does not reserve any resources, ports, or other state but is meant to provide a way to discover the types of capabilities of the browser including which codecs may be supported.

These capabilities provide generally persistent cross-origin information on the device and thus increases the fingerprinting surface of the application. In privacy-sensitive contexts, browsers can consider mitigations such as reporting only a common subset of the capabilities.(This is a fingerprinting vector.)

Parameter Type Nullable Optional Description
kind DOMString ✘ ✘
Return type: RTCRtpCapabilities
setParameters

The setParameters method updates how track is encoded and transmitted to a remote peer.

When the setParameters method is called, the user agent MUST run the following steps:

  1. Let sender be the RTCRtpSender object on which setParameters is invoked.
  2. Let p be a new promise.
  3. Return p, performing the next steps in parallel.
  4. If any parameter in the parameters argument, marked as a Read-only parameter, has a value that is different from the corresponding parameter value returned from getParameters(), abort these steps and return a promise rejected with InvalidModificationError. Note that this also applies to transactionId.
  5. If the scaleResolutionDownBy parameter in the parameters argument has a value less than 1.0, abort these steps and return a promise rejected with RangeError.
  6. Set the RTCRtpSender's internal transactionId slot to a previously unused value.
  7. Resolve p with undefined.

If codecs are reordered, the new order indicates the preference for sending, with the first codec being the codec with highest preference. If a codec is removed, that codec will not be used to send. The effect of reordering or removing codecs lasts until the codecs are renegotiated. After the codecs are renegotiated, they are set to the value negotiated, and setParameters needs to be called again to re-apply codec preferences.

setParameters does not cause SDP renegotiation and can only be used to change what the media stack is sending or receiving within the envelope negotiated by Offer/Answer. The attributes in the RTCRtpParameters dictionary are designed to not enable this, so attributes like ssrc that cannot be changed are read only. Other things, like bitrate, are controlled using limits such as maxBitrate, where the User Agent needs to ensure it does not exceed the maximum bitrate specified by maxBitrate, while at the same time making sure it satisfies constraints on bitrate specified in other places such as the SDP.

Parameter Type Nullable Optional Description
parameters RTCRtpParameters ✘ ✔
Return type: Promise<void>
getParameters

The getParameters method returns the RTCRtpSender object's current parameters for how track is encoded and transmitted to a remote RTCRtpReceiver. It may used with setParameters to change the parameters in the following way:

Example 2
var params = sender.getParameters();
// ... make changes to RTCRtpParameters
params.encodings[0].active = false;
sender.setParameters(params)
No parameters.
Return type: RTCRtpParameters
replaceTrack

Attempts to replace the track being sent with another track provided, without renegotiation.

To avoid track identifiers changing on the remote receiving end when a track is replaced, the sender MUST retain the original track identifier and stream associations and use these in subsequent negotiations.

When the replaceTrack method is invoked, the user agent MUST run the following steps:

  1. Let sender be the RTCRtpSender object on which replaceTrack is invoked.

  2. Let connection be the RTCPeerConnection object that created sender.

  3. If sender is stopped, return a promise rejected with an InvalidStateError.

  4. Let withTrack be the argument to this method.

  5. Let transceiver be the RTCRtpTransceiver object associated with sender.

  6. If withTrack.kind differs from the transceiver kind of transceiver, return a promise rejected with a TypeError.

  7. If transceiver is not yet associated with a media description [JSEP] (section 3.4.1.), then set sender's track attribute to withTrack, and return a promise resolved with undefined.

  8. Let p be a new promise.

  9. Run the following steps in parallel:

    1. Determine if negotiation is needed to transmit withTrack in place of the sender's existing track. Ignore which MediaStream the track resides in and the id attribute of the track in this determination. If negotiation is needed, then reject p with InvalidModificationError and abort these steps.

    2. Have the sender switch seamlessly to transmitting withTrack in place of what it is sending, without negotiating.

    3. Queue a task that sets sender's track attribute to withTrack, and resolves p with undefined.

  10. Return p.

Note

Changing dimensions and/or frame rates might not require negotiation. Cases that may require negotiation include:

  1. Changing a resolution to a value outside of the negotiated imageattr bounds, as described in [RFC6236].
  2. Changing a frame rate to a value that causes the block rate for the codec to be exceeded.
  3. A video track differing in raw vs. pre-encoded format.
  4. An audio track having a different number of channels.
  5. Sources that also encode (typically hardware encoders) might be unable to produce the negotiated codec; similarly, software sources might not implement the codec that was negotiated for an encoding source.
Parameter Type Nullable Optional Description
withTrack MediaStreamTrack ✘ ✘
Return type: Promise<void> 
dictionary RTCRtpParameters {
    DOMString                                 transactionId;
    sequence<RTCRtpEncodingParameters>        encodings;
    sequence<RTCRtpHeaderExtensionParameters> headerExtensions;
    RTCRtcpParameters                         rtcp;
    sequence<RTCRtpCodecParameters>           codecs;
    RTCDegradationPreference                  degradationPreference = "balanced";
};

Dictionary RTCRtpParameters Members

transactionId of type DOMString

An unique identifier for the last set of parameters applied. Ensures that setParameters can only be called based on a previous getParameters, and that there are no intervening changes.

encodings of type sequence<RTCRtpEncodingParameters>

A sequence containing parameters for RTP encodings of media.

headerExtensions of type sequence<RTCRtpHeaderExtensionParameters>

A sequence containing parameters for RTP header extensions.

rtcp of type RTCRtcpParameters

Parameters used for RTCP.

codecs of type sequence<RTCRtpCodecParameters>

A sequence containing the codecs that an RTCRtpSender will choose from in order to send media.

degradationPreference of type RTCDegradationPreference, defaulting to "balanced"

When bandwidth is constrained and the RtpSender needs to choose between degrading resolution or degrading framerate, degradationPreference indicates which is preferred.

dictionary RTCRtpEncodingParameters {
    unsigned long       ssrc;
    RTCRtpRtxParameters rtx;
    RTCRtpFecParameters fec;
    boolean             active;
    RTCPriorityType     priority;
    unsigned long       maxBitrate;
    unsigned long       maxFramerate;
    DOMString           rid;
    double              scaleResolutionDownBy = 1;
};

Dictionary RTCRtpEncodingParameters Members

ssrc of type unsigned long

The SSRC of the RTP source stream of this encoding (non-RTX, non-FEC RTP stream). Read-only parameter.

rtx of type RTCRtpRtxParameters

The parameters used for RTX, or unset if RTX is not being used.

fec of type RTCRtpFecParameters

The parameters used for FEC, or unset if FEC is not being used.

active of type boolean

For an RTCRtpSender, indicates that this encoding is actively being sent. Setting it to false causes this encoding to no longer be sent. Setting it to true causes this encoding to be sent. For an RTCRtpReceiver, indicates that this encoding is being decoded. Setting it to false causes this encoding to no longer be decoded. Setting it to true causes this encoding to be decoded.

priority of type RTCPriorityType

Indicates the priority of this encoding. It is specified in [RTCWEB-TRANSPORT], Section 4.

maxBitrate of type unsigned long

Indicates the maximum bitrate that can be used to send this encoding. The encoding may also be further constrained by other limits (such as maxFramerate or per-transport or per-session bandwidth limits) below the maximum specified here. maxBitrate is the Transport Independent Application Specific Maximum (TIAS) bandwidth defined in [RFC3890] Section 6.2.2, which is the maximum bandwidth needed without counting IP or other transport layers like TCP or UDP.

maxFramerate of type unsigned long

Indicates the maximum framerate that can be used to send this encoding.

rid of type DOMString

If set, this RTP encoding will be sent with the RID header extension as defined by [JSEP]. The RID is not modifiable via setParameters. It can only be set or modified in addTransceiver or addTrack.

scaleResolutionDownBy of type double, defaulting to 1.0

If the sender's kind is "video", the video's resolution will be scaled down in each dimension by the given value before sending. For example, if the value is 2.0, the video will be scaled down by a factor of 2 in each dimension, resulting in sending a video of one quarter the size. If the value is 1.0, the video will not be affected. The value must be greater than or equal to 1.0.

enum RTCDegradationPreference {
    "maintain-framerate",
    "maintain-resolution",
    "balanced"
};
Enumeration description
maintain-framerate

Degrade resolution in order to maintain framerate.
maintain-resolution

Degrade framerate in order to maintain resolution.
balanced

Degrade a balance of framerate and resolution.

 
dictionary RTCRtpRtxParameters {
    unsigned long ssrc;
};

Dictionary RTCRtpRtxParameters Members

ssrc of type unsigned long

The SSRC of the RTP stream for RTX. Read-only parameter.

dictionary RTCRtpFecParameters {
    unsigned long ssrc;
};

Dictionary RTCRtpFecParameters Members

ssrc of type unsigned long

The SSRC of the RTP stream for FEC. Read-only parameter.

dictionary RTCRtcpParameters {
    DOMString cname;
    boolean   reducedSize;
};

Dictionary RTCRtcpParameters Members

cname of type DOMString

The Canonical Name (CNAME) used by RTCP (e.g. in SDES messages). Read-only parameter.

reducedSize of type boolean

Whether reduced size RTCP [RFC5506] is configured (if true) or compound RTCP as specified in [RFC3550] (if false). Read-only parameter.

dictionary RTCRtpHeaderExtensionParameters {
    DOMString      uri;
    unsigned short id;
    boolean        encrypted;
};

Dictionary RTCRtpHeaderExtensionParameters Members

uri of type DOMString

The URI of the RTP header extension, as defined in [RFC5285]. Read-only parameter.

id of type unsigned short

The value put in the RTP packet to identify the header extension. Read-only parameter.

encrypted of type boolean

Whether the header extension is encryted or not. Read-only parameter.

dictionary RTCRtpCodecParameters {
    unsigned short payloadType;
    DOMString      mimeType;
    unsigned long  clockRate;
    unsigned short channels = 1;
    DOMString      sdpFmtpLine;
};

Dictionary RTCRtpCodecParameters Members

payloadType of type unsigned short

The RTP payload type. This value can be set to control which codec should be used to send a given encoding.

mimeType of type DOMString

The codec MIME type. Valid types are listed in [IANA-RTP-2].

clockRate of type unsigned long

The codec clock rate expressed in Hertz.

channels of type unsigned short, defaulting to 1

The number of channels (mono=1, stereo=2).

sdpFmtpLine of type DOMString

The a=fmtp line in the SDP corresponding to the codec, as defined by [JSEP] (section 5.6.).

dictionary RTCRtpCapabilities {
    sequence<RTCRtpCodecCapability>           codecs;
    sequence<RTCRtpHeaderExtensionCapability> headerExtensions;
};

Dictionary RTCRtpCapabilities Members

codecs of type sequence<RTCRtpCodecCapability>

Supported codecs.

headerExtensions of type sequence<RTCRtpHeaderExtensionCapability>

Supported RTP header extensions.

dictionary RTCRtpCodecCapability {
    DOMString mimeType;
};

Dictionary RTCRtpCodecCapability Members

mimeType of type DOMString

The codec MIME type. Valid types are listed in [IANA-RTP-2].

dictionary RTCRtpHeaderExtensionCapability {
    DOMString uri;
};

Dictionary RTCRtpHeaderExtensionCapability Members

uri of type DOMString

The URI of the RTP header extension, as defined in [RFC5285].

5.3 RTCRtpReceiver Interface

The RTCRtpReceiver interface allows an application to control the receipt of a MediaStreamTrack. When attributes on an RTCRtpReceiver are modified, a negotiation is triggered to signal the changes regarding what the application wants to receive to the other side.

interface RTCRtpReceiver {
    readonly attribute MediaStreamTrack  track;
    readonly attribute RTCDtlsTransport  transport;
    readonly attribute RTCDtlsTransport? rtcpTransport;
    static RTCRtpCapabilities          getCapabilities(DOMString kind);
    RTCRtpParameters                   getParameters();
    sequence<RTCRtpContributingSource> getContributingSources();
};

Attributes

track of type MediaStreamTrack, readonly

The track attribute is the track that is immutably associated with this RTCRtpReceiver object.

transport of type RTCDtlsTransport, readonly

The transport attribute is the transport over which media for the receiver's track is received in the form of RTP packets. When BUNDLE is used, many RTCRtpReceiver objects will share one transport and will all receive RTP over the same transport. When RTCP mux is used, rtcpTransport will be null, and both RTP and RTCP traffic will flow over transport.

rtcpTransport of type RTCDtlsTransport, readonly , nullable

The rtcpTransport attribute is the transport over which RTCP is sent and received. When BUNDLE is used, many RTCRtpReceiver objects will share one RTCRtpReceiver.rtcpTransport and will all send and receive RTCP over the same transport. When RTCP mux is used, rtcpTransport will be null, and both RTP and RTCP traffic will flow over transport.

Methods

getCapabilities, static

The RTCRtpReceiver.getCapabilities method returns the most optimistic view of the capabilities of the system for receiving media of the given kind. It does not reserve any resources, ports, or other state but is meant to provide a way to discover the types of capabilities of the browser including which codecs may be supported.

These capabilities provide generally persistent cross-origin information on the device and thus increases the fingerprinting surface of the application. In privacy-sensitive contexts, browsers can consider mitigations such as reporting only a common subset of the capabilities.(This is a fingerprinting vector.)

Parameter Type Nullable Optional Description
kind DOMString ✘ ✘
Return type: RTCRtpCapabilities
getParameters

The getParameters method returns the RTCRtpReceiver object's current parameters for how track is decoded.

No parameters.
Return type: RTCRtpParameters
getContributingSources

Returns an RTCRtpContributingSource for each unique CSRC or SSRC received by this RTCRtpReceiver in the last 10 seconds.

No parameters.
Return type: sequence<RTCRtpContributingSource>

The RTCRtpContributingSource objects contain information about a given contributing source, including the time the most recent time a packet was received from the source. The browser MUST keep information from RTP packets received in the previous 10 seconds. Each time an RTP packet is received, the RTCRtpContributingSource objects are updated. If the RTP packet contains CSRCs, then the RTCRtpContributingSource objects corresponding to those CSRCs are updated. If the RTP packet contains no CSRCs, then the RTCRtpContributingSource object corresponding to the SSRC is updated.

interface RTCRtpContributingSource {
    readonly attribute DOMHighResTimeStamp timestamp;
    readonly attribute unsigned long       source;
    readonly attribute byte?               audioLevel;
    readonly attribute boolean?            voiceActivityFlag;
};

Attributes

timestamp of type DOMHighResTimeStamp, readonly

The timestamp of type DOMHighResTimeStamp [HIGHRES-TIME], indicating the time of reception of the most recent RTP packet containing the source. The timestamp is defined in [HIGHRES-TIME] and corresponds to a local clock.

source of type unsigned long, readonly

The CSRC or SSRC value of the contributing source.

audioLevel of type byte, readonly , nullable

The audio level contained in the last RTP packet received from this source. If the source was set from an SSRC, this will be the level value defined in [RFC6464]. If an RFC 6464 extension header is not present, the browser will compute the value as if it had come from RFC 6464 and use that. If the source was set from a CSRC, this will be the level value defined in [RFC6465]. RFC 6464 and 6465 define the level as a integral value from 0 to 127 representing the audio level in negative decibels relative to the loudest signal that they system could possibly encode. Thus, 0 represents the loudest signal the system could possibly encode, and 127 represents silence.

voiceActivityFlag of type boolean, readonly , nullable

Whether the last RTP packet received from this source contains voice activity (true) or not (false). Since the "V" bit is supported in [RFC6464] but not [RFC6465], the voiceActivityFlag attribute will only be set for RTP packets received from peers enabling the client-mixer header extension with the "vad" extension set to "on".

5.4 RTCRtpTransceiver Interface

The RTCRtpTransceiver interface represents a combination of an RTCRtpSender and an RTCRtpReceiver that share a common mid.

The transceiver kind of an RTCRtpTransceiver is defined by the kind of the associated RTCRtpReceiver's MediaStreamTrack object.

interface RTCRtpTransceiver {
    readonly attribute DOMString?                 mid;
    [SameObject]
    readonly attribute RTCRtpSender               sender;
    [SameObject]
    readonly attribute RTCRtpReceiver             receiver;
    readonly attribute boolean                    stopped;
    readonly attribute RTCRtpTransceiverDirection direction;
    void setDirection(RTCRtpTransceiverDirection direction);
    void stop();
    void setCodecPreferences(sequence<RTCRtpCodecCapability> codecs);
};

Attributes

mid of type DOMString, readonly , nullable

The mid attribute is the mid negotatiated and present in the local and remote descriptions as defined in [JSEP]. Before negotiation is complete, the mid value may be null. If there is no MID value in the remote SDP, and no MID value was previously assigned, a random value will be created for the mid as described in [JSEP] (section 3.5.2.1.) when the remote SDP is set. After rollbacks, the value may change from a non-null value to null.

sender of type RTCRtpSender, readonly

The sender attribute is the RTCRtpSender corresponding to the RTP media that may be sent with mid = mid.

receiver of type RTCRtpReceiver, readonly

The receiver attribute is the RTCRtpReceiver corresponding to the RTP media that may be received with mid = mid.

stopped of type boolean, readonly

The stopped attribute indicates that the sender of this transceiver will no longer send, and that the receiver will no longer receive. It is true if either stop has been called or if setting the local or remote description has caused the RTCRtpReceiver to be stopped.

direction of type RTCRtpTransceiverDirection, readonly

The direction attribute indicates the direction of this transceiver. The value of direction is independent of the value of encodings[].active since one cannot be deduced from the other. If the stop() method is called, direction retains the value it had prior to calling stop().

Methods

setDirection

The setDirection method sets the direction of the RTCRtpTransceiver. Future calls to createOffer and createAnswer mark the corresponding media description as sendrecv, sendonly, recvonly or inactive as defined in [JSEP] (section 5.2.2. and section 5.3.2.). Calling setDirection() sets the negotiation-needed flag.

Parameter Type Nullable Optional Description
direction RTCRtpTransceiverDirection ✘ ✘
Return type: void
stop

The stop method stops the RTCRtpTransceiver. The sender of this transceiver will no longer send, and the receiver will no longer receive.

No parameters.
Return type: void
setCodecPreferences

The setCodecPreferences method overrides the default codec preferences used by the user agent. When generating a session description using either createOffer or createAnswer, the user agent MUST use the indicated codecs, in the order specified in the codecs argument, for the media section corresponding to this RTCRtpTransceiver. Note that calls to createAnswer will use only the common subset of these codecs and the codecs that appear in the offer.

This method allows applications to disable the negotiation of specific codecs. It also allows an application to cause a remote peer to prefer the codec that appears first in the list for sending.

Codec preferences remain in effect for all calls to createOffer and createAnswer that include this RTCRtpTransceiver until this method is called again. Setting codecs to an empty sequence resets codec preferences to any default value.

Parameter Type Nullable Optional Description
codecs sequence<RTCRtpCodecCapability> ✘ ✘
Return type: void

5.5 RTCDtlsTransport Interface

The RTCDtlsTransport interface allows an application access to information about the Datagram Transport Layer Security (DTLS) transport over which RTP and RTCP packets are sent and received by RTCRtpSender and RTCRtpReceiver objects, as well other data such as SCTP packets sent and received by data channels. In particular, DTLS adds security to an underlying transport, and the RTCDtlsTransport interface allows access to information about the underlying transport and the security added.

interface RTCDtlsTransport {
    readonly attribute RTCIceTransport       transport;
    readonly attribute RTCDtlsTransportState state;
    sequence<ArrayBuffer> getRemoteCertificates();
             attribute EventHandler          onstatechange;
};

Attributes

transport of type RTCIceTransport, readonly

The transport attribute is the underlying transport that is used to send and receive packets. The underlying transport may not be shared between multiple active RTCDtlsTransport objects.

state of type RTCDtlsTransportState, readonly

The state attribute MUST return the state of the transport.

onstatechange of type EventHandler
This event handler, of event handler event type statechange, MUST be fired any time the RTCDtlsTransport state changes.

Methods

getRemoteCertificates

Returns the certificate chain in use by the remote side, with each certificate encoded in binary Distinguished Encoding Rules (DER) [X690]. getRemoteCertificates() will return an empty list prior to selection of the remote certificate, which will be completed by the time RTCDtlsTransportState transitions to "connected".

No parameters.
Return type: sequence<ArrayBuffer>

RTCDtlsTransportState Enum

enum RTCDtlsTransportState {
    "new",
    "connecting",
    "connected",
    "closed",
    "failed"
};
Enumeration description
new DTLS has not started negotiating yet.
connecting DTLS is in the process of negotiating a secure connection.
connected DTLS has completed negotiation of a secure connection.
closed The transport has been closed.
failed The transport has failed as the result of an error (such as a failure to validate the remote fingerprint).

5.6 RTCIceTransport Interface

The RTCIceTransport interface allows an application access to information about the ICE transport over which packets are sent and received. In particular, ICE manages peer-to-peer connections which involve state which the application may want to access.

interface RTCIceTransport {
    readonly attribute RTCIceRole           role;
    readonly attribute RTCIceComponent      component;
    readonly attribute RTCIceTransportState state;
    readonly attribute RTCIceGatheringState gatheringState;
    sequence<RTCIceCandidate> getLocalCandidates();
    sequence<RTCIceCandidate> getRemoteCandidates();
    RTCIceCandidatePair?      getSelectedCandidatePair();
    RTCIceParameters?         getLocalParameters();
    RTCIceParameters?         getRemoteParameters();
             attribute EventHandler         onstatechange;
             attribute EventHandler         ongatheringstatechange;
             attribute EventHandler         onselectedcandidatepairchange;
};

Attributes

role of type RTCIceRole, readonly

The role attribute MUST return the ICE role of the transport.

component of type RTCIceComponent, readonly

The component attribute MUST return the ICE component of the transport. When RTP/RTCP mux is used, a single RTCIceTransport transports both RTP and RTCP and component is set to "RTP".

state of type RTCIceTransportState, readonly

The state attribute MUST return the state of the transport.

gatheringState of type RTCIceGatheringState, readonly

The gathering state attribute MUST return the gathering state of the transport.

onstatechange of type EventHandler
This event handler, of event handler event type statechange, MUST be fired any time the RTCIceTransport state changes.
ongatheringstatechange of type EventHandler
This event handler, of event handler event type gatheringstatechange, MUST be fired any time the RTCIceTransportgathering state changes.
onselectedcandidatepairchange of type EventHandler
This event handler, of event handler event type selectedcandidatepairchange, MUST be fired any time the RTCIceTransport's selected candidate pair changes.

Methods

getLocalCandidates

Returns a sequence describing the local ICE candidates gathered for this RTCIceTransport and sent in onicecandidate

No parameters.
Return type: sequence<RTCIceCandidate>
getRemoteCandidates

Returns a sequence describing the remote ICE candidates received by this RTCIceTransport via addIceCandidate()

No parameters.
Return type: sequence<RTCIceCandidate>
getSelectedCandidatePair

Returns the selected candidate pair on which packets are sent, or null if there is no such pair.

No parameters.
Return type: RTCIceCandidatePair, nullable
getLocalParameters

Returns the local ICE parameters received by this RTCIceTransport via setLocalDescription, or null if the parameters have not yet been received.

No parameters.
Return type: RTCIceParameters, nullable
getRemoteParameters

Returns the remote ICE parameters received by this RTCIceTransport via setRemoteDescription or null if the parameters have not yet been received.

No parameters.
Return type: RTCIceParameters, nullable 
dictionary RTCIceParameters {
    DOMString usernameFragment;
    DOMString password;
};

Dictionary RTCIceParameters Members

usernameFragment of type DOMString

The ICE username fragment as defined in [ICE], Section 7.1.2.3.

password of type DOMString

The ICE password as defined in [ICE], Section 7.1.2.3.

dictionary RTCIceCandidatePair {
    RTCIceCandidate local;
    RTCIceCandidate remote;
};

Dictionary RTCIceCandidatePair Members

local of type RTCIceCandidate

The local ICE candidate.

remote of type RTCIceCandidate

The remote ICE candidate.

RTCIceTransportState Enum

enum RTCIceTransportState {
    "new",
    "checking",
    "connected",
    "completed",
    "failed",
    "disconnected",
    "closed"
};
Enumeration description
new The RTCIceTransport is gathering candidates and/or waiting for remote candidates to be supplied, and has not yet started checking.
checking The RTCIceTransport has received at least one remote candidate and is checking candidate pairs and has either not yet found a connection or consent checks [RFC7675] have failed on all previously successful candidate pairs. In addition to checking, it may also still be gathering.
connected The RTCIceTransport has found a usable connection, but is still checking other candidate pairs to see if there is a better connection. It may also still be gathering and/or waiting for additional remote candidates. If consent checks [RFC7675] fail on the connection in use, and there are no other successful candidate pairs available, then the state transitions to "checking" (if there are candidate pairs remaining to be checked) or "disconnected" (if there are no candidate pairs to check, but the peer is still gathering and/or waiting for additional remote candidates).
completed The RTCIceTransport has finished gathering, received an indication that there are no more remote candidates, finished checking all candidate pairs and found a connection. If consent checks [RFC7675] subsequently fail on all successful candidate pairs, the state transitions to "failed".
failed The RTCIceTransport has finished gathering, received an indication that there are no more remote candidates, finished checking all candidate pairs, and all pairs have either failed connectivity checks or have lost consent.
disconnected Liveness checks have failed. This is more aggressive than failed, and may trigger intermittently (and resolve itself without action) on a flaky network. Alternatively, the RTCIceTransport has finished checking all existing candidates pairs and failed to find a connection (or consent checks [RFC7675] once successful, have now failed), but is still gathering and/or waiting for additional remote candidates.
closed The RTCIceTransport has shut down and is no longer responding to STUN requests.

The failed and completed states require an indication that there are no additional remote candidates. This can be indicated either by canTrickleIceCandidates being set to false, or the processing of an end-of-candidates indication as described in [JSEP].

Some example transitions might be:

ICE state transition diagram
Fig. 2 Non-normative ICE state transition diagram

RTCIceRole Enum

enum RTCIceRole {
    "controlling",
    "controlled"
};
Enumeration description
controlling A controlling agent as defined by [ICE], Section 3.
controlled A controlled agent as defined by [ICE], Section 3.

RTCIceComponent Enum

enum RTCIceComponent {
    "RTP",
    "RTCP"
};
Enumeration description
RTP The ICE Transport is used for RTP (or RTP/RTCP-multiplexing), as defined in [ICE], Section 4.1.1.1. Protocols multiplexed with RTP (e.g. data channel) share its component ID.
RTCP The ICE Transport is used for RTCP as defined by [ICE], Section 4.1.1.1.

5.7 RTCTrackEvent

The track event uses the RTCTrackEvent interface.

Firing an RTCTrackEvent event named e with an RTCRtpReceiver receiver, a MediaStreamTrack track and a MediaStream[] streams, 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 RTCTrackEvent interface with the receiver attribute set to receiver, track attribute set to track, streams attribute set to streams, MUST be created and dispatched at the given target.

[Constructor(DOMString type, RTCTrackEventInit eventInitDict)]
interface RTCTrackEvent : Event {
    readonly attribute RTCRtpReceiver           receiver;
    readonly attribute MediaStreamTrack         track;
    readonly attribute FrozenArray<MediaStream> streams;
    readonly attribute RTCRtpTransceiver        transceiver;
};

Constructors

RTCTrackEvent
readonly attribute RTCRtpReceiver receiver
Parameter Type Nullable Optional Description
type DOMString ✘ ✘
eventInitDict RTCTrackEventInit ✘ ✘

Attributes

receiver of type RTCRtpReceiver, readonly

The receiver attribute represents the RTCRtpReceiver object associated with the event.

track of type MediaStreamTrack, readonly

The track attribute represents the MediaStreamTrack object that is associated with the RTCRtpReceiver identified by receiver.

streams of type FrozenArray<MediaStream>, readonly

The streams attribute returns an array of MediaStream objects representing the MediaStreams that this event's track is a part of.

transceiver of type RTCRtpTransceiver, readonly

The transceiver attribute represents the RTCRtpTransceiver object associated with the event.

dictionary RTCTrackEventInit : EventInit {
    required RTCRtpReceiver        receiver;
    required MediaStreamTrack      track;
             sequence<MediaStream> streams = [];
    required RTCRtpTransceiver     transceiver;
};

Dictionary RTCTrackEventInit Members

receiver of type RTCRtpReceiver, required

The receiver attribute represents the RTCRtpReceiver object associated with the event.

track of type MediaStreamTrack, required

The track attribute represents the MediaStreamTrack object that is associated with the RTCRtpReceiver identified by receiver.

streams of type sequence<MediaStream>, defaulting to []

The streams attribute returns an array of MediaStream objects representing the MediaStreams that this event's track is a part of.

transceiver of type RTCRtpTransceiver, required

The transceiver attribute represents the RTCRtpTransceiver object associated with the event.

6. Peer-to-peer Data API

The Peer-to-peer Data API lets a web application send and receive generic application data peer-to-peer. The API for sending and receiving data models the behavior of WebSockets [WEBSOCKETS-API].

6.1 RTCPeerConnection Interface Extensions

The Peer-to-peer data API extends the RTCPeerConnection interface as described below.

partial interface RTCPeerConnection {
    readonly attribute RTCSctpTransport? sctp;
    RTCDataChannel createDataChannel([TreatNullAs=EmptyString] USVString label,
                                     optional RTCDataChannelInit dataChannelDict);
             attribute EventHandler      ondatachannel;
};

Attributes

sctp of type RTCSctpTransport, readonly , nullable

The SCTP transport over which SCTP data is sent and received. If SCTP has not been negotiated, the value is null.

ondatachannel of type EventHandler
The event type of this event handler is datachannel.

Methods

createDataChannel

Creates a new RTCDataChannel object with the given label. The RTCDataChannelInit dictionary can be used to configure properties of the underlying channel such as data reliability.

When the createDataChannel method is invoked, the user agent MUST run the following steps.

  1. Let connection be the RTCPeerConnection object on which the method is invoked.

  2. If connection's [[isClosed]] slot is true, throw an InvalidStateError exception and abort these steps.

  3. Let channel be a newly created RTCDataChannel object.

  4. Initialize channel's label attribute to the value of the first argument.

  5. If the second (dictionary) argument is present, set channel's ordered, maxPacketLifeTime, maxRetransmits, protocol, negotiated and id attributes to the values of their corresponding dictionary members (if present in the dictionary).

  6. If negotiated is false and label is longer than 65535 bytes long, throw a TypeError.
  7. If negotiated is false and protocol is longer than 65535 bytes long, throw a TypeError.

  8. If both the maxPacketLifeTime and maxRetransmits attributes are set (not null), then throw a SyntaxError exception and abort these steps.

  9. If an attribute, either maxPacketLifeTime or maxRetransmits, has been set to indicate unreliable mode, and that value exceeds the maximum value supported by the user agent, the value MUST be set to the user agents maximum value.

  10. If id attribute is uninitialized (not set via the dictionary), initialize it to a value generated by the user agent, according to the WebRTC DataChannel Protocol specification, and skip to the next step. Otherwise, if the value of the id attribute is taken by an existing RTCDataChannel, throw a ResourceInUse exception and abort these steps.

  11. Return channel and continue the following steps in the background.

  12. Create channel's associated underlying data transport and configure it according to the relevant properties of channel.

  13. If channel was the first RTCDataChannel created on connection, mark connection as needing negotiation.

Parameter Type Nullable Optional Description
label USVString ✘ ✘
dataChannelDict RTCDataChannelInit ✘ ✔
Return type: RTCDataChannel

6.1.1 RTCSctpTransport Interface

The RTCSctpTransport interface allows an application access to information about the SCTP data channels tied to a particular SCTP association.

interface RTCSctpTransport {
    readonly attribute RTCDtlsTransport transport;
    readonly attribute unsigned long    maxMessageSize;
};
Attributes
transport of type RTCDtlsTransport, readonly

The transport over which all SCTP packets for data channels will be sent and received.

maxMessageSize of type unsigned long, readonly

The maximum size of data that can be passed to RTCDataChannel's send() method.

6.2 RTCDataChannel

The RTCDataChannel interface represents a bi-directional data channel between two peers. A RTCDataChannel is created via a factory method on an RTCPeerConnection object. The messages sent between the browsers are described in [RTCWEB-DATA] and [RTCWEB-DATA-PROTOCOL].

There are two ways to establish a connection with RTCDataChannel. The first way is to simply create a RTCDataChannel at one of the peers with the negotiated RTCDataChannelInit dictionary member unset or set to its default value false. This will announce the new channel in-band and trigger a RTCDataChannelEvent with the corresponding RTCDataChannel object at the other peer. The second way is to let the application negotiate the RTCDataChannel. To do this, create a RTCDataChannel object with the negotiated RTCDataChannelInit 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 negotiated RTCDataChannelInit 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. The actual wire protocol between the peers is specified by the WebRTC DataChannel Protocol specification [RTCWEB-DATA].

A 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.

A RTCDataChannel, created with createDataChannel or dispatched via a RTCDataChannelEvent, MUST initially be in the connecting state. When the RTCDataChannel object's underlying data transport is ready, the user agent MUST announce the RTCDataChannel as open.

When the user agent is to announce a RTCDataChannel as open, the user agent MUST queue a task to run the following steps:

  1. If the associated RTCPeerConnection object's [[isClosed]] slot is true, abort these steps.

  2. Let channel be the RTCDataChannel object to be announced.

  3. Set channel's readyState attribute to open.

  4. Fire a simple event named open at channel.

When an underlying data transport is to be announced (the other peer created a channel with negotiated unset or set to false), the user agent of the peer that did not initiate the creation process MUST queue a task to run the following steps:

  1. If the associated RTCPeerConnection object's [[isClosed]] slot is true, abort these steps.

  2. Let channel be a newly created RTCDataChannel object.

  3. Let configuration be an information bundle received from the other peer as a part of the process to establish the underlying data transport described by the WebRTC DataChannel Protocol specification [RTCWEB-DATA-PROTOCOL].

  4. Initialize channel's label, ordered, maxPacketLifeTime, maxRetransmits, protocol, negotiated and id attributes to their corresponding values in configuration.

  5. Set channel's readyState attribute to connecting.

  6. Fire a datachannel event named datachannel with channel at the RTCPeerConnection object.

An RTCDataChannel object's underlying data transport may be torn down in a non-abrupt manner by running the closing procedure. When that happens the user agent MUST, unless the procedure was initiated by the close method, queue a task that sets the object's readyState attribute to closing. This will eventually render the data transport closed.

When a RTCDataChannel object's underlying data transport has been closed, the user agent MUST queue a task to run the following steps:

  1. Let channel be the RTCDataChannel object whose transport was closed.

    Issue 4
    The data transport protocol will specify what happens to, e.g. buffered data, when the data transport is closed.

  2. Set channel's readyState attribute to closed.

  3. If the transport was closed with an error, fire an NetworkError event at channel.

  4. Fire a simple event named close at channel.

interface RTCDataChannel : EventTarget {
    readonly attribute USVString           label;
    readonly attribute boolean             ordered;
    readonly attribute unsigned short?     maxPacketLifeTime;
    readonly attribute unsigned short?     maxRetransmits;
    readonly attribute USVString           protocol;
    readonly attribute boolean             negotiated;
    readonly attribute unsigned short      id;
    readonly attribute RTCDataChannelState readyState;
    readonly attribute unsigned long       bufferedAmount;
             attribute unsigned long       bufferedAmountLowThreshold;
             attribute EventHandler        onopen;
             attribute EventHandler        onbufferedamountlow;
             attribute EventHandler        onerror;
             attribute EventHandler        onclose;
    void close();
             attribute EventHandler        onmessage;
             attribute DOMString           binaryType;
    void send(USVString data);
    void send(Blob data);
    void send(ArrayBuffer data);
    void send(ArrayBufferView data);
};

Attributes

label of type USVString, readonly

The label attribute represents a label that can be used to distinguish this RTCDataChannel object from other RTCDataChannel objects. Scripts are allowed to create multiple RTCDataChannel objects with the same label. The attribute MUST return the value to which it was set when the RTCDataChannel object was created.

ordered of type boolean, readonly

The ordered attribute returns true if the RTCDataChannel is ordered, and false if other of order delivery is allowed. The attribute MUST be initialized to true by default and MUST return the value to which it was set when the RTCDataChannel was created.

maxPacketLifeTime of type unsigned short, readonly , nullable

The maxPacketLifeTime attribute returns the length of the time window (in milliseconds) during which transmissions and retransmissions may occur in unreliable mode, 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 created.

maxRetransmits of type unsigned short, readonly , nullable

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 created.

protocol of type USVString, readonly

The protocol attribute returns the name of the sub-protocol used with this RTCDataChannel if any, or the empty string otherwise. The attribute MUST be initialized to the empty string by default and MUST return the value to which it was set when the RTCDataChannel was created.

negotiated of type boolean, readonly

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 created.

id of type unsigned short, readonly

The id attribute returns the id for this RTCDataChannel. The id was either assigned by the user agent at channel creation time or selected by the script. The attribute MUST return the value to which it was set when the RTCDataChannel was created.

readyState of type RTCDataChannelState, readonly

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).

bufferedAmount of type unsigned long, readonly

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 thus 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).

bufferedAmountLowThreshold of type unsigned long

The bufferedAmountLowThreshold attribute sets the threshold at which the bufferedAmount is considered to be low. When the bufferedAmount decreases from above this threshold to equal or below it, the bufferedamountlow event fires. The bufferedAmountLowThreshold is initially zero on each new RTCDataChannel, but the application may change its value at any time.

onopen of type EventHandler
The event type of this event handler is open.
onbufferedamountlow of type EventHandler
The event type of this event handler is bufferedamountlow.
onerror of type EventHandler
The event type of this event handler is error.
onclose of type EventHandler
The event type of this event handler is close.
onmessage of type EventHandler
The event type of this event handler is message.
binaryType of type DOMString

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 a RTCDataChannel object is created, the binaryType attribute MUST be initialized to the string "blob".

This attribute controls how binary data is exposed to scripts. See the [WEBSOCKETS-API] for more information.

Methods

close

Closes the RTCDataChannel. It may be called regardless of whether the RTCDataChannel object was created by this peer or the remote peer.

When the close method is called, the user agent MUST run the following steps:

  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.

No parameters.
Return type: void
send

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

Parameter Type Nullable Optional Description
data USVString ✘ ✘
Return type: void
send

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

Parameter Type Nullable Optional Description
data Blob ✘ ✘
Return type: void
send

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

Parameter Type Nullable Optional Description
data ArrayBuffer ✘ ✘
Return type: void
send

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

Parameter Type Nullable Optional Description
data ArrayBufferView ✘ ✘
Return type: void 
dictionary RTCDataChannelInit {
    boolean        ordered = true;
    unsigned short maxPacketLifeTime;
    unsigned short maxRetransmits;
    USVString      protocol = "";
    boolean        negotiated = false;
    unsigned short id;
};

Dictionary RTCDataChannelInit Members

ordered of type boolean, defaulting to true

If set to false, data is allowed to be delivered out of order. The default value of true, guarantees that data will be delivered in order.

maxPacketLifeTime of type unsigned short

Limits the time during which the channel will transmit or retransmit data if not acknowledged. This value may be clamped if it exceeds the maximum value supported by the user agent.

maxRetransmits of type unsigned short

Limits the number of times a channel will retransmit data if not successfully delivered. This value may be clamped if it exceeds the maximum value supported by the user agent.

protocol of type USVString, defaulting to ""

Subprotocol name used for this channel.

negotiated of type boolean, defaulting to false

The default value of false tells the user agent to announce the channel in-band and instruct the other peer to dispatch a corresponding RTCDataChannel object. If set to true, it is up to the application to negotiate the channel and create a RTCDataChannel object with the same id at the other peer.

id of type unsigned short

Overrides the default selection of id for this channel.

The send() method is overloaded to handle different data argument types. When any version of the method is called, the user agent MUST run the following steps:

  1. Let channel be the RTCDataChannel object on which data is to be sent.

  2. If channel's readyState attribute is connecting, throw an InvalidStateError exception and abort these steps.

  3. Execute the sub step that corresponds to the type of the methods argument:

    • string object:

      Let data be the object and increase the bufferedAmount attribute by the number of bytes needed to express data as UTF-8.

    • Blob object:

      Let data be the raw data represented by the Blob object and increase the bufferedAmount attribute by the size of data, in bytes.

    • ArrayBuffer object:

      Let data be the data stored in the buffer described by the ArrayBuffer object and increase the bufferedAmount attribute by the length of the ArrayBuffer in bytes.

    • ArrayBufferView object:

      Let data be the data stored in the section of the buffer described by the ArrayBuffer object that the ArrayBufferView object references and increase the bufferedAmount attribute by the length of the ArrayBufferView in bytes.

  4. If channel's underlying data transport is not established yet, or if the closing procedure has started, then abort these steps.

  5. Attempt to send data on channel's underlying data transport; if the data cannot be sent, e.g. because it would need to be buffered but the buffer is full, the user agent MUST abruptly close channel's underlying data transport with an error.

enum RTCDataChannelState {
    "connecting",
    "open",
    "closing",
    "closed"
};
Enumeration description
connecting

The user agent is attempting to establish the underlying data transport. This is the initial state of a RTCDataChannel object created with createDataChannel.
open

The underlying data transport is established and communication is possible. This is the initial state of a RTCDataChannel object dispatched as a part of a 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.

6.3 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)]
interface RTCDataChannelEvent : Event {
    readonly attribute RTCDataChannel channel;
};

Constructors

RTCDataChannelEvent
readonly attribute RTCDataChannel channel
Parameter Type Nullable Optional Description
type DOMString ✘ ✘
eventInitDict RTCDataChannelEventInit ✘ ✘

Attributes

channel of type RTCDataChannel, readonly

The channel attribute represents the RTCDataChannel object associated with the event.

dictionary RTCDataChannelEventInit : EventInit {
    RTCDataChannel channel;
};

Dictionary RTCDataChannelEventInit Members

channel of type RTCDataChannel
Issue 5

TODO

6.4 Garbage Collection

A RTCDataChannel object MUST not be garbage collected if its

7. Peer-to-peer DTMF

This section describes an interface on RTCRtpSender to send DTMF (phone keypad) values across an RTCPeerConnection. Details of how DTMF is sent to the other peer are described in [RTCWEB-AUDIO].

7.1 RTCRtpSender Interface Extensions

The Peer-to-peer DTMF API extends the RTCRtpSender interface as described below.

partial interface RTCRtpSender {
    readonly attribute RTCDTMFSender? dtmf;
};

Attributes

dtmf of type RTCDTMFSender, readonly , nullable

The dtmf attribute returns a RTCDTMFSender which can be used to send DTMF. A null value indicates that this RTCRtpSender cannot send DTMF.

7.2 RTCDTMFSender

[NoInterfaceObject]
interface RTCDTMFSender {
    void insertDTMF(DOMString tones,
                    optional unsigned long duration = 100,
                    optional unsigned long interToneGap = 70);
             attribute EventHandler ontonechange;
    readonly attribute DOMString    toneBuffer;
    readonly attribute long         duration;
    readonly attribute long         interToneGap;
};

Attributes

ontonechange of type EventHandler

The event type of this event handler is tonechange.

toneBuffer of type DOMString, readonly

The toneBuffer attribute MUST return a list of the tones remaining to be played out. For the syntax, content, and interpretation of this list, see insertDTMF.

duration of type long, readonly

The duration attribute MUST return the current tone duration value. This value will be the value last set via the insertDTMF method, or the default value of 100 ms if insertDTMF was called without specifying the duration.

interToneGap of type long, readonly

The interToneGap attribute MUST return 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.

Methods

insertDTMF

An RTCDTMFSender object's insertDTMF method is used to send DTMF tones.

The tones parameter is treated as a series of characters. The characters 0 through 9, A through D, #, and * generate the associated DTMF tones. The characters a to d are equivalent to A to D. The character ',' indicates a delay of 2 seconds before processing the next character in the tones parameter. All other characters MUST be considered unrecognized. As noted in [RTCWEB-AUDIO] Section 3, support for the characters 0 through 9, A through D, #, and * are required.

The duration parameter indicates the duration in ms to use for each character passed in the tones parameters. The duration cannot be more than 8000 ms or less than 40 ms. The default duration is 100 ms for each tone.

The interToneGap parameter indicates the gap between tones. It MUST be at least 30 ms. The default value is 70 ms.

The browser MAY increase the duration and interToneGap times to cause the times that DTMF start and stop to align with the boundaries of RTP packets but it MUST not increase either of them by more than the duration of a single RTP audio packet.

When the insertDTMF() method is invoked, the user agent MUST run the following steps:

  1. Set the object's toneBuffer attribute to the value of the first argument, the duration attribute to the value of the second argument, and the interToneGap attribute to the value of the third argument.
  2. If toneBuffer contains any unrecognized characters, throw an InvalidCharacterError exception and abort these steps.
  3. If toneBuffer is an empty string, return.
  4. If the value of the duration attribute is less than 40, set it to 40. If, on the other hand, the value is greater than 6000, set it to 6000.
  5. If the value of the interToneGap attribute is less than 30, set it to 30.
  6. If a Playout task is scheduled to be run; abort these steps; otherwise queue a task that runs the following steps (Playout task):
    1. If toneBuffer is an empty string, fire an event named tonechange with an empty string at the RTCDTMFSender object and abort these steps.
    2. Remove the first character from toneBuffer and let that character be tone.
    3. Start playout of tone for duration ms on the associated RTP media stream, using the appropriate codec.
    4. Queue a task to be executed in duration + interToneGap ms from now that runs the steps labelled Playout task.
    5. Fire an event named tonechange with a string consisting of tone at the RTCDTMFSender object.

Calling insertDTMF with an empty tones parameter can be used to cancel all tones queued to play after the currently playing tone.

Parameter Type Nullable Optional Description
tones DOMString ✘ ✘
duration unsigned long = 100 ✘ ✔
interToneGap unsigned long = 70 ✘ ✔
Return type: void

7.3 RTCDTMFToneChangeEvent

The tonechange event uses the RTCDTMFToneChangeEvent interface.

Firing a 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)]
interface RTCDTMFToneChangeEvent : Event {
    readonly attribute DOMString tone;
};

Constructors

RTCDTMFToneChangeEvent
readonly attribute DOMString tone
Parameter Type Nullable Optional Description
type DOMString ✘ ✘
eventInitDict RTCDTMFToneChangeEventInit ✘ ✘

Attributes

tone of type DOMString, readonly

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.

dictionary RTCDTMFToneChangeEventInit : EventInit {
    DOMString tone;
};

Dictionary RTCDTMFToneChangeEventInit Members

tone of type DOMString
Issue 6

TODO

8. Statistics Model

8.1 Introduction

The basic statistics model is that the browser maintains a set of statistics referenced by a selector. The selector may, for example, be a MediaStreamTrack. For a track to be a valid selector, it MUST be a MediaStreamTrack that is sent or received by the RTCPeerConnection object on which the stats request was issued. The calling Web application provides the selector to the getStats() method and the browser emits (in the JavaScript) a set of statistics that it believes is relevant to the selector.

Issue 7
Evaluate the need for other selectors than MediaStreamTrack.

The statistics returned are designed in such a way that repeated queries can be linked by the RTCStats id dictionary member. Thus, a Web application can make measurements over a given time period by requesting measurements at the beginning and end of that period.

8.2 RTCPeerConnection Interface Extensions

The Statistics API extends the RTCPeerConnection interface as described below.

partial interface RTCPeerConnection {
    Promise<RTCStatsReport> getStats(optional MediaStreamTrack? selector = null);
};

Methods

getStats

Gathers stats for the given selector and reports the result asynchronously.

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

  1. Let selectorArg be the methods first argument.

  2. If selectorArg is neither null nor a valid selector, return a promise rejected with a TypeError.

  3. Let p be a new promise.

  4. Run the following steps in parallel:

    1. Start gathering the stats indicated by selectorArg. If selectorArg is null, stats MUST be gathered for the whole RTCPeerConnection object.

    2. When the relevant stats have been gathered, resolve p with a new RTCStatsReport object, representing the gathered stats.

  5. Return p.

Parameter Type Nullable Optional Description
selector MediaStreamTrack = null ✔ ✔
Return type: Promise<RTCStatsReport>

8.3 RTCStatsCallback

callback RTCStatsCallback = void (RTCStatsReport report);

Callback RTCStatsCallback Parameters

report of type RTCStatsReport

A RTCStatsReport representing the gathered stats.

8.4 RTCStatsReport Object

The getStats() method delivers a successful result in the form of an RTCStatsReport object. An RTCStatsReport object is a map between strings that identify the inspected objects (id attribute in RTCStats instances), and their corresponding RTCStats-derived dictionaries.

An RTCStatsReport may be composed of several RTCStats-derived dictionaries, each reporting stats for one underlying object that the implementation thinks is relevant for the selector. One achieves the total for the selector by summing over all the stats of a certain type; for instance, if a MediaStreamTrack is carried by multiple SSRCs over the network, the RTCStatsReport may contain one RTCStats-derived dictionary per SSRC (which can be distinguished by the value of the "ssrc" stats attribute).

interface RTCStatsReport {
    readonly maplike<DOMString, object>;
};

This interface has "entries", "forEach", "get", "has", "keys", "values", @@iterator methods and a "size" getter brought by readonly maplike.

Use these to retrieve the various dictionaries descended from RTCStats that this stats report is composed of. The set of supported property names [WEBIDL-1] is defined as the ids of all the RTCStats-derived dictionaries that have been generated for this stats report.

8.5 RTCStats Dictionary

An RTCStats dictionary represents the stats gathered by inspecting a specific object relevant to a selector. 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.

Issue 8
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 an RTCStats-derived dictionary.

dictionary RTCStats {
    DOMHighResTimeStamp timestamp;
    RTCStatsType        type;
    DOMString           id;
};

Dictionary RTCStats Members

timestamp of type DOMHighResTimeStamp

The timestamp, of type DOMHighResTimeStamp [HIGHRES-TIME], associated with this object. The time is relative to the UNIX epoch (Jan 1, 1970, UTC).

type of type RTCStatsType

The type of this object.

The type attribute MUST be initialized to the name of the most specific type this RTCStats dictionary represents.

id of type DOMString

A unique id that is associated with the object that was inspected to produce this RTCStats object. Two RTCStats objects, extracted from two different RTCStatsReport objects, MUST have the same id if they were produced by inspecting the same underlying object. User agents are free to pick any format for the id as long as it meets the requirements above.

Issue 9
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",
    "outboundrtp"
};
Enumeration description
inboundrtp Inbound RTP.
outboundrtp Outbound RTP.

8.6 Derived Stats Dictionaries

dictionary RTCRTPStreamStats : RTCStats {
    unsigned long ssrc;
    DOMString     remoteId;
};

Dictionary RTCRTPStreamStats Members

ssrc of type unsigned long

...

remoteId of type DOMString

The remoteId can be used to look up the corresponding RTCStats object that represents stats reported by the other peer.

dictionary RTCInboundRTPStreamStats : RTCRTPStreamStats {
    unsigned long packetsReceived;
    unsigned long bytesReceived;
};

Dictionary RTCInboundRTPStreamStats Members

packetsReceived of type unsigned long

...

bytesReceived of type unsigned long

...

dictionary RTCOutboundRTPStreamStats : RTCRTPStreamStats {
    unsigned long packetsSent;
    unsigned long bytesSent;
};

Dictionary RTCOutboundRTPStreamStats Members

packetsSent of type unsigned long

...

bytesSent of type unsigned long

...

8.7 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:

Example 3
var baselineReport, currentReport;
var selector = pc.getSenders()[0].track;

pc.getStats(selector).then(function (report) {
    baselineReport = report;
})
.then(function() {
    return new Promise(function(resolve) {
        setTimeout(resolve, aBit); // ... wait a bit
    });
})
.then(function() {
    return pc.getStats(selector);
})
.then(function (report) {
    currentReport = report;
    processStats();
})
.catch(function (error) {
  log(error.toString());
});

function processStats() {
    // compare the elements from the current report with the baseline
    currentReport.forEach (now => {
        if (now.type != "outboundrtp")
            return;

        // get the corresponding stats from the baseline report
        base = baselineReport.get(now.id);

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

9. Identity

9.1 Identity Provider Interaction

WebRTC offers and answers (and hence the channels established by RTCPeerConnection objects) can be authenticated by using a web-based Identity Provider (IdP). The idea is that the entity sending an offer or answer acts as the Authenticating Party (AP) and obtains an identity assertion from the IdP which it attaches to the session description. The consumer of the session description (i.e., the RTCPeerConnection on which setRemoteDescription is called) 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; the browser need only know how to load the IdP's JavaScript, the location of which is determined by the IdP's identity, and the generic interface to generating and validating 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 browser was written.

9.1.1 Identity Provider Selection

An IdP is used to generate an identity assertion as follows:

  1. If the setIdentityProvider() method has been called, the IdP provided shall be used.
  2. If the setIdentityProvider() method has not been called, then the user agent MAY 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.

9.1.2 Instantiating an IdP Proxy

In order to communicate with the IdP, the user agent loads the IdP JavaScript from the IdP. The URI for the IdP script is a well-known URI formed from the domain and protocol fields, as specified in [RTCWEB-SECURITY-ARCH].

The IdP MAY generate an HTTP redirect to another "https" origin, the browser MUST treat a redirect to any other scheme as a fatal error.

The user agent instantiates an isolated interpreted context, a JavaScript realm that operates in the origin of the loaded JavaScript. Note that a redirect will change the origin of the loaded script.

The realm is populated with a global that implements WorkerGlobalScope [WEBWORKERS].

The user agent provides an instance of RTCIdentityProviderRegistrar named rtcIdentityProvider in the global scope of the realm. This object is used by the IdP to interact with the user agent.

A global property can only be set by the user agent or the IdP proxy itself. Therefore, the IdP proxy can be assured that requests it receives originate from the user agent. This ensures that an arbitrary origin is unable to instantiate an IdP proxy and impersonate this API in order obtain identity assertions.

interface RTCIdentityProviderGlobalScope : WorkerGlobalScope {
    readonly attribute RTCIdentityProviderRegistrar rtcIdentityProvider;
};
Attributes
rtcIdentityProvider of type RTCIdentityProviderRegistrar, readonly
This object is used by the IdP to register an RTCIdentityProvider instance with the browser.

9.2 Registering an IdP Proxy

An IdP proxy implements the RTCIdentityProvider methods, which are the means by which the user agent is able to request that an identity assertion be generated or validated.

Once instantiated, the IdP script is executed. The IdP MUST call the register() function on the RTCIdentityProviderRegistrar instance during script execution. If an IdP is not registered during this script execution, the user agent cannot use the IdP proxy and MUST fail any future attempt to interact with the IdP.

interface RTCIdentityProviderRegistrar {
    void register(RTCIdentityProvider idp);
};

Methods

register

This method is invoked by the IdP when its script is first executed. This registers RTCIdentityProvider methods with the user agent.

Parameter Type Nullable Optional Description
idp RTCIdentityProvider ✘ ✘
Return type: void

9.2.1 Interface Exposed by Identity Providers

The callback functions in RTCIdentityProvider are exposed by identity providers and is called by RTCPeerConnection to acquire or validate identity assertions.

dictionary RTCIdentityProvider {
    required GenerateAssertionCallback generateAssertion;
    required ValidateAssertionCallback validateAssertion;
};
Dictionary RTCIdentityProvider Members
generateAssertion of type GenerateAssertionCallback, required

A user agent invokes this method on the IdP to request the generation of an identity assertion.

The IdP provides a promise that resolves to an RTCIdentityAssertionResult to successfully generate an identity assertion. Any other value, or a rejected promise, is treated as an error.

validateAssertion of type ValidateAssertionCallback, required

A user agent invokes this method on the IdP to request the validation of an identity assertion.

The IdP returns a Promise that resolves to an RTCIdentityValidationResult to successfully validate an identity assertion and to provide the actual identity. Any other value, or a rejected promise, is treated as an error.

callback GenerateAssertionCallback = Promise<RTCIdentityAssertionResult> (DOMString contents,
                                                                          DOMString origin,
                                                                          optional DOMString usernameHint);
Callback GenerateAssertionCallback Parameters
contents of type DOMString
The contents parameter includes the information that the user agent wants covered by the identity assertion. A successful validation of the provided assertion MUST produce this string.
origin of type DOMString
The origin parameter identifies the origin of the RTCPeerConnection that triggered this request. An IdP can use this information as input to policy decisions about use. This value is generated by the user agent based on the origin of the document that created the RTCPeerConnection and therefore can be trusted to be correct.
usernameHint of type DOMString
The IdP selects the identity to assert. The optional usernameHint parameter is the same value that was passed to setIdentityProvider.
callback ValidateAssertionCallback = Promise<RTCIdentityValidationResult> (DOMString assertion,
                                                                           DOMString origin);
Callback ValidateAssertionCallback Parameters
assertion of type DOMString
The assertion parameter includes the assertion that was recovered from an a=identity in the session description; that is, the value that was part of the RTCIdentityAssertionResult provided by the IdP that generated the assertion.
origin of type DOMString
The origin parameter identifies the origin of the RTCPeerConnection that triggered this request. An IdP can use this information as input to policy decisions about use.

9.2.2 Identity Assertion and Validation Results

dictionary RTCIdentityAssertionResult {
    required RTCIdentityProviderDetails idp;
    required DOMString                  assertion;
};
Dictionary RTCIdentityAssertionResult Members
idp of type RTCIdentityProviderDetails, required

An IdP provides these details to identify the IdP that validates the identity assertion. This struct contains the same information that is provided to setIdentityProvider.

assertion of type DOMString, required

An identity assertion. This is an opaque string that MUST contain all information necessary to assert identity. This value is consumed by the validating IdP.

dictionary RTCIdentityProviderDetails {
    required DOMString domain;
             DOMString protocol = "default";
};
Dictionary RTCIdentityProviderDetails Members
domain of type DOMString, required

The domain name of the IdP that validated the associated identity assertion.

protocol of type DOMString, defaulting to "default"

The protocol parameter used for the IdP.

dictionary RTCIdentityValidationResult {
    required DOMString identity;
    required DOMString contents;
};
Dictionary RTCIdentityValidationResult Members
identity of type DOMString, required

The validated identity of the peer.

contents of type DOMString, required

The payload of the identity assertion. An IdP that validates an identity assertion MUST return the same string that was provided to the original IdP that generated the assertion.

The user agent uses the contents string to determine if the identity assertion matches the session description.

9.3 Requesting Identity Assertions

The identity assertion request process is triggered by a call to createOffer, createAnswer, or getIdentityAssertion. When these calls are invoked and an identity provider has been set, the following steps are executed:

  1. The RTCPeerConnection instantiates an IdP as described in Identity Provider Selection and Registering an IdP Proxy. If the IdP cannot be loaded, instantiated, or the IdP proxy is not registered, this process fails.

  2. The RTCPeerConnection invokes the generateAssertion method on the RTCIdentityProvider methods registered by the IdP.

    The RTCPeerConnection generates the contents parameter to this method as described in [RTCWEB-SECURITY-ARCH]. The value of contents includes the fingerprint of the certificate that was selected or generated during the construction of the RTCPeerConnection. The origin parameter contains the origin of the script that calls the RTCPeerConnection method that triggers this behavior. The usernameHint value is the same value that is provided to setIdentityProvider, if any such value was provided.

  3. The IdP returns a Promise to the RTCPeerConnection. If the user has been authenticated by the IdP, and the IdP is willing to generate an identity assertion, the IdP resolves the promise with an identity assertion in the form of an RTCIdentityAssertionResult.

    This step depends entirely on the IdP. The methods by which an IdP authenticates users or generates assertions is not specified, though they could involve interacting with the IdP server or other servers.

  4. The RTCPeerConnection MAY store the identity assertion for use with future offers or answers. If a fresh identity assertion is needed for any reason, applications can create a new RTCPeerConnection.

  5. If the identity request was triggered by a createOffer() or createAnswer(), then the assertion is converted to a JSON string, base64-encoded and inserted into an a=identity attribute in the session description.

This process can fail. The IdP proxy MAY reject the promise, or the process of loading and registering the IdP could fail. If assertion generation fails, then the promise for the corresponding function call is 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.

9.3.1 User Login Procedure

An IdP MAY reject an attempt to generate an identity assertion if it is unable to verify that a user is authenticated. This might be due to the IdP not having the necessary authentication information available to it (such as cookies).

Rejecting the promise returned by generateAssertion will cause the error to propagate to the application. Login errors are indicated by rejecting the promise with an object that has a name attribute set to "IdpLoginError".

If the rejection object also contains a loginUrl attribute, this value will be passed to the application in the idpLoginUrl attribute. This URL might link to page where a user can 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 window; the resulting page then SHOULD provide the user with an opportunity to enter any 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 consist of the DOMString "LOGINDONE". This message informs the application that another attempt at generating an identity assertion is likely to be successful.

9.4 Verifying Identity Assertions

Identity assertion validation happens when setRemoteDescription is invoked on RTCPeerConnection. The process runs asynchronously, meaning that validation of an identity assertion might not block the completion of setRemoteDescription.

The identity assertion request process involves the following asynchronous steps:

  1. The RTCPeerConnection awaits any prior identity validation. Only one identity validation can run at a time for an RTCPeerConnection. This can happen because the resolution of setRemoteDescription is not blocked by identity validation unless there is a target peer identity.

  2. The RTCPeerConnection loads the identity assertion from the session description and decodes the base64 value, then parses the resulting JSON. The idp parameter of the resulting dictionary contains a domain and an optional protocol value that identifies the IdP, as described in [RTCWEB-SECURITY-ARCH].

  3. The RTCPeerConnection instantiates the identified IdP as described in 9.1.1 Identity Provider Selection and 9.2 Registering an IdP Proxy. If the IdP cannot be loaded, instantiated or the IdP proxy is not registered, this process fails.

  4. The RTCPeerConnection invokes the validateAssertion method registered by the IdP.

    The assertion parameter is taken from the decoded identity assertion. The origin parameter contains the origin of the script that calls the RTCPeerConnection method that triggers this behavior.

  5. The IdP proxy returns a promise and performs the validation process asynchronously.

    The IdP proxy verifies the identity assertion using whatever means necessary. Depending on the authentication protocol this could involve interacting with the IDP server.

  6. Once the assertion is successfully verified, the IdP proxy resolves the promise with an RTCIdentityValidationResult containing the validated identity and the original contents that are the payload of the assertion.

  7. The RTCPeerConnection decodes the contents and validates that it contains a fingerprint value for every a=fingerprint attribute in the session description. This ensures that the certificate used by the remote peer for communications is covered by the identity assertion.

    If a peer offers a certificate that doesn't match an a=fingerprint line in the negotiated session description, the user agent MUST NOT permit communication with that peer.

  8. The RTCPeerConnection validates that the domain portion of the identity matches the domain of the IdP as described in [RTCWEB-SECURITY-ARCH].

  9. The RTCPeerConnection resolves the peerIdentity attribute with a new instance of RTCIdentityAssertion that includes the IdP domain and peer identity.

  10. 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.

This process can fail at any step above. If identity validation fails, the peerIdentity promise is rejected with a DOMException that has a name of OperationError.

If identity validation fails and there is a target peer identity for the RTCPeerConnection, the promise returned by setRemoteDescription MUST be rejected.

If identity validation fails and there is no a target peer identity, the value of the peerIdentity MUST be set to a new, unresolved promise instance. This permits the use of renegotiation (or a subsequent answer, if the session description was a provisional answer) to resolve or reject the identity.

The browser SHOULD limit the time that it will allow for identity validation. This includes both the loading of the IdP proxy and the identity assertion validation. Failure to do so potentially causes the operation to take an indefinite amount of time. This timer can be cancelled when the IdP produces a response. The timer running to completion is treated as equivalent to an error from the IdP.

9.5 RTCPeerConnection Interface Extensions

The Identity API extends the RTCPeerConnection interface as described below.

partial interface RTCPeerConnection {
    void               setIdentityProvider(DOMString provider,
                                           optional DOMString protocol,
                                           optional DOMString usernameHint);
    Promise<DOMString> getIdentityAssertion();
    readonly attribute Promise<RTCIdentityAssertion> peerIdentity;
    readonly attribute DOMString?                    idpLoginUrl;
};

Attributes

peerIdentity of type Promise<RTCIdentityAssertion>, readonly

A promise that resolves with the identity of the peer if the identity is successfully validated.

This promise is rejected if an identity assertion is present in a remote session description and validation of that assertion fails for any reason. If the promise is rejected, a new unresolved value is created, unless there a target peer identity has been established. If this promise successfully resolves, the value will not change.

idpLoginUrl of type DOMString, readonly , nullable

The URL that an application can navigate to so that the user can login to the IdP, as described in 9.3.1 User Login Procedure.

Methods

setIdentityProvider

Sets the identity provider to be used for a given RTCPeerConnection object. Applications need not make this call; if the browser is already configured for an IdP, then that configured IdP might be used to get an assertion.

When the setIdentityProvider method is invoked, the user agent MUST run the following steps:

  1. If the RTCPeerConnection object's [[isClosed]] slot is true, throw an InvalidStateError exception and abort these steps.

  2. Set the current identity provider values to the triplet (provider, protocol, usernameHint).

  3. If any identity provider value has changed, discard any stored identity assertion.

Identity provider information is not used until an identity assertion is required, either in response to a call to getIdentityAssertion, or a session description is requested with a call to either createOffer or createAnswer.

Parameter Type Nullable Optional Description
provider DOMString ✘ ✘
protocol DOMString ✘ ✔
usernameHint DOMString ✘ ✔
Return type: void
getIdentityAssertion

Initiates the process of obtaining an identity assertion. Applications need not make this call. It is merely intended to allow them to start the process of obtaining identity assertions before a call is initiated. If an identity is needed, either because the browser has been configured with a default identity provider or because the setIdentityProvider method was called, then an identity will be automatically requested when an offer or answer is created.

When getIdentityAssertion is invoked, queue a task to run the following steps:

  1. If the RTCPeerConnection object's [[isClosed]] slot is true, throw an InvalidStateError exception and abort these steps.

  2. Request an identity assertion from the IdP.

  3. Resolve the promise with the base64 and JSON encoded assertion.

No parameters.
Return type: Promise<DOMString> 
[Constructor(DOMString idp, DOMString name)]
interface RTCIdentityAssertion {
    attribute DOMString idp;
    attribute DOMString name;
};

Attributes

idp of type DOMString

The domain name of the identity provider that validated this identity.

name of type DOMString

An RFC5322-conformant [RFC5322] representation of the verified peer identity. This identity will have been verified via the procedures described in [RTCWEB-SECURITY-ARCH].

9.6 Examples

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.

Example 4
pc.setIdentityProvider("example.com", "default", "alice@example.com");

This example shows how to consume identity assertions inside a Web application.

Example 5
pc.peerIdentity.then(identity =>
  console.log("IdP= " + identity.idp + " identity=" + identity.name));

10. Media Stream API Extensions for Network Use

10.1 Introduction

The MediaStreamTrack interface, as defined in the [GETUSERMEDIA] specification, typically represents a stream of data of audio or video. One or more MediaStreamTracks can be collected in a MediaStream (strictly speaking, a MediaStream as defined in [GETUSERMEDIA] may contain zero or more MediaStreamTrack objects).

A MediaStreamTrack may be extended to represent a media flow that either comes from or is sent to a remote peer (and not just the local camera, for instance). The extensions required to enable this capability on the MediaStreamTrack object will be described in this section. How the media is transmitted to the peer is described in [RTCWEB-RTP], [RTCWEB-AUDIO], and [RTCWEB-TRANSPORT].

A MediaStreamTrack sent to another peer will appear as one and only one MediaStreamTrack to the recipient. A peer is defined as a user agent that supports this specification. In addition, the sending side application can indicate what MediaStream object(s) the MediaStreamTrack is member of. The corresponding MediaStream object(s) on the receiver side will be created (if not already present) and populated accordingly.

As also described earlier in this document, the objects RTCRtpSender and RTCRtpReceiver can be used by the application to get more fine grained control over the transmission and reception of MediaStreamTracks.

Channels are the smallest unit considered in the MediaStream specification. Channels are intended to be encoded together for transmission as, for instance, an RTP payload type. All of the channels that a codec needs to encode jointly MUST be in the same MediaStreamTrack and the codecs SHOULD be able to encode, or discard, all the channels in the track.

The concepts of an input and output to a given MediaStreamTrack apply in the case of MediaStreamTrack objects transmitted over the network as well. A MediaStreamTrack created by an RTCPeerConnection object (as described previously in this document) will take as input the data received from a remote peer. Similarly, a MediaStreamTrack from a local source, for instance a camera via [GETUSERMEDIA], will have an output that represents what is transmitted to a remote peer if the object is used with an RTCPeerConnection object.

The concept of duplicating MediaStream and MediaStreamTrack objects as described in [GETUSERMEDIA] is also applicable here. This feature can be used, for instance, in a video-conferencing scenario to display the local video from the user's camera and microphone in a local monitor, while only transmitting the audio to the remote peer (e.g. in response to the user using a "video mute" feature). Combining different MediaStreamTrack objects into new MediaStream objects is useful in certain situations.

Note

In this document, we only specify aspects of the following objects that are relevant when used along with an RTCPeerConnection. Please refer to the original definitions of the objects in the [GETUSERMEDIA] document for general information on using MediaStream and MediaStreamTrack.

10.2 MediaStream

10.2.1 id

The id attribute specified in MediaStream returns an id that is unique to this stream, so that streams can be recognized at the remote end of the RTCPeerConnection API.

When a MediaStream is created to represent a stream obtained from a remote peer, the id attribute is initialized from information provided by the remote source.

Note

The id of a MediaStream object is unique to the source of the stream, but that does not mean it is not possible to end up with duplicates. For example, the tracks of a locally generated stream could be sent from one user agent to a remote peer using RTCPeerConnection and then sent back to the original user agent in the same manner, in which case the original user agent will have multiple streams with the same id (the locally-generated one and the one received from the remote peer).

10.3 MediaStreamTrack

A MediaStreamTrack object's reference to its MediaStream in the non-local media source case (an RTP source, as is the case for MediaStreamTracks received over an RTCPeerConnection ) is always strong.

When an RTCPeerConnection receives data on an RTP source for the first time, it MUST update the muted state of the corresponding MediaStreamTrack with the value false.

When an RTCPeerConnection's RTP source is temporarily unable to receive media due to a loss of connection or if a mute signal has been received, it MUST update the muted state of the corresponding MediaStreamTrack with the value true. When media data is available again, the muted state MUST be updated with the value false.

Issue 10

The mute signal mentioned in the previous paragraph is yet to be defined.

The procedure update a track's muted state is specified in [GETUSERMEDIA].

When a track comes from a remote peer and the remote peer has permanently stopped sending data the ended event MUST be fired on the track, as specified in [GETUSERMEDIA].

Issue 11

How do you know when it has stopped? This seems like an SDP question, not a media-level question. (Suggestion: when the track is ended, either through port 0, or removing the a=msid attrib)

When a remote source is notified that a MediaStreamTrack, using the source, has ended [GETUSERMEDIA] the User Agent MAY choose to free resources allocated for the incoming stream, for instance turn off the decoder.

10.3.1 MediaTrackSupportedConstraints, MediaTrackCapabilities, MediaTrackConstraints and MediaTrackSettings

The basics of MediaTrackSupportedConstraints, MediaTrackCapabilites, MediaTrackConstraints and MediaTrackSettings is outlined in [GETUSERMEDIA]. However, the MediaTrackSettings for a MediaStreamTrack sourced by a RTCPeerConnection will only be populated to the extent that data is supplied by means of the remote RTCSessionDescription applied via setRemoteDescription and the actual RTP data. This means that certain settings, such as facingMode, echoCancellation , latency, deviceId and groupId, will always return null.

10.4 Isolated Media Streams

A MediaStream acquired using getUserMedia() is, by default, accessible to an application. This means that the application is able to access the contents of tracks, modify their content, and send that media to any peer it chooses.

WebRTC supports calling scenarios where media is sent to a specifically identified peer, without the contents of media streams being accessible to applications. This is enabled by use of the peerIdentity parameter to getUserMedia().

An application willingly relinquishes access to media by including a peerIdentity parameter in the MediaStreamConstraints. This attribute is set to a DOMString containing the identity of a specific peer.

The MediaStreamConstraints dictionary is expanded to include the peerIdentity parameter.

partial dictionary MediaStreamConstraints {
    DOMString peerIdentity;
};

Dictionary MediaStreamConstraints Members

peerIdentity of type DOMString

If set, peerIdentity isolates media from the application. Media can only be sent to the identified peer.

A user that is prompted to provide consent for access to a camera or microphone can be shown the value of the peerIdentity parameter, so that they can be informed that the consent is more narrowly restricted.

When the peerIdentity option is supplied to getUserMedia(), all of the MediaStreamTracks in the resulting MediaStream are isolated so that content is not accessible to any application. Isolated MediaStreamTracks can be used for two purposes:

A MediaStreamTrack that is added to another MediaStream remains isolated. When an isolated MediaStreamTrack is added to a MediaStream with a different peerIdentity, the MediaStream gets a combination of isolation restrictions. A MediaStream containing MediaStreamTrack instances with mixed isolation properties can be displayed, but cannot be sent using RTCPeerConnection.

Any peerIdentity property MUST be retained on cloned copies of MediaStreamTracks.

10.4.1 Extended MediaStreamTrack Properties

MediaStreamTrack is expanded to include an isolated attribute and a corresponding event. This allows an application to quickly and easily determine whether a track is accessible.

partial interface MediaStreamTrack {
    readonly attribute boolean      isolated;
             attribute EventHandler onisolationchange;
};
Attributes
isolated of type boolean, readonly

A MediaStreamTrack is isolated (and the corresponding isolated attribute set to true) when content is inaccessible to the owning document. This occurs as a result of setting the peerIdentity option. A track is also isolated if it comes from a cross origin source.

onisolationchange of type EventHandler

This event handler, of type isolationchange, is fired when the value of the isolated attribute changes.

10.4.2 Isolated Streams and RTCPeerConnection

A MediaStreamTrack with a peerIdentity option set can be added to any RTCPeerConnection. However, the content of an isolated track MUST NOT be transmitted unless all of the following constraints are met:

  • A MediaStreamTrack from a stream acquired using the peerIdentity option can be transmitted if the RTCPeerConnection has successfully validated the identity of the peer AND that identity is the same identity that was used in the peerIdentity option associated with the track. That is, the name attribute of the peerIdentity attribute of the RTCPeerConnection instance MUST match the value of the peerIdentity option passed to getUserMedia().

    Rules for matching identity are described in [RTCWEB-SECURITY-ARCH].

  • The peer has indicated that it will respect the isolation properties of streams. That is, a DTLS connection with a promise to respect stream confidentiality, as defined in [RTCWEB-ALPN] has been established.

Failing to meet these conditions means that no media can be sent for the affected MediaStreamTrack. Video MUST be replaced by black frames, audio MUST be replaced by silence, and equivalently information-free content MUST be provided for other media types.

Remotely sourced MediaStreamTracks MUST be isolated if they are received over a DTLS connection that has been negotiated with track isolation. This protects isolated media from the application in the receiving browser. These tracks MUST only be displayed to a user using the appropriate media element (e.g., <video> or <audio>).

Any MediaStreamTrack that has the peerIdentity option set causes all tracks sent using the same RTCPeerConnection to be isolated at the receiving peer. All DTLS connections created for a RTCPeerConnection with isolated local streams MUST be negotiated so that media remains isolated at the remote peer. This causes non-isolated media to become isolated at the receiving peer if any isolated tracks are added to the same RTCPeerConnection.

Note

Tracks that are not bound to a particular peerIdentity do not cause other streams to be isolated, these tracks simply do not have their content transmitted.

If a stream becomes isolated after initially being accessible, or an isolated stream is added to an active session, then media for that stream is replaced by information-free content (e.g., black frames or silence).

10.4.3 Protection Afforded by Media Isolation

Media isolation ensures that the content of a MediaStreamTrack is not accessible to web applications. However, to ensure that media with a peerIdentity option set can be sent to peers, some meta-information about the media will be exposed to applications.

Applications will be able to observe the parameters of the media that affect session negotiation and conversion into RTP. This includes the codecs that might be supported by the track, the bitrate, the number of packets, and the current settings that are set on the MediaStreamTrack.

In particular, the statistics that RTCPeerConnection records are not reduced in capability. New statistics that might compromise isolation MUST be avoided, or explicitly suppressed for isolated streams.

Most of these data are exposed to the network when the media is transmitted. Only the settings for the MediaStreamTrack present a new source of information. This can includes the frame rate and resolution of video tracks, the bandwidth of audio tracks, and other information about the source, which would not otherwise be revealed to a network observer. Since settings don't change at a high frequency or in response to changes in media content, settings only reveal limited reveal information about the content of a track. However, any setting that might change dynamically in response to the content of an isolated MediaStreamTrack MUST have changes suppressed.

11. Examples and Call Flows

This section is non-normative.

11.1 Simple Peer-to-peer Example

When two peers decide they are going to set up a connection to each other, they both go through these steps. The STUN/TURN server configuration describes a server they can use to get things like their public IP address or to set up NAT traversal. They also have to send data for the signaling channel to each other using the same out-of-band mechanism they used to establish that they were going to communicate in the first place.

Example 6
var signalingChannel = new SignalingChannel();
var configuration = { "iceServers": [{ "urls": "stuns:stun.example.org" }] };
var pc;

// call start() to initiate
function start() {
    pc = new RTCPeerConnection(configuration);

    // send any ice candidates to the other peer
    pc.onicecandidate = function (evt) {
        signalingChannel.send(JSON.stringify({ "candidate": evt.candidate }));
    };

    // let the "negotiationneeded" event trigger offer generation
    pc.onnegotiationneeded = function () {
        pc.createOffer().then(function (offer) {
            return pc.setLocalDescription(offer);
        })
        .then(function () {
            // send the offer to the other peer
            signalingChannel.send(JSON.stringify({ "desc": pc.localDescription }));
        })
        .catch(logError);
    };

    // once remote video track arrives, show it in the remote video element
    pc.ontrack = function (evt) {
        if (evt.track.kind === "video")
          remoteView.srcObject = evt.streams[0];
    };

    // get a local stream, show it in a self-view and add it to be sent
    navigator.mediaDevices.getUserMedia({ "audio": true, "video": true }, function (stream) {
        selfView.srcObject = stream;
        if (stream.getAudioTracks().length > 0)
            pc.addTrack(stream.getAudioTracks()[0], stream);
        if (stream.getVideoTracks().length > 0)
            pc.addTrack(stream.getVideoTracks()[0], stream);
    }, logError);
}

signalingChannel.onmessage = function (evt) {
    if (!pc)
        start();

    var message = JSON.parse(evt.data);
    if (message.desc) {
        var desc = message.desc;

        // if we get an offer, we need to reply with an answer
        if (desc.type == "offer") {
            pc.setRemoteDescription(desc).then(function () {
                return pc.createAnswer();
            })
            .then(function (answer) {
                return pc.setLocalDescription(answer);
            })
            .then(function () {
                signalingChannel.send(JSON.stringify({ "desc": pc.localDescription }));
            })
            .catch(logError);
        } else if (desc.type == "answer") {
            pc.setRemoteDescription(desc).catch(logError);
        } else {
            log("Unsupported SDP type. Your code may differ here.");
        }
    } else
        pc.addIceCandidate(message.candidate).catch(logError);
};

function logError(error) {
    log(error.name + ": " + error.message);
}

11.2 Simple Peer-to-peer Example with Warm-up

When two peers decide they are going to set up a connection to each other and want to have the ICE, DTLS, and media connections "warmed up" such that they are ready to send and receive media immediately, they both go through these steps.

Example 7
var signalingChannel = new SignalingChannel();
var configuration = { "iceServers": [{ "urls": "stuns:stun.example.org" }] };
var pc;
var audio = null;
var audioSendTrack = null;
var video = null;
var videoSendTrack = null;
var started = false;

// Call warmp() to warm-up ICE, DTLS, and media, but not send media yet.
function warmup(answerer) {
    pc = new RTCPeerConnection(configuration);
    if (!answerer) {
      audio = pc.addTransceiver("audio");
      video = pc.addTransceiver("video");
    }

    // send any ice candidates to the other peer
    pc.onicecandidate = function (evt) {
        signalingChannel.send(JSON.stringify({ "candidate": evt.candidate }));
    };

    // let the "negotiationneeded" event trigger offer generation
    pc.onnegotiationneeded = function () {
        pc.createOffer().then(function (offer) {
            return pc.setLocalDescription(offer);
        })
        .then(function () {
            // send the offer to the other peer
            signalingChannel.send(JSON.stringify({ "desc": pc.localDescription }));
        })
        .catch(logError);
    };

    // once remote video track arrives, show it in the remote video element
    pc.ontrack = function (evt) {
        if (evt.track.kind === "audio") {
          if (answerer) {
            audio = evt.transceiver;
            audio.setDirection("sendrecv");
            if (started && audioSendTrack) {
              audio.sender.replaceTrack(audioSendTrack);
            }
          }
        } else if (evt.track.kind === "video") {
          if (answerer) {
            video = evt.transceiver;
            video.setDirection("sendrecv");
            if (started && videoSendTrack) {
              video.sender.replaceTrack(audioSendTrack);
            }
          }
          remoteView.srcObject = evt.streams[0];
        }
    };

    // get a local stream, show it in a self-view and add it to be sent
    navigator.mediaDevices.getUserMedia({ "audio": true, "video": true }, function (stream) {
        selfView.srcObject = stream;
        if (stream.getAudioTracks().length > 0) {
          sendAudioTrack = stream.getVideoTracks()[0];
          if (started) {
            audio.sender.replaceTrack(sendAudioTrack);
          }
        }
        if (stream.getVideoTracks().length > 0) {
          sendVideoTrack = stream.getVideoTracks()[0];
          if (started) {
            video.sender.replaceTrack(sendVideoTrack);
          }
        }
    }, logError);
}

// Call start() to start sending media.
function start() {
  started = true;
  signalingChannel.send(JSON.stringify({ "start": true }));
}

signalingChannel.onmessage = function (evt) {
    if (!pc)
        warmup(true);

    var message = JSON.parse(evt.data);
    if (message.desc) {
        var desc = message.desc;

        // if we get an offer, we need to reply with an answer
        if (desc.type == "offer") {
            pc.setRemoteDescription(desc).then(function () {
                return pc.createAnswer();
            })
            .then(function (answer) {
                return pc.setLocalDescription(answer);
            })
            .then(function () {
                signalingChannel.send(JSON.stringify({ "desc": pc.localDescription }));
            })
            .catch(logError);
        } else
            pc.setRemoteDescription(desc).catch(logError);
    } else if (message.start) {
      started = true;
      if (audio && sendAudioTrack) {
        audio.sender.replaceTrack(sendVideoTrack);
      }
      if (video && sendVideoTrack) {
        video.sender.replaceTrack(sendVideoTrack);
      }
    } else
        pc.addIceCandidate(message.candidate).catch(logError);
};

function logError(error) {
    log(error.name + ": " + error.message);
}

11.3 Simple Peer-to-peer Example with media before signaling

The answerer may wish to send media in parallel with sending the answer, and the offerer may wish to render the media before the answer arrives.

Example 8
var signalingChannel = new SignalingChannel();
var configuration = { "iceServers": [{ "urls": "stuns:stun.example.org" }] };
var pc;

function findReceiver(mid) {
    for (var i = 0; i < receivers.length; i++) {
        var receiver = receiver[i];
        if (receiver.mid == videoSender.mid) {
             return receiver;
        }
    }
    return null;
}

// call start() to initiate
function start() {
    pc = new RTCPeerConnection(configuration);

    // send any ice candidates to the other peer
    pc.onicecandidate = function (evt) {
        signalingChannel.send(JSON.stringify({ "candidate": evt.candidate }));
    };

    // let the "negotiationneeded" event trigger offer generation
    pc.onnegotiationneeded = function () {
        pc.createOffer().then(function (offer) {
            return pc.setLocalDescription(offer);
        })
        .then(function () {
            // send the offer to the other peer
            signalingChannel.send(JSON.stringify({ "desc": pc.localDescription }));
        })
        .catch(logError);
    };

    // get a local stream, show it in a self-view and add it to be sent
    navigator.mediaDevices.getUserMedia({ "audio": true, "video": true }, function (stream) {
        selfView.srcObject = stream;
        var remoteStream = new MediaStream();
        if (stream.getAudioTracks().length > 0) {
            var audioSender = pc.addTrack(stream.getAudioTracks()[0], stream);
            remoteStream.addTrack(findReceiver(audioSender.mid).track);
        }
        if (stream.getVideoTracks().length > 0) {
            var videoSender = pc.addTrack(stream.getVideoTracks()[0], stream);
            remoteStream.addTrack(findReceiver(videoSender.mid).track);
        }
        // Render the media even before ontrack fires.
        removeView.srcObject = remoteStream;
    }, logError);
}

signalingChannel.onmessage = function (evt) {
    if (!pc)
        start();

    var message = JSON.parse(evt.data);
    if (message.desc) {
        var desc = message.desc;

        // if we get an offer, we need to reply with an answer
        if (desc.type == "offer") {
            pc.setRemoteDescription(desc).then(function () {
                return pc.createAnswer();
            })
            .then(function (answer) {
                return pc.setLocalDescription(answer);
            })
            .then(function () {
                signalingChannel.send(JSON.stringify({ "desc": pc.localDescription }));
            })
            .catch(logError);
        } else
            pc.setRemoteDescription(desc).catch(logError);
    } else
        pc.addIceCandidate(message.candidate).catch(logError);
};

function logError(error) {
    log(error.name + ": " + error.message);
}

11.4 Simple Simulcast Example

A client wants to send multiple RTP encodings (simulcast) to a server.

Example 9
var signalingChannel = new SignalingChannel();
var pc;

// call start() to initiate
function start() {
    pc = new RTCPeerConnection({});

    // let the "negotiationneeded" event trigger offer generation
    pc.onnegotiationneeded = function () {
        pc.createOffer().then(function (offer) {
            return pc.setLocalDescription(offer);
        })
        .then(function () {
            // send the offer to the other peer
            signalingChannel.send(JSON.stringify({ "desc": pc.localDescription }));
        })
        .catch(logError);
    };

    // get a local stream, show it in a self-view and add it to be sent
    navigator.mediaDevices.getUserMedia({ "audio": true, "video": true }, function (stream) {
        selfView.srcObject = stream;
        if (stream.getAudioTracks().length > 0)
            pc.addTransceiver(stream.getAudioTracks()[0], {direction: "sendonly"});
        if (stream.getVideoTracks().length > 0) {
            pc.addTransceiver(stream.getVideoTracks()[0], {
                direction: "sendonly",
                sendEncodings: [
                    {
                      rid: "f",
                    },
                    {
                      rid: "h",
                      scaleDownResolutionBy: 2.0
                    },
                    {
                      rid: "q",
                      scaleDownResolutionBy: 4.0
                    }
                ]
            });
        }
    }, logError);
}

signalingChannel.onmessage = function (evt) {
    var message = JSON.parse(evt.data);
    if (message.desc) {
        var desc = message.desc;
        pc.setRemoteDescription(message.desc).catch(logError);
    } else
        pc.addIceCandidate(message.candidate).catch(logError);
};

function logError(error) {
    log(error.name + ": " + error.message);
}

11.5 Advanced Peer-to-peer Example

This example shows the more complete functionality.

Issue 12

TODO

Example 10

      

    

    

      
11.6 Peer-to-peer Data Example

      

        
This example shows how to create a
        RTCDataChannel object and perform the offer/answer
        exchange required to connect the channel to the other peer. The
        RTCDataChannel is used in the context of a simple
        chat application and listeners are attached to monitor when the channel
        is ready, messages are received and when the channel is closed.

        
Example 11
var signalingChannel = new SignalingChannel();
var configuration = { "iceServers": [{ "urls": "stuns:stun.example.org" }] };
var pc;
var channel;

// call start(true) to initiate
function start(isInitiator) {
    pc = new RTCPeerConnection(configuration);

    // send any ice candidates to the other peer
    pc.onicecandidate = function (evt) {
signalingChannel.send(JSON.stringify({ "candidate": evt.candidate }));
    };

    // let the "negotiationneeded" event trigger offer generation
    pc.onnegotiationneeded = function () {
pc.createOffer().then(function (offer) {
    return pc.setLocalDescription(offer);
})
.then(function () {
    // send the offer to the other peer
    signalingChannel.send(JSON.stringify({ "desc": pc.localDescription }));
})
.catch(logError);
    };

    if (isInitiator) {
// create data channel and setup chat
channel = pc.createDataChannel("chat");
setupChat();
    } else {
// setup chat on incoming data channel
pc.ondatachannel = function (evt) {
    channel = evt.channel;
    setupChat();
};
    }
}

signalingChannel.onmessage = function (evt) {
    if (!pc)
start(false);

    var message = JSON.parse(evt.data);
    if (message.desc) {
var desc = message.desc;

// if we get an offer, we need to reply with an answer
if (desc.type == "offer") {
    pc.setRemoteDescription(desc).then(function () {
        return pc.createAnswer();
    })
    .then(function (answer) {
        return pc.setLocalDescription(answer);
    })
    .then(function () {
        signalingChannel.send(JSON.stringify({ "desc": pc.localDescription }));
    })
    .catch(logError);
} else
    pc.setRemoteDescription(desc).catch(logError);
    } else
pc.addIceCandidate(message.candidate).catch(logError);
};

function setupChat() {
    channel.onopen = function () {
// e.g. enable send button
enableChat(channel);
    };

    channel.onmessage = function (evt) {
showChatMessage(evt.data);
    };
}

function sendChatMessage(msg) {
    channel.send(msg);
}

function logError(error) {
    log(error.name + ": " + error.message);
}

      

    

    

      
11.7 Call Flow Browser to Browser

      
Issue 13
Editors' Note: This example flow needs to be discussed
      on the list and is likely wrong in many ways.

      
This shows an example of one possible call flow between two browsers.
      This does not show the procedure to get access to local media or every
      callback that gets fired but instead tries to reduce it down to only show
      the key events and messages.

      
A message sequence chart detailing a call flow between two browsers

    

    

      
11.8 DTMF Example

      
Examples assume that sender is an RTCRtpSender.

      
Sending the DTMF signal "1234" with 500 ms duration per tone:

      
Example 12
if (sender.dtmf) {
    var duration = 500;
    sender.dtmf.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):

      
Example 13
if (sender.dtmf) {
  sender.dtmf.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.dtmf.insertDTMF("1234");
} else
    log("DTMF function not available");

      
Send a 1-second "1" tone followed by a 2-second "2" tone:

      
Example 14
if (sender.dtmf) {
  sender.dtmf.ontonechange = function (e) {
      if (e.tone == "1")
          sender.dtmf.insertDTMF("2", 2000);
  };
  sender.dtmf.isertDTMF("1", 1000);
} else
    log("DTMF function not available");

      
It is always safe to append to the tone buffer. This example appends
      before any tone playout has started as well as during playout.

      
Example 15
if (sender.dtmf) {
  sender.dtmf.insertDTMF("123");
  // append more tones to the tone buffer before playout has begun
  sender.dtmf.insertDTMF(sender.toneBuffer + "456");

  sender.dtmf.ontonechange = function (e) {
      if (e.tone == "1")
          // append more tones when playout has begun
          sender.dtmf.insertDTMF(sender.toneBuffer + "789");
  };
} else
    log("DTMF function not available");

      
Send the DTMF signal "123" and abort after sending "2".

      
Example 16
if (sender.dtmf) {
  sender.dtmf.ontonechange = function (e) {
      if (e.tone == "2")
          // empty the buffer to not play any tone after "2"
          sender.dtmf.insertDTMF("");
  };
  sender.dtmf.insertDTMF("123");
} else
    log("DTMF function not available");

    

  

  

    
12. Event summary
This section is non-normative.

    
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).
          
        

        

          message
          MessageEvent
          [webmessaging]
          A message was successfully received.
        

        

          bufferedamountlow
          Event
          The RTCDataChannel object's
          bufferedAmount
          decreases from above its bufferedAmountLowThreshold to less than
          or equal to its bufferedAmountLowThreshold.
        

        

          error
          ErrorEvent
          Any error occured from the data channel.
        

        

          close
          Event
          
            The RTCDataChannel object's underlying data
            transport has bee closed.
          
        

      
    

    
The following events fire on RTCPeerConnection
    objects:

    

      
        

          Event name
          Interface
          Fired when...
        

      
      
        

          connecting
          Event
          
            
Issue 14
TODO

          		
        

        

          track
          RTCTrackEvent
          
            A new incoming MediaStreamTrack has been created, and
            an associated RTCRtpReceiver has been added to the
            set of receivers.
          
        

        

          negotiationneeded
          Event
          The browser wishes to inform the application that session
          negotiation needs to be done (i.e. a createOffer call followed by
          setLocalDescription).
        

        

          signalingstatechange
          Event
          
            The signaling state has changed. This state change is the
            result of either setLocalDescription or
            setRemoteDescription being invoked.
          
        

        

          iceconnectionstatechange
          Event
          
            The RTCPeerConnection's ICE connection state
            has changed.
          
        

        

          icegatheringstatechange
          Event
          
            The RTCPeerConnection's ICE gathering state has
            changed.
          
        

        

          icecandidate
          RTCPeerConnectionIceEvent
          A new RTCIceCandidate is made available to
          the script.
        

        

          connectionstatechange
          Event
          
            The RTCPeerConnection connectionState has changed.
          
        

        

          icecandidateerror
          RTCPeerConnectionIceErrorEvent
          A failure occured when gathering ICE candidates.
        

        

          datachannel
          RTCDataChannelEvent
          A new RTCDataChannel is dispatched to the
          script in response to the other peer creating a channel.
        

        

          isolationchange
          Event
          A new Event is dispatched to the script when
          the isolated attribute on a MediaStreamTrack
          changes.
        

      
    

    
The following events fire on RTCDTMFSender
    objects:

    

      
        

          Event name
          Interface
          Fired when...
        

      
      
        

          tonechange
          RTCDTMFToneChangeEvent
          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 RTCIceTransport
    objects:

    

      
        

          Event name
          Interface
          Fired when...
        

      
      
        

          statechange
          Event
          The RTCIceTransport state changes.
        

        

          gatheringstatechange
          Event
          The RTCIceTransport gathering state
          changes.
        

        

          selectedcandidatepairchange
          Event
          The RTCIceTransport's selected candidate pair
          changes.
        

      
    

    
The following events fire on RTCDtlsTransport
    objects:

    

      
        

          Event name
          Interface
          Fired when...
        

      
      
        

          statechange
          Event
          The RTCDtlsTransport state changes.
        

      
    

  

  

    
13. Privacy and Security Considerations
This section is non-normative.

    
This section is non-normative; it specifies no new behaviour, but
    instead summarizes information already present in other parts of the
    specification. The overall security considerations of the general set of
    APIs and protocols used in WebRTC are described in
    [RTCWEB-SECURITY-ARCH].

    

      
13.1 Impact on same origin policy

      
This document extends the Web platform with the ability to set up real
      time, direct communication between browsers and other devices, including
      other browsers.

      
This means that data and media can be shared between applications
      running in different browsers, or between an application running in the
      same browser and something that is not a browser, something that is an
      extension to the usual barriers in the Web model against sending data
      between entities with different origins.

      
The WebRTC specification provides no user prompts or chrome indicators
      for communication; it assumes that once the Web page has been allowed to
      access media, it is free to share that media with other entities as it
      chooses. Peer-to-peer exchanges of data view WebRTC datachannels can thus
      occur without any user explicit consent or involvement, similarly as a
      server-mediated exchange (e.g. via Web Sockets) could occur without user
      involvement.

      
The peerIdentity mechanism loads and executes
      JavaScript code from a third-party server acting as an identity provider.
      That code is executed in a separate JavaScript realm and does not affect
      the protections afforded by the same origin policy.

    

    

      
13.2 Revealing IP addresses

      
Even without WebRTC, the Web server providing a Web application will
      know the public IP address to which the application is delivered. Setting
      up communications exposes additional information about the
      browser’s network context to the web application, and may include
      the set of (possibly private) IP addresses available to the browser for
      WebRTC use. Some of this information has to be passed to the
      corresponding party to enable the establishment of a communication
      session.

      
Revealing IP addresses can leak location and means of connection; this
      can be sensitive. Depending on the network environment, it can also
      increase the fingerprinting surface and create persistent cross-origin
      state that cannot easily be cleared by the user.

      
A connection will always reveal the IP addresses proposed for
      communication to the corresponding party. The application can limit this
      exposure by choosing not to use certain addresses using the settings
      exposed by the RTCIceTransportPolicy dictionary, and by using
      relays (for instance TURN servers) rather than direct connections between
      participants. One will normally assume that the IP address of TURN
      servers is not sensitive information. These choices can for instance be
      made by the application based on whether the user has indicated consent
      to start a media connection with the other party.

      
Mitigating the exposure of IP addresses to the application itself
      requires limiting the IP addresses that can be used, which will impact
      the ability to communicate on the most direct path between endpoints.
      Browsers are encouraged to provide appropriate controls for deciding
      which IP addresses are made available to applications, based on the
      security posture desired by the user. The choice of which addresses to
      expose is controlled by local policy (see [RTCWEB-IP-HANDLING] for
      details).

    

    

      
13.3 Impact on local network

      
Since the browser is an active platform executing in a trusted network
      environment (inside the firewall), it is important to limit the damage
      that the browser can do to other elements on the local network, and it is
      important to protect data from interception, manipulation and
      modification by untrusted participants.

      
Mitigations include:

      
      
These measures are specified in the relevant IETF documents.

    

    

      
13.4 Confidentiality of Communications

      
The fact that communication is taking place cannot be hidden from
      adversaries that can observe the network, so this has to be regarded as
      public information.

      
A mechanism, peerIdentity, is provided that gives
      Javascript the option of requesting media that the same javascript cannot
      access, but can only be sent to certain other entities.

    

    

      
13.5 Persistent information exposed by WebRTC

      
As described above, the list of IP addresses exposed by the WebRTC API
      can be used as a persistent cross-origin state.

      
Beyond IP addresses, the WebRTC API exposes information about the
      underlying media system via the RTCRtpSender.getCapabilities
      and RTCRtpReceiver.getCapabilities methods, including
      detailed and ordered information about the codecs that the system is able
      to produce and consume. A subset of that information is likely to be
      represented in the SDP session descriptions generated, exposed and
      transmitted during session
      negotiation. That information is in most cases persistent across time
      and origins, and increases the fingerprint surface of a given device.

      
When establishing DTLS connections, the WebRTC API can generate
      certificates that can be persisted by the application (e.g. in
      IndexedDB). These certificates are not shared across origins, and get
      cleared when persistent storage is cleared for the origin.

    

  

  

    
14. Change Log

    
This section will be removed before publication.

    
    
    
Changes since February 15, 2016

    
    
      
  1. [#475] Definition of Active for an RTCRtpReceiver
    2. 
      
  2. [#500] Reserve and use RangeError for scaleResolutionDownBy <
      1.0
    3. 
      
  3. [#504] Add getParameters() method to RTCRtpReceiver
    4. 
      
  4. [#509] RID unmodifiable in setParameters()
    5. 
      
  5. [#510] Gather spec text about the ICE Agent at one
      place
    6. 
      
  6. [#512] Use 'connection' as configuration target instead of User
      Agent
    7. 
      
  7. [#505] Add activateReceiver method to RTCRtpTransceiver
    8. 
      
  8. [#516] Support for DTMF tones A-D
    9. 
      
  9. [#499] Certificate API: add getAlgorithm method
    10. 
      
  10. [#507] Make the definition of addIceCandidate() more explicit
    11. 
      
  11. [#525] Add STUN Error Code reference
    12. 
      
  12. [#524] Add error codes reference (RTCPeerConnectionIceErrorEvent)
      
    13. 
      
  13. [#522] Let setting ice candidate pool size trigger start of
      gathering
    14. 
      
  14. [#519] Relation between local track and outgoing encoding
    15. 
      
  15. [#520] Add text about 'remote sources' and how they are stopped
    16. 
      
  16. [#527] Enable trickling of end-of-candidates through addIceCandidate
      
    17. 
      
  17. [#544] Remove "public" from ice transport policy
    18. 
      
  18. [#547] Datachannel label and protocol are USVString
    19. 
      
  19. [#552] Never close the RTCPeerConnection if setting a local/remote
      description fails
    20. 
      
  20. [#535] Update MID to be random values when not received in offer
    21. 
      
  21. [#553] Move 'closed' state from RTCSignalingState to
      RTCPeerConnectionState
    22. 
      
  22. [#557] Splitting apart RTCIceConnectionState and
      RTCIceTransportState
    23. 
      
  23. [#560] Changing from callback interface to dictionary for
      RTCIdentityProvider
    24. 
      
  24. [#574] Make RTCSessionDescription readonly, and createOffer return
      dictionary
    25. 
      
  25. [#577] Make RtpSender.track nullable
    26. 
      
  26. [#587] Defining how track settings are set for remote tracks
    27. 
      
  27. [#603] Add closed state and same state checks to update ice
      connection/gathering state steps
    28. 
      
  28. [#604] ReplaceTrack: Use sender's transceiver to determine if a
      'simple track swap' is enough
    29. 
      
  29. [#606] RTCIceCandidate: Use nullable members in init dictionary to
      describe constructor behavior
    30. 
      
  30. [#466] Use an enum to describe directionality of RTP Stream
    31. 
      
  31. [#602] addTransceiver(): Throw a TypeError on a bogus track kind
    32. 
      
  32. [#610] Server cannot be reached - Issues with IPv6
    33. 
      
  33. [#611] Clarify ICE consent freshness feedback
    34. 
      
  34. [#618] Fix RTCPeerConnection legacy overloads
    35. 
      
  35. [#620] RTCRtpTransceiver: add setDirection and readonly direction
      attribute
    36. 
      
  36. [#625] Unifiy DTMF time with rtcweb WG
    37. 
      
  37. [#630] Add ICE candidate type references
    38. 
      
  38. [#635] pc.addTrack: Add kind check when reusing a sender and skip
      early returns
    39. 
      
  39. [#636] replaceTrack: Use 'transceiver kind' instead of track.kind
      (track may be null)
    40. 
    

    
Changes since January 26, 2016

    
    
      
  1. [#485] Update SOTD as the document is now quite stable and the group
      is looking for wide review
    2. 
      
  2. [#468, #335] Replace DOMError with DOMException
    3. 
      
  3. [#472, #319] Update error reports to align with existing DOM
      Errors
    4. 
      
  4. [#491, #479] Specify error when rejecting invalid SDP changes
    5. 
      
  5. [#462] Add PeerConnection.activateSender() and update early media
      example
    6. 
      
  6. [#434] Change setParameters call to be Async
    7. 
    

    
Changes since December 22, 2015

    
    
      
  1. [#179, #439] Document IP address leakage in RTCIceCandidate
    2. 
      
  2. [#439] Complete security considerations based on security
      questionnaire and IP address discussions #439
    3. 
      
  3. [#446] Non-nullable RTCTrackEvent args means Init dict members are
      required
    4. 
      
  4. [#449] Clarify flow of SDP exchanges (Update simple p2p example)
    5. 
      
  5. [#451] Clean up event handler attribute descriptions
    6. 
      
  6. [#452, #438] Make replaceTrack() handle "not sending yet" case
    7. 
      
  7. [#454] Add contributing source voice activity flag
    8. 
      
  8. [#455, #439] Add references to parsing stun/turn URLs section
    9. 
      
  9. [#456, #338] SDP changes between the createOffer and
      setLocalDescription (add JSEP reference)
    10. 
      
  10. [#459] Add non-normative ICE state transitions
    11. 
      
  11. [#460, #461] getRemoteCertificates() behavior in "new" and
      "connecting" states
    12. 
      
  12. [#465, #140] Use ErrorEvent as interface for events emitted by
      RTCDataChannel.onerror
    13. 
      
  13. [#469, #382, #373] Reject changes to peerIdentity and certificates in
      setConfiguration
    14. 
      
  14. [#474, #406] Define RTCIceTransport.component when RTP/RTCP mux is in
      use
    15. 
    

    
Changes since November 23, 2015

    
    
      
  1. [#353] Plan X: Add an API for using RID to do simulcast
    2. 
      
  2. [#365] Adding an accessor for the browser-configured ICE servers
    3. 
      
  3. [#398] Make RtpTransceiver.mid nullable and remove RtpSender.mid and
      RtpReceiver.mid
    4. 
      
  4. [#402, #391] Remove requirement about DTMF tones A-D
    5. 
      
  5. [#403, #377] Use positive values for AudioLevel
    6. 
      
  6. [#401, #267] Add bitrate definition
    7. 
      
  7. [#404] Remove 'Events on MediaStream' section (duplicates new text in
      Media Capture spec)
    8. 
      
  8. [#410, #328] Make RTCBundlePolicy Enum section normative
    9. 
      
  9. [#411, #408] Clarify component for IceTransport when RTP/RTCP mux is
      used
    10. 
      
  10. [#414] Define ReSpec processor for cross-reference to JSEP
    11. 
      
  11. [#418] Make degradationPreference per-sender instead of
      per-encoding
    12. 
      
  12. [#416] RTCRtpSender.replaceTrack() fixes (e.g. handle closed
      RTCPeerConnection)
    13. 
      
  13. [#421] Require sdp in RTCSessionDescription{,Init}
    14. 
      
  14. [#422] Remove confusing paragraph on fourth party interception
    15. 
      
  15. [#423] Add specific references to JSEP where possible
    16. 
      
  16. [#428] Don't create a default stream in 'dispatch a receiver'
      steps
    17. 
      
  17. [#429] Adding expires attribute to generateCertificate
    18. 
      
  18. [#430] Add maxFramerate knob for simulcast
    19. 
      
  19. [#432] Update RTCIceTransportPolicy
    20. 
      
  20. [#433] Use unsigned long ssrc in stats
    21. 
      
  21. [#424] Editorial: Distinguish states from their attribute
      representation
    22. 
    

    
Changes since October 6, 2015

    
    
      
  1. [#325] Adding additional members to RTCIceCandidate dictionary
    2. 
      
  2. [#327] Adding sha-256 to the certificate management options for
      RSA
    3. 
      
  3. [#342] Using DOMTimestamp for RTCCertificate::expires
    4. 
      
  4. [#293] Add RTCRtpTransceiver and PeerConnection.addMedia
    5. 
      
  5. [#366, #343] Use RTCDegradationPreference
    6. 
      
  6. [#374] Throw on too long label/protocol in createDataChannel()
    7. 
      
  7. [#266] Tidy up setLocal/RemoteDescription processing model
    8. 
      
  8. [#361] Adding setCodecPreferences to RTCRtpTransceiver
    9. 
      
  9. [#371] Add RtcpMuxPolicy
    10. 
      
  10. [#385, #312] Don't invoke public API in legacy function section
    11. 
      
  11. [#394, #393] don't throw on empty iceServers list
    12. 
    

    
Changes since September 22, 2015

    
    
      
  1. [#289, #153] Add way to set size of ICE candidate pool
    2. 
      
  2. [#256] Fix prose on getStats() wo/selector + move type check to sync
      section
    3. 
      
  3. [#242] Remove SyntaxError on malformed ICE candidate
    4. 
      
  4. [#284] Add icecandidateerror event for indicating ICE gathering
      errors
    5. 
      
  5. [#298] Add support for codec reordering and removal in
      RtpParameters
    6. 
      
  6. [#311] Fixing syntax for required RTCCertificate arguments
    7. 
      
  7. [#280] Add extra IceTransport read-only attributes and methods
    8. 
      
  8. [#291] Add PeerConnection.connectionState
    9. 
      
  9. [#300, #4, #6, #276] Add API to get SSRC and audio levels
    10. 
      
  10. [#301] Fix RTCStatsReport with object and maplike instead of
      getter
    11. 
      
  11. [#302] (Partly) removing interface use for RTCSessionDescription and
      RTCIceCandidate
    12. 
      
  12. [#314, #299] Update the operations queue to handle promises and
      closed signalling
    13. 
      
  13. [#273] Add a bunch of fields to RtpParameters and
      RtpEncodingParameters
    14. 
    

    
Changes since June 11, 2015

    
    
      
  1. [#234] Add RTCRtpParameters, RTCRtpSender.getParameters, and
      RTCRtpSender.setParameters
    2. 
      
  2. [#225] Support for pending and current SDP
    3. 
      
  3. [#229] Removing the weird optionality from RTCSessionDescription and
      its constructor.
    4. 
      
  4. [#235] Modernize getStats() with promises
    5. 
      
  5. [#243] Mark candidate property of RTCIceCandidateInit required
    6. 
      
  6. [#248] Fix error handling for certificate management
    7. 
      
  7. [#259] Change type of RtpEncodingParameters.priority to an enum
    8. 
      
  8. [#21, #262] Sort out 2119 MUSTs and SHOULDs
    9. 
      
  9. [#268] Add RtpEncodingParameters.maxBitrate
    10. 
      
  10. [#241] Add RtpSender.transport, RtpReceiver.transport,
      RTCDtlsTransport, RTCIceTransport, etc
    11. 
      
  11. [#224, #261] Sort out when responding PeerConnection reaches
      iceConnetionState completed
    12. 
      
  12. [#303] Replace track without renegotiation
    13. 
      
  13. [#269] Add RTCRtpSender.getCapabilities and
      RTCRtpReceiver.getCapabilities
    14. 
    

    
Changes since March 6, 2015

    
    
      
  1. [PR #167] Removed RTCPeerConnection.createDTMFSender and added
      RTCRtpSender.dtmf, along with corresponding examples.
    2. 
      
  2. [PR #184] RTCPeerConnection will NOT connect unless identity is
      verified.
    3. 
      
  3. [PR #27] Documenting practice with candidate events
    4. 
      
  4. [PR #203] Rewrote mitigations text for security considerations
      section
    5. 
      
  5. [PR #192] Added support for auth tokens. Fixes #190
    6. 
      
  6. [PR #207] Update ice config examples to use multiple urls and *s
      schemes
    7. 
      
  7. [PR #210] Optional RTCConfiguration in RTCPC constructor
    8. 
      
  8. [PR #171] Add RTCAnswerOptions (with common RTCOfferAnswerOptions
      dictionary)
    9. 
      
  9. [PR #178] Identity provider interface redesign
    10. 
      
  10. [PR #193] Add .mid property to sender/receiver. Fixes #191
    11. 
      
  11. [PR #218] Enqueue addIceCandidate
    12. 
      
  12. [PR #213 (1)] Rename updateIce() to setConfiguration()
    13. 
      
  13. [PR #213 (2)] Make RTCPeerConnection.setConfiguration() replace the
      existing configuration
    14. 
      
  14. [PR #214] Certificate management API (Bug 21880)
    15. 
      
  15. [PR #220] Clarify muted state (proposed fix for issue #139)
    16. 
      
  16. [PR #221] Define when RTCRtpReceivers are created and dispatced
      (issue #198)
    17. 
      
  17. [PR #215] Adding expires attribute to certificate management
    18. 
      
  18. [PR #233] Add a "bufferedamountlow" event
    19. 
    

    
Changes since December 5, 2014

    
    
      
  1. Properly define the negotiationneeded event, and its interactions
      with other API calls.
    2. 
      
  2. Add support for RTCRtpSender and RTCRtpReceiver.
    3. 
      
  3. Update misleading local/RemoteDescription attribute text.
    4. 
      
  4. Add RTCBundlePolicy.
    5. 
      
  5. All callback-based methods have been moved to a legacy section, and
      replaced by same-named overloads using Promises instead.
    6. 
      
  6. [PR #194] Added first version of Security Considerations (more work
      needed)
    7. 
      
  7. Updated identity provider structure.
    8. 
    

    
Changes since June 4, 2014

    
    
      
  1. Bug 25724: Allow garbage collection of closed PeerConnections
    2. 
      
  2. Bug 27214: Add onicegatheringstatechange event
    3. 
      
  3. Bug 26644: Fixing end of candidates event
    4. 
    

    
Changes since April 10, 2014

    
    
      
  1. Bug 25774: Mixed isolation
    2. 
    

    
Changes since April 10, 2014

    
    
      
  1. Bug 25855: Clarification about conformance requirements phrased as
      algorithms
    2. 
      
  2. Bug 25892: SignalingStateChange event should be fired only if there
      is a change in signaling state.
    3. 
      
  3. Bug 25152: createObjectURL used in examples is no longer supported by
      Media Capture and Streams.
    4. 
      
  4. Bug 25976: DTMFSender.insertDTMF steps should validate the values of
      duration and interToneGap.
    5. 
      
  5. Bug 25189: Mandatory errorCallback is missing in examples for
      getStats.
    6. 
      
  6. Bug 25840: Creating DataChannel with same label.
    7. 
      
  7. Updated comment above example ice state transitions (discussed in Bug
      25257).
    8. 
      
  8. Updated insertDTMF() algorithm to ignore unrecognized characters (as
      discussed in bug 25977).
    9. 
      
  9. Made formatting of references to ice connection state
      consistent.
    10. 
      
  10. Made insertDTMF() throw on unrecognized characters (used to
      ignore).
    11. 
      
  11. Removed requestIdentity from RTCConfiguration and
      RTCOfferAnswerOptions. Removed RTCOfferAnswerOptions as a result.
    12. 
      
  12. Adding isolated property and associated event to
      MediaStreamTrack.
    13. 
    

    
Changes since March 21, 2014

    
    
      
  1. Changes to identity-related text:
        
      
          
    • Removed noaccess constraint
      • 
          
    • Add the ability to peerIdentity constrain RTCPeerConnection,
          which limits communication to a single peer
      • 
          
    • Change the way that the browser communicates with IdP to a
          message channel
          (http://www.w3.org/TR/webmessaging/#message-channels)
      • 
          
    • Improved error feedback from IdP interactions (added new events
          with more detailed context)
      • 
          
    • Changed the way that an IdP is able to request user login
          (LOGINNEEDED message)
      • 
        
    
      
    2. 
      
  2. Bug 25155: maxRetransmitTime is not the name of the SCTP concept it
      points to.
    3. 
    

    
Changes since January 27, 2014

    
    
      
  1. Refined identity assertion generation and validation.
    2. 
      
  2. Default DTMF gap changed from 50 to 70 ms.
    3. 
      
  3. Bug 24875: Examples in the WebRTC spec are not updated As per the
      modified API.
    4. 
    

    
Changes since August 30, 2013

    
    
      
  1. Make RTCPeerConnection close method be idempotent.
    2. 
      
  2. Clarified ICE server configuration could contain URI types other than
      STUN and TURN.
    3. 
      
  3. Changed the DTMF timing values.
    4. 
      
  4. Allow offerToReceiveAudio/video indicate number of streams to
      offer.
    5. 
      
  5. ACTION-98: Added text about clamping of maxRetransmitTime and
      maxRetransmits.
    6. 
      
  6. ACTION-88: Removed nullable types from dictionaries (added attribute
      default values for attributes that would be left uninitialized without
      the init dictionary present.
    7. 
      
  7. InvalidMediaStreamTrackError changed to InvalidParameter.
    8. 
      
  8. Fire NetworkError when the data transport is closed with an
      error.
    9. 
      
  9. Add an exception for data channel with trying to use existing
      code.
    10. 
      
  10. Change maxRetransmits to be an unsigned type.
    11. 
      
  11. Clarify state changes when ICE restarts.
    12. 
      
  12. Added InvalidStateError exception for operations on a
      RTCPeerConnection that is closed.
    13. 
      
  13. Major changes to Identity Proxy section.
    14. 
      
  14. (ACTION: 95) Moved IceTransports (constraint) to RTCConfiguration
      dictionary.
    15. 
      
  15. (ACTION: 95) Introduced RTCOfferAnswerOptions and RTCOfferOptions
      dictionaries.
    16. 
      
  16. (ACTION: 95) Removed constraints argument from addStream() (and
      removed IANA Constraints section).
    17. 
      
  17. Added validation of the RTCConfiguration dictionary argument(s).
    18. 
      
  18. Added getConfiguration() on RTCPeerConnection.
    19. 
    

    
Changes since June 3, 2013

    
    
      
  1. Removed synchronous section left-overs.
    2. 
      
  2. RTCIceServer now accepts multiple URLs.
    3. 
      
  3. Redefined the meaning of negotiated for DataChannel.
    4. 
      
  4. Made iceServers a sequence (instead of an Array).
    5. 
      
  5. Updated error reporting (to use DOMError and camel cased names).
    6. 
      
  6. Added success and failure callbacks to addIceCandidate().
    7. 
      
  7. Made local/remoteDescription attributes nullable.
    8. 
      
  8. Added username member to RTCIceServer dictionary.
    9. 
    

    
Changes since March 22, 2013

    
    
      
  1. Added IceRestart constraint.
    2. 
      
  2. Big updates on DataChannel API to use new channel setup
      procedures.
    3. 
    

    
Changes since Feb 22, 2013

    
    
      
  1. Example review: Updated DTMF and Stats examples. Added text about
      when to fire "negotiationneeded" event to align with examples.
    2. 
      
  2. Updated RTCPeerConnection state machine. Added a shared processing
      model for setLocalDescription()/setRemoteDescription().
    3. 
      
  3. Updated simple callflow to match the current API.
    4. 
    

    
Changes since Jan 16, 2013

    
    
      
  1. Initial import of Statistics API to version 2.
    2. 
      
  2. Integration of Statistics API version 2.5 started.
    3. 
      
  3. Updated Statistics API to match Boston/list discussions.
    4. 
      
  4. Extracted API extensions introduced by features, such as the P2P Data
      API, from the RTCPeerConnection API.
    5. 
      
  5. Updated DTMF algorithm to dispatch an event when insertDTMF() is
      called with an empty string to cancel future tones.
    6. 
      
  6. Updated DTMF algorithm to not cancel and reschedule if a playout task
      is running (only update toneBuffer and other values).
    7. 
    

    
Changes since Dec 12, 2012

    
    
      
  1. Changed AudioMediaStreamTrack to RTCDTMFSender and gave it its own
      section. Updated text to reflect most recent agreements. Also added
      examples section.
    2. 
      
  2. Replaced the localStreams and remoteStreams attributes with functions
      returning sequences of MediaStream objects.
    3. 
      
  3. Added spec text for attributes and methods adopted from the WebSocket
      interface.
    4. 
      
  4. Changed the state ENUMs and transition diagrams.
    5. 
      
  5. Aligned the data channel processing model a bit more with WebSockets
      (mainly closing the underlying transport).
    6. 
    

    
Changes since Nov 13, 2012

    
    
      
  1. Made some clarifications as to how operation queuing works, and fixed
      a few errors with the error handling description.
    2. 
      
  2. Introduced new representation of tracks in a stream (removed
      MediaStreamTrackList). Added algorithm for creating a track to represent
      an incoming network media component.
    3. 
      
  3. Renamed MediaStream.label to MediaStream.id (the definition needs
      some more work).
    4. 
    

    
Changes since Nov 03, 2012

    
    
      
  1. Added text describing the queuing mechanism for
      RTCPeerConnection.
    2. 
      
  2. Updated simple P2P example to include all mandatory (error)
      callbacks.
    3. 
      
  3. Updated P2P data example to include all mandatory (error) callbacks.
      Also added some missing RTC prefixes.
    4. 
    

    
Changes since Oct 19, 2012

    
    
      
  1. Clarified how createOffer() and createAnswer() use their
      callbacks.
    2. 
      
  2. Made all failure callbacks mandatory.
    3. 
      
  3. Added error object types, general error handling principles, and
      rules for when errors should be thrown.
    4. 
    

    
Changes since Sept 23, 2012

    
    
      
  1. Restructured the document layout and created separate sections for
      features like Peer-to-peer Data API, Statistics and Identity.
    2. 
    

    
Changes since Aug 16, 2012

    
    
      
  1. Replaced stringifier with serializer on RTCSessionDescription and
      RTCIceCandidate (used when JSON.stringify() is called).
    2. 
      
  2. Removed offer and createProvisionalAnswer arguments from the
      createAnswer() method.
    3. 
      
  3. Removed restart argument from the updateIce() method.
    4. 
      
  4. Made RTCDataChannel an EventTarget
    5. 
      
  5. Updated simple RTCPeerConnection example to match spec changes.
    6. 
      
  6. Added section about RTCDataChannel garbage collection.
    7. 
      
  7. Added stuff for identity proxy.
    8. 
      
  8. Added stuff for stats.
    9. 
      
  9. Added stuff peer and ice state reporting.
    10. 
      
  10. Minor changes to sequence diagrams.
    11. 
      
  11. Added a more complete RTCDataChannel example
    12. 
      
  12. Various fixes from Dan's Idp API review.
    13. 
      
  13. Patched the Stats API.
    14. 
    

    
Changes since Aug 13, 2012

    
    
      
  1. Made the RTCSessionDescription and RTCIceCandidate constructors take
      dictionaries instead of a strings. Also added detailed stringifier
      algorithm.
    2. 
      
  2. Went through the list of issues (issue numbers are only valid with
      HEAD at fcda53c460). Closed (fixed/wontfix): 1, 8, 10, 13, 14, 16, 18,
      19, 22, 23, 24. Converted to notes: 4, 12. Updated: 9.
    3. 
      
  3. Incorporate changes
      proposed by Li Li.
      
    4. 
      
  4. Use an enum for DataChannelState and fix IDLs where using an optional
      argument also requires all previous optional arguments to have a default
      value.
    5. 
    

    
Changes since Jul 20, 2012

    
    
      
  1. Added RTC Prefix to names (including the notes below).
    2. 
      
  2. Moved to new definition of configuration and ice servers object.
    3. 
      
  3. Added correlating lines to candidate structure.
    4. 
      
  4. Converted setLocalDescription and setRemoteDescription to be
      asynchronous.
    5. 
      
  5. Added call flows.
    6. 
    

    
Changes since Jul 13, 2012

    
    
      
  1. Removed peer attribute from RTCPeerConnectionIceEvent (duplicates
      functionality of Event.target attribute).
    2. 
      
  2. Removed RTCIceCandidateCallback (no longer used).
    3. 
      
  3. Removed RTCPeerConnectionEvent (we use a simple event instead).
    4. 
      
  4. Removed RTCSdpType argument from setLocalDescription() and
      setRemoteDescription(). Updated simple example to match.
    5. 
    

    
Changes since May 28, 2012

    
    
      
  1. Changed names to use RTC Prefix.
    2. 
      
  2. Changed the data structure used to pass in STUN and TURN servers in
      configuration.
    3. 
      
  3. Updated simple RTCPeerConnection example (RTCPeerConnection
      constructor arguments; use icecandidate event).
    4. 
      
  4. Initial import of new Data API.
    5. 
      
  5. Removed some left-overs from the old Data Stream API.
    6. 
      
  6. Renamed "underlying data channel" to "underlying data transport".
      Fixed closing procedures. Fixed some typos.
    7. 
    

    
Changes since April 27, 2012

    
    
      
  1. Major rewrite of RTCPeerConnection section to line up with IETF JSEP
      draft.
    2. 
      
  2. Added simple RTCPeerConnection example. Initial update of
      RTCSessionDescription and RTCIceCandidate to support serialization and
      construction.
    3. 
    

    
Changes since 21 April 2012

    
    
      
  1. Moved MediaStream and related definitions to getUserMedia.
    2. 
      
  2. Removed section "Obtaining local multimedia content".
    3. 
      
  3. Updated getUserMedia() calls in examples (changes in Media Capture TF
      spec).
    4. 
      
  4. Introduced MediaStreamTrackList interface with support for adding and
      removing tracks.
    5. 
      
  5. Updated the algorithm that is run when RTCPeerConnection receives a
      stream (create new stream when negotiated instead of when data
      arrives).
    6. 
    

    
Changes since 12 January 2012

    
    
      
  1. Clarified the relation of Stream, Track, and Channel.
    2. 
    

    
Changes since 17 October 2011

    
    
      
  1. Tweak the introduction text and add a reference to the IETF RTCWEB
      group.
    2. 
      
  2. Changed the first argument to getUserMedia to be an object.
    3. 
      
  3. Added a MediaStreamHints object as a second argument to
      RTCPeerConnection.addStream.
    4. 
      
  4. Added AudioMediaStreamTrack class and DTMF interface.
    5. 
    

    
Changes since 23 August 2011

    
    
      
  1. Separated the SDP and ICE Agent into separate agents and added
      explicit state attributes for each.
    2. 
      
  2. Removed the send method from PeerConenction and associated callback
      function.
    3. 
      
  3. Modified MediaStream() constructor to take a list of MediaStreamTrack
      objects instead of a MediaStream. Removed text about MediaStream parent
      and child relationship.
    4. 
      
  4. Added abstract.
    5. 
      
  5. Moved a few paragraphs from the MediaStreamTrack.label section to the
      MediaStream.label section (where they belong).
    6. 
      
  6. Split MediaStream.tracks into MediaStream.audioTracks and
      MediaStream.videoTracks.
    7. 
      
  7. Removed a sentence that implied that track access is limited to
      LocalMediaStream.
    8. 
      
  8. Updated a few getUserMedia()-examples to use MediaStreamOptions.
    9. 
      
  9. Replaced calls to URL.getObjectURL() with URL.createObjectURL() in
      example code.
    10. 
      
  10. Fixed some broken getUserMedia() links.
    11. 
      
  11. Introduced state handling on MediaStreamTrack (removed state handling
      from MediaStream).
    12. 
      
  12. Reintroduced onended on MediaStream to simplify checking if all
      tracks are ended.
    13. 
      
  13. Aligned the MediaStreamTrack ended event dispatching behavior with
      that of MediaStream.
    14. 
      
  14. Updated the LocalMediaStream.stop() algorithm to implicitly use the
      end track algorithm.
    15. 
      
  15. Replaced an occurrence the term finished track with ended track (to
      align with rest of spec).
    16. 
      
  16. Moved (and extended) the explanation about track references and media
      sources from LocalMediaStream to MediaStreamTrack.
    17. 
    

  

  

    
A. Acknowledgements

    
The editors wish to thank the Working Group chairs and Team Contact,
    Harald Alvestrand, Stefan Håkansson, Erik Lagerway and Dominique
    Hazaël-Massieux, for their support. Substantial text in this
    specification was provided by many people including Martin Thomson, Harald
    Alvestrand, Justin Uberti, Eric Rescorla, Peter Thatcher, Jan-Ivar Bruaroey
    and Peter Saint-Andre. Dan Burnett would like to acknowledge the
    significant support received from Voxeo and Aspect during the development
    of this specification.

    
The RTCRtpSender and RTCRtpReceiver objects were initially described in
    the W3C ORTC CG, and have
    been adapted for use in this specification.

  




B. References
B.1 Normative references
[FIPS-180-4]
U.S. Department of Commerce/National Institute of Standards and Technology. FIPS PUB 180-4 Secure Hash Standard. URL: http://nvlpubs.nist.gov/nistpubs/FIPS/NIST.FIPS.180-4.pdf

[GETUSERMEDIA]
Daniel Burnett; Adam Bergkvist; Cullen Jennings; Anant Narayanan. W3C. Media Capture and Streams. 14 April 2015. W3C Last Call Working Draft. URL: http://www.w3.org/TR/mediacapture-streams/

[HIGHRES-TIME]
Ilya Grigorik; James Simonsen; Jatinder Mann. W3C. High Resolution Time Level 2. 25 February 2016. W3C Working Draft. URL: http://www.w3.org/TR/hr-time-2/

[HTML]
Ian Hickson. WHATWG. HTML Standard. Living Standard. URL: https://html.spec.whatwg.org/multipage/

[HTML5]
Ian Hickson; Robin Berjon; Steve Faulkner; Travis Leithead; Erika Doyle Navara; Edward O'Connor; Silvia Pfeiffer. W3C. HTML5. 28 October 2014. W3C Recommendation. URL: http://www.w3.org/TR/html5/

[ICE]
J. Rosenberg. IETF. Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols. April 2010. Proposed Standard. URL: https://tools.ietf.org/html/rfc5245

[JSEP]
Justin Uberti; Cullen Jennings; Eric Rescorla. IETF. Javascript Session Establishment Protocol. 9 March 2016. Active Internet-Draft. URL: http://datatracker.ietf.org/doc/draft-ietf-rtcweb-jsep/

[RFC2119]
S. Bradner. IETF. Key words for use in RFCs to Indicate Requirement Levels. March 1997. Best Current Practice. URL: https://tools.ietf.org/html/rfc2119

[RFC4566]
M. Handley; V. Jacobson; C. Perkins. IETF. SDP: Session Description Protocol. July 2006. Proposed Standard. URL: https://tools.ietf.org/html/rfc4566

[RFC5245]
J. Rosenberg. IETF. Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols. April 2010. Proposed Standard. URL: https://tools.ietf.org/html/rfc5245

[RFC5389]
J. Rosenberg; R. Mahy; P. Matthews; D. Wing. IETF. Session Traversal Utilities for NAT (STUN). October 2008. Proposed Standard. URL: https://tools.ietf.org/html/rfc5389

[RFC5888]
G. Camarillo; H. Schulzrinne. IETF. The Session Description Protocol (SDP) Grouping Framework. June 2010. Proposed Standard. URL: https://tools.ietf.org/html/rfc5888

[RFC6236]
I. Johansson; K. Jung. IETF. Negotiation of Generic Image Attributes in the Session Description Protocol (SDP). May 2011. Proposed Standard. URL: https://tools.ietf.org/html/rfc6236

[RFC6464]
J. Lennox, Ed.; E. Ivov; E. Marocco. IETF. A Real-time Transport Protocol (RTP) Header Extension for Client-to-Mixer Audio Level Indication. December 2011. Proposed Standard. URL: https://tools.ietf.org/html/rfc6464

[RFC6465]
E. Ivov, Ed.; E. Marocco, Ed.; J. Lennox. IETF. A Real-time Transport Protocol (RTP) Header Extension for Mixer-to-Client Audio Level Indication. December 2011. Proposed Standard. URL: https://tools.ietf.org/html/rfc6465

[RFC6544]
J. Rosenberg; A. Keranen; B. B. Lowekamp; A. B. Roach. IETF. TCP Candidates with Interactive Connectivity Establishment (ICE). March 2012. Proposed Standard. URL: https://tools.ietf.org/html/rfc6544

[RFC7064]
S. Nandakumar; G. Salgueiro; P. Jones; M. Petit-Huguenin. IETF. URI Scheme for the Session Traversal Utilities for NAT (STUN) Protocol. November 2013. Proposed Standard. URL: https://tools.ietf.org/html/rfc7064

[RFC7065]
M. Petit-Huguenin; S. Nandakumar; G. Salgueiro; P. Jones. IETF. Traversal Using Relays around NAT (TURN) Uniform Resource Identifiers. November 2013. Proposed Standard. URL: https://tools.ietf.org/html/rfc7065

[RFC7675]
M. Perumal; D. Wing; R. Ravindranath; T. Reddy; M. Thomson. IETF. Session Traversal Utilities for NAT (STUN) Usage for Consent Freshness. October 2015. Proposed Standard. URL: https://tools.ietf.org/html/rfc7675

[RTCWEB-ALPN]
M. Thomson. IETF. Application Layer Protocol Negotiation for Web Real-Time Communications. 23 July 2014. Active Internet-Draft. URL: http://datatracker.ietf.org/doc/draft-ietf-rtcweb-alpn/

[RTCWEB-AUDIO]
JM. Valin; C. Bran. IETF. WebRTC Audio Codec and Processing Requirements. 27 January 2014. Active Internet-Draft. URL: http://datatracker.ietf.org/doc/draft-ietf-rtcweb-audio/

[RTCWEB-DATA]
R. Jesup; S. Loreto; M. Tuexen. IETF. RTCWeb Data Channels. 21 October 2013. Active Internet-Draft. URL: http://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-channel/

[RTCWEB-DATA-PROTOCOL]
R. Jesup; S. Loreto; M. Tuexen. IETF. RTCWeb Data Channel Protocol. 21 October 2013. Active Internet-Draft. URL: http://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-protocol/

[RTCWEB-RTP]
C. Perkins; M. Westerlund; J. Ott. IETF. Web Real-Time Communication (WebRTC): Media Transport and Use of RTP. 16 December 2013. Active Internet-Draft. URL: http://datatracker.ietf.org/doc/draft-ietf-rtcweb-rtp-usage/

[RTCWEB-SECURITY-ARCH]
Eric Rescorla. IETF. WebRTC Security Architecture. 22 January 2014. Active Internet-Draft. URL: http://datatracker.ietf.org/doc/draft-ietf-rtcweb-security-arch/

[RTCWEB-TRANSPORT]
H. Alvestrand. IETF. Transports for RTCWEB. 22 January 2014. Active Internet-Draft. URL: http://datatracker.ietf.org/doc/draft-ietf-rtcweb-transports/

[SDP]
J. Rosenberg; H. Schulzrinne. IETF. An Offer/Answer Model with Session Description Protocol (SDP). June 2002. Proposed Standard. URL: https://tools.ietf.org/html/rfc3264

[TRAM-TURN-THIRD-PARTY-AUTHZ]
T. Reddy; P. Patil; R. Ravindranath; J. Uberti. IETF. Session Traversal Utilities for NAT (STUN) Extension for Third Party Authorization. 25 February 2015. Internet Draft (work in progress). URL: https://datatracker.ietf.org/doc/draft-ietf-tram-turn-third-party-authz/

[TSVWG-RTCWEB-QOS]
S. Dhesikan; C. Jennings; D. Druta; P. Jones; J. Polk. IETF. DSCP and other packet markings for RTCWeb QoS. 12 November 2014. Internet Draft (work in progress). URL: https://datatracker.ietf.org/doc/draft-ietf-tsvwg-rtcweb-qos

[WEBIDL-1]
Cameron McCormack; Boris Zbarsky. W3C. WebIDL Level 1. 8 March 2016. W3C Candidate Recommendation. URL: http://www.w3.org/TR/WebIDL-1/

[WEBWORKERS]
Ian Hickson. W3C. Web Workers. 24 September 2015. W3C Working Draft. URL: http://www.w3.org/TR/workers/

[WebCryptoAPI]
Ryan Sleevi; Mark Watson. W3C. Web Cryptography API. 11 December 2014. W3C Candidate Recommendation. URL: http://www.w3.org/TR/WebCryptoAPI/

[X509V3]
ITU-T Recommendation X.509 version 3 (1997). "Information Technology - Open Systems Interconnection - The Directory Authentication Framework"  ISO/IEC 9594-8:1997.

[X690]
Recommendation X.690 — Information Technology — ASN.1 Encoding Rules — Specification of Basic Encoding Rules (BER), Canonical Encoding Rules (CER), and Distinguished Encoding Rules (DER). International Telecommunication Union.

[webmessaging]
Ian Hickson. W3C. HTML5 Web Messaging. 19 May 2015. W3C Recommendation. URL: http://www.w3.org/TR/webmessaging/

B.2 Informative references
[IANA-RTP-2]
IANA. RTP Payload Format media types. URL: http://www.iana.org/assignments/rtp-parameters/rtp-parameters.xhtml#rtp-parameters-2

[INDEXEDDB]
Nikunj Mehta; Jonas Sicking; Eliot Graff; Andrei Popescu; Jeremy Orlow; Joshua Bell. W3C. Indexed Database API. 8 January 2015. W3C Recommendation. URL: http://www.w3.org/TR/IndexedDB/

[RFC3550]
H. Schulzrinne; S. Casner; R. Frederick; V. Jacobson. IETF. RTP: A Transport Protocol for Real-Time Applications. July 2003. Internet Standard. URL: https://tools.ietf.org/html/rfc3550

[RFC3890]
M. Westerlund. IETF. A Transport Independent Bandwidth Modifier for the Session Description Protocol (SDP). September 2004. Proposed Standard. URL: https://tools.ietf.org/html/rfc3890

[RFC5285]
D. Singer; H. Desineni. IETF. A General Mechanism for RTP Header Extensions. July 2008. Proposed Standard. URL: https://tools.ietf.org/html/rfc5285

[RFC5322]
P. Resnick, Ed.. IETF. Internet Message Format. October 2008. Draft Standard. URL: https://tools.ietf.org/html/rfc5322

[RFC5506]
I. Johansson; M. Westerlund. IETF. Support for Reduced-Size Real-Time Transport Control Protocol (RTCP): Opportunities and Consequences. April 2009. Proposed Standard. URL: https://tools.ietf.org/html/rfc5506

[RTCWEB-IP-HANDLING]
Guo-wei Shieh; Justin Uberti. IETF. WebRTC IP Address Handling Recommendations. 19 October 2015. Active Internet-Draft. URL: https://datatracker.ietf.org/doc/draft-shieh-rtcweb-ip-handling/

[RTCWEB-OVERVIEW]
H. Alvestrand. IETF. Overview: Real Time Protocols for Brower-based Applications. 14 February 2014. Active Internet-Draft. URL: http://datatracker.ietf.org/doc/draft-ietf-rtcweb-overview/

[RTCWEB-SECURITY]
Eric Rescorla. IETF. Security Considerations for WebRTC. 22 January 2014. Active Internet-Draft. URL: http://datatracker.ietf.org/doc/draft-ietf-rtcweb-security/

[STUN-PARAMETERS]
IETF. IANA. STUN Error Codes. April 2011. IANA Parameter Assignment. URL: http://www.iana.org/assignments/stun-parameters/stun-parameters.xhtml#stun-parameters-6

[TRICKLE-ICE]
E. Ivov; E. Rescorla; J. Uberti. IETF. Trickle ICE: Incremental Provisioning of Candidates for the Interactive Connectivity Establishment (ICE) Protocol. 7 February 2014. Internet Draft (work in progress). URL: http://datatracker.ietf.org/doc/draft-ietf-mmusic-trickle-ice

[WEBSOCKETS-API]
Ian Hickson. W3C. The WebSocket API. 20 September 2012. W3C Candidate Recommendation. URL: http://www.w3.org/TR/websockets/

[XMLHttpRequest]
Anne van Kesteren; Julian Aubourg; Jungkee Song; Hallvord Steen et al. W3C. XMLHttpRequest Level 1. 30 January 2014. W3C Working Draft. URL: http://www.w3.org/TR/XMLHttpRequest/