4.2.1 RTCConfiguration Type

dictionary RTCConfiguration {
    sequence<RTCIceServer> iceServers;
    RTCIceTransportPolicy  iceTransportPolicy = "all";
    RTCBundlePolicy        bundlePolicy = "balanced";
    DOMString              peerIdentity;
};

4.2.2 RTCIceServer Type

dictionary RTCIceServer {
    (DOMString or sequence<DOMString>) urls;
    DOMString                          username;
    DOMString                          credential;
};

In network topologies with multiple layers of NATs, it is desirable to have a STUN server between every layer of NATs in addition to the TURN servers to minimize the peer to peer network latency.

An example array of RTCIceServer objects is:

[ { "urls": "stun:stun1.example.net" }, { "urls": "turn:turn.example.org", "username": "user", "credential": "myPassword" } ]

4.2.3 RTCIceTransportPolicy Enum

enum RTCIceTransportPolicy {
    "none",
    "relay",
    "all"
};

4.2.4 RTCBundlePolicy Enum

enum RTCBundlePolicy {
    "balanced",
    "max-compat",
    "max-bundle"
};

4.2.5 Offer/Answer Options

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

dictionary RTCOfferOptions {
    long    offerToReceiveVideo;
    long    offerToReceiveAudio;
    boolean voiceActivityDetection = true;
    boolean iceRestart = false;
};

enum RTCIdentityOption {
    "yes",
    "no",
    "ifconfigured"
};

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 an associated ICE agent [ICE], RTCPeerConnection signaling state, ICE gathering state, and ICE connection state. These are initialized when the object is created.

An RTCPeerConnection object has two associated stream sets. A local streams set, representing streams that are currently sent, and a remote streams set, representing streams that are currently received with this RTCPeerConnection object. The stream sets are initialized to empty sets when the RTCPeerConnection object is created.

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

1. Validate the RTCConfiguration argument by running the steps defined by the updateIce() method.

2. Let connection be a newly created RTCPeerConnection object.

3. Create an ICE Agent as defined in [ICE] and let connection's RTCPeerConnection ICE Agent be that ICE Agent and provide it the the ICE servers list. The ICE Agent will proceed with gathering as soon as the ICE transports setting is not set to none. At this point the ICE Agent does not know how many ICE components it needs (and hence the number of candidates to gather), but it can make a reasonable assumption such as 2. As the RTCPeerConnection object gets more information, the ICE Agent can adjust the number of components.

4. Set connection's RTCPeerConnection signalingState to stable.

5. Set connection's RTCPeerConnection ice connection state to new.

6. Set connection's RTCPeerConnection ice gathering state to new.

7. Initialize an internal variable to represent a queue of operations with an empty set.

8. Return connection.

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

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

2. If the length of the operations array is exactly 1, execute the function from the front of the queue asynchronously.

3. When the asynchronous operation completes (either successfully or with an error), remove the corresponding object from the operations array. After removal, if the array is non-empty, execute the first object queued asynchronously and repeat this step on completion.

The general idea is to have only one among createOffer, setLocalDescription, createAnswer and setRemoteDescription executing at any given time. If subsequent calls are made while one of them is still executing, they are added to a queue and processed when the previous operation is fully completed. It is valid, and expected, for normal error handling procedures to be applied.

Additionally, during the lifetime of the RTCPeerConnection object, the following procedures are followed when an ICE event occurs:

1. If the RTCPeerConnection ice gathering state is new and the ICE transports setting is not set to none, the user agent MUST queue a task to start gathering ICE addresses and set the ice gathering state to gathering.

2. If the ICE Agent has found one or more candidate pairs for each MediaStreamTrack that forms a valid connection, the ICE connection state is changed to "connected".

3. When the ICE Agent finishes checking all candidate pairs, if at least one connection has been found for each MediaStreamTrack, the RTCPeerConnection ice connection state is changed to "completed"; otherwise "failed".

When the ICE Agent needs to notify the script about the candidate gathering progress, 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 RTCPeerConnection signalingState is closed, abort these steps.

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

   • A new candidate is available.

     Add the candidate to connection's localDescription and create a RTCIceCandidate object to represent the candidate. Let newCandidate be that object.

   • The gathering process is done.

     Set connection's ice gathering state to completed and let newCandidate be null.

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

User agents negotiate the codec resolution, bitrate, and other media parameters. It is RECOMMENDED that user agents initially negotiate for the maximum resolution of a video stream. For streams that are then rendered (using a video element), it is RECOMMENDED that user agents renegotiate for a resolution that matches the rendered display size.

The word "components" in this context refers to an RTP media flow and does not have anything to do with how [ICE] uses the term "component".

When a user agent has reached the point where a MediaStream can be created to represent incoming components, the user agent MUST run the following steps:

1. Let connection be the RTCPeerConnection expecting this media.

2. Create a MediaStream object stream, to represent the incoming media stream.

3. Run the algorithm to represent an incoming component with a track for each incoming component.

   Note

   The creation of new incoming MediaStreams may be triggered either by SDP negotiation or by the receipt of media on a given flow.

4. Queue a task to run the following substeps:

   1. If the connection's RTCPeerConnection signalingState is closed, abort these steps.

   2. Add stream to connection's remote streams set.

   3. Fire a stream event named addstream with stream at the connection object.

When a user agent has negotiated media for a component that belongs to a media stream that is already represented by an existing MediaStream object, the user agent MUST associate the component with that MediaStream object.

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

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

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

3. By definition, stream is now ended.

   Note

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

4. Queue a task to run the following substeps:

   1. If the connection's RTCPeerConnection signalingState is closed, abort these steps.

   2. Remove stream from connection's remote streams set.

   3. Fire a stream event named removestream with stream at the connection object.

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

If something in the browser changes that causes the RTCPeerConnection object to need to initiate a new session description negotiation, a negotiationneeded event is fired at the RTCPeerConnection object.

In particular, if an RTCPeerConnection object is consuming a MediaStream on which a track is added, by, e.g., the addTrack() method being invoked, the RTCPeerConnection object MUST fire the "negotiationneeded" event. Removal of media components must also trigger "negotiationneeded".

4.3.2 Interface Definition

[ Constructor (RTCConfiguration configuration)]
interface RTCPeerConnection : EventTarget  {
    Promise<RTCSessionDescription> createOffer (optional RTCOfferOptions options);
    Promise<RTCSessionDescription> createAnswer ();
    Promise<void>                  setLocalDescription (RTCSessionDescription description);
    readonly    attribute RTCSessionDescription? localDescription;
    Promise<void>                  setRemoteDescription (RTCSessionDescription description);
    readonly    attribute RTCSessionDescription? remoteDescription;
    readonly    attribute RTCSignalingState      signalingState;
    void                           updateIce (RTCConfiguration configuration);
    Promise<void>                  addIceCandidate (RTCIceCandidate candidate);
    readonly    attribute RTCIceGatheringState   iceGatheringState;
    readonly    attribute RTCIceConnectionState  iceConnectionState;
    readonly    attribute boolean?               canTrickleIceCandidates;
    RTCConfiguration               getConfiguration ();
    sequence<MediaStream>          getLocalStreams ();
    sequence<MediaStream>          getRemoteStreams ();
    MediaStream?                   getStreamById (DOMString streamId);
    void                           addStream (MediaStream stream);
    void                           removeStream (MediaStream stream);
    void                           close ();
                attribute EventHandler           onnegotiationneeded;
                attribute EventHandler           onicecandidate;
                attribute EventHandler           onsignalingstatechange;
                attribute EventHandler           onaddstream;
                attribute EventHandler           onremovestream;
                attribute EventHandler           oniceconnectionstatechange;
                attribute EventHandler           onicegatheringstatechange;
};

4.3.3 Legacy Interface Extensions

partial interface RTCPeerConnection {
    void createOffer (RTCSessionDescriptionCallback successCallback, RTCPeerConnectionErrorCallback failureCallback, optional RTCOfferOptions options);
    void setLocalDescription (RTCSessionDescription description, VoidFunction successCallback, RTCPeerConnectionErrorCallback failureCallback);
    void createAnswer (RTCSessionDescriptionCallback successCallback, RTCPeerConnectionErrorCallback failureCallback);
    void setRemoteDescription (RTCSessionDescription description, VoidFunction successCallback, RTCPeerConnectionErrorCallback failureCallback);
    void addIceCandidate (RTCIceCandidate candidate, VoidFunction successCallback, RTCPeerConnectionErrorCallback failureCallback);
};

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 RTCPeerConnection signalingState is closed, no such event handler can be triggered and it is therefore safe to garbage collect the object.

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

4.4.1 RTCPeerState Enum

enum RTCSignalingState {
    "stable",
    "have-local-offer",
    "have-remote-offer",
    "have-local-pranswer",
    "have-remote-pranswer",
    "closed"
};

The non-normative peer state transitions are:

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
• close(): closed

Callee transition:

• new RTCPeerConnection(): stable
• setRemote(offer): have-remote-offer
• setLocal(pranswer): have-local-pranswer
• setLocal(answer): stable
• close(): closed

4.4.2 RTCIceGatheringState Enum

enum RTCIceGatheringState {
    "new",
    "gathering",
    "complete"
};

4.4.3 RTCIceConnectionState Enum

enum RTCIceConnectionState {
    "new",
    "checking",
    "connected",
    "completed",
    "failed",
    "disconnected",
    "closed"
};

States take either the value of any component or all components, as outlined below:

• checking occurs if ANY component has received a candidate and can start checking
• connected occurs if ALL components have established a working connection
• completed occurs if ALL components have finalized the running of their ICE processes
• failed occurs if ANY component has given up trying to connect
• disconnected occurs if ANY component has failed liveness checks
• closed occurs only if RTCPeerConnection.close() has been called.

If a component is discarded as a result of signaling (e.g. RTCP mux or BUNDLE), the state may advance directly from checking to completed.

Some example transitions might be:

• new RTCPeerConnection(): new
• (new, remote candidates received): checking
• (checking, found usable connection): connected
• (checking, gave up): failed
• (connected, finished all checks): completed
• (completed, lost connectivity): disconnected
• (any state, ICE restart occurs): new
• close(): closed

The non-normative ICE state transitions are:

4.5.1 RTCPeerConnectionErrorCallback

callback RTCPeerConnectionErrorCallback = void (DOMError error);

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.

4.6.2 RTCSdpError

interface RTCSdpError : DOMError {
    readonly    attribute long sdpLineNumber;
};

4.7.1 RTCSdpType

The RTCSdpType enum describes the type of an RTCSessionDescription instance.

enum RTCSdpType {
    "offer",
    "pranswer",
    "answer"
};

4.7.2 RTCSessionDescription Class

dictionary RTCSessionDescriptionInit {
    RTCSdpType type;
    DOMString  sdp;
};

[ Constructor (optional RTCSessionDescriptionInit descriptionInitDict)]
interface RTCSessionDescription {
                attribute RTCSdpType? type;
                attribute DOMString?  sdp;
    serializer = {attribute};
};

4.7.3 RTCSessionDescriptionCallback

callback RTCSessionDescriptionCallback = void (RTCSessionDescription sdp);

4.8.1 RTCIceCandidate Type

This class is a future extensible carrier for the data contained in it and does not perform any substantive processing.

dictionary RTCIceCandidateInit {
    DOMString      candidate;
    DOMString      sdpMid;
    unsigned short sdpMLineIndex;
};

[ Constructor (optional RTCIceCandidateInit candidateInitDict)]
interface RTCIceCandidate {
                attribute DOMString?      candidate;
                attribute DOMString?      sdpMid;
                attribute unsigned short? sdpMLineIndex;
    serializer = {attribute};
};

4.8.2 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.

dictionary RTCPeerConnectionIceEventInit : EventInit {
    RTCIceCandidate candidate;
};

[ Constructor (DOMString type, RTCPeerConnectionIceEventInit eventInitDict)]
interface RTCPeerConnectionIceEvent : Event {
    readonly    attribute RTCIceCandidate candidate;
};

5.1.1 Attributes

ondatachannel of type EventHandler,

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

5.1.2 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. If the RTCPeerConnection object's RTCPeerConnection signalingState is closed, throw an InvalidStateError exception and abort these steps.

2. Let channel be a newly created RTCDataChannel object.

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

4. 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).

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

6. 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.

7. 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.

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

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

Parameter	Type	Nullable	Optional	Description
label	DOMString	✘	✘
dataChannelDict	RTCDataChannelInit	✘	✔

Return type: RTCDataChannel

5.2.1 Attributes

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.

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).

id of type unsigned short, readonly

The RTCDataChannel.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.

label of type DOMString, readonly

The RTCDataChannel.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.

maxPacketLifeTime of type unsigned short, readonly , nullable

The RTCDataChannel.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 RTCDataChannel.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.

negotiated of type boolean, readonly

The RTCDataChannel.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.

onclose of type EventHandler,

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

onerror of type EventHandler,

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

onmessage of type EventHandler,

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

onopen of type EventHandler,

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

ordered of type boolean, readonly

The RTCDataChannel.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.

protocol of type DOMString, readonly

The RTCDataChannel.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.

readyState of type RTCDataChannelState, readonly

The RTCDataChannel.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).

5.2.2 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 RTCDataChannel 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	DOMString	✘	✘

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

5.2.3 Dictionary RTCDataChannelInit Members

id of type unsigned short

Overrides the default selection of id for this channel.

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..

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.

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.

protocol of type DOMString, defaulting to ""

Subprotocol name used for this channel.

5.3.1 Constructors

RTCDataChannelEvent

readonly attribute RTCDataChannel channel

Parameter	Type	Nullable	Optional	Description
type	DOMString	✘	✘
eventInitDict	RTCDataChannelEventInit	✘	✘

5.3.2 Attributes

channel of type RTCDataChannel, readonly

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

5.3.3 Dictionary RTCDataChannelEventInit Members

channel of type RTCDataChannel

TODO

6.1.1 Methods

createDTMFSender

The createDTMFSender() method creates an RTCDTMFSender that references the given MediaStreamTrack. The MediaStreamTrack MUST be an element of a MediaStream that's currently in the RTCPeerConnection object's local streams set; if not, throw an InvalidParameter exception and abort these steps.

Parameter	Type	Nullable	Optional	Description
track	MediaStreamTrack	✘	✘

Return type: RTCDTMFSender

6.2.1 Attributes

canInsertDTMF of type boolean, readonly

The canInsertDTMF attribute MUST indicate if the RTCDTMFSender is capable of sending DTMF.

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.

ontonechange of type EventHandler,

This event handler uses the RTCDTMFToneChangeEvent interface to return the character for each tone as it is played out. See RTCDTMFToneChangeEvent for details.

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.

track of type MediaStreamTrack, readonly

The track attribute MUST return the MediaStreamTrack given as argument to the createDTMFSender() method.

6.2.2 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.

The duration parameter indicates the duration in ms to use for each character passed in the tones parameters. The duration cannot be more than 6000 ms or less than 40 ms. The default duration is 100 ms for each tone.

The interToneGap parameter indicates the gap between tones. 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.

Issue 2

ISSUE: How are invalid values handled?

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

1. If the associated MediaStreamTrack is not connected to the associated RTCPeerConnection, return.
2. If the canInsertDTMF attribute is false, return.
3. 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.
4. If toneBuffer contains any unrecognized characters, throw an InvalidCharacterError exception and abort these steps.
5. If toneBuffer is an empty string, return.
6. 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.
7. If the value of the interToneGap attribute is less than 30, set it to 30.
8. 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	long = 100	✘	✔
interToneGap	long = 70	✘	✔

Return type: void

6.3.1 Constructors

RTCDTMFToneChangeEvent

readonly attribute DOMString tone

Parameter	Type	Nullable	Optional	Description
type	DOMString	✘	✘
eventInitDict	RTCDTMFToneChangeEventInit	✘	✘

6.3.2 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.

6.3.3 Dictionary RTCDTMFToneChangeEventInit Members

tone of type DOMString

TODO

7.2.1 Methods

getStats

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

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

1. If the RTCPeerConnection object's RTCPeerConnection signalingState is closed, throw an InvalidStateError exception.

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

3. Let selectorArg be the methods first argument.

4. If selectorArg is an invalid selector, the user agent MUST queue a task to invoke the failure callback (the method's third argument).

5. Start gathering the stats indicated by selectorArg. In case selectorArg is null, stats MUST be gathered for the whole RTCPeerConnection object.

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

Parameter	Type	Nullable	Optional	Description
selector	MediaStreamTrack	✔	✘
successCallback	RTCStatsCallback	✘	✘
failureCallback	RTCPeerConnectionErrorCallback	✘	✘

Return type: void

7.3.1 Callback RTCStatsCallback Parameters

report of type RTCStatsReport

A RTCStatsReport representing the gathered stats.

7.4.1 Methods

RTCStats

Getter to retrieve the RTCStats objects that this stats report is composed of.

The set of supported property names [WEBIDL] is defined as the ids of all the RTCStats objects that has been generated for this stats report. The order of the property names is left to the user agent.

Parameter	Type	Nullable	Optional	Description
id	DOMString	✘	✘

Return type: getter

7.5.1 Dictionary RTCStats Members

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.

Note

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.

timestamp of type DOMHiResTimeStamp

The timestamp, of type DOMHiResTimeStamp [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.

7.6.1 Dictionary RTCRTPStreamStats Members

remoteId of type DOMString

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

ssrc of type DOMString

...

7.6.2 Dictionary RTCInboundRTPStreamStats Members

bytesReceived of type unsigned long

...

packetsReceived of type unsigned long

...

7.6.3 Dictionary RTCOutboundRTPStreamStats Members

bytesSent of type unsigned long

...

packetsSent of type unsigned long

...

8.1.1 Identity Provider Selection

In order to communicate with the IdP, the browser instantiates an isolated interpreted context, effectively an invisible IFRAME. The initial contents of the context are loaded from a URI derived from the IdP's domain name, as described in [RTCWEB-SECURITY-ARCH].

For purposes of generating assertions, the IdP shall be chosen as follows:

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 browser can use an IdP configured into the browser.

In order to verify assertions, the IdP domain name and protocol are taken from the domain and protocol fields of the identity assertion.

8.1.2 Instantiating an IdP Proxy

The browser creates an IdP proxy by loading an isolated, invisible IFRAME with HTML content from the IdP URI. The URI for the IdP is a well-known URI formed from the domain and protocol fields, as specified in [RTCWEB-SECURITY-ARCH].

When an IdP proxy is requiured, the browser performs the following steps:

1. An invisible, sandboxed IFRAME is created within the browser context. The IFRAME sandbox attribute is set to "allow-forms allow-scripts allow-same-origin" to limit the capabilities available to the IdP. The browser MUST prevent the IdP proxy from navigating the browsing context to a different location. The browser MUST prevent the IdP proxy from interacting with the user (this includes, in particular, popup windows and user dialogs).
2. Once the IdP proxy is created, the browser creates a MessageChannel [webmessaging] within the context of the IdP proxy and assigns one port from the channel to a variable named rtcwebIdentityPort on the window. This message channel forms the basis of communication between the browser and the IdP proxy. Since it is an essential security property of the web sandbox that a page is unable to insert objects into content from another origin, this ensures that the IdP proxy can trust that messages originating from window.rtcwebIdentityPort are from RTCPeerConnection and not some other page. This protection ensures that pages from other origins are unable to instantiate IdP proxies and obtain identity assertions.
3. The IdP proxy completes loading and informs the RTCPeerConnection that it is ready by sending a "READY" message to the message channel port [RTCWEB-SECURITY-ARCH]. Once this message is received by the RTCPeerConnection, the IdP is considered ready to receive requests to generate or verify identity assertions.

[TODO: This is not sufficient unless we expect the IdP to protect this information. Otherwise, the a=identity information can be copied from a session with "good" properties to any other session with the same fingerprint information. Since we want to reuse credentials, that would be bad.] The identity mechanism MUST provide an indication to the remote side of whether it requires the stream contents to be protected. Implementations MUST have an user interface that indicates the different cases and identity for these.

8.2.1 User Login Procedure

An IdP could respond to a request to generate an identity assertion with a "LOGINNEEDED" error. This indicates that the site does not have the necessary information available to it (such as cookies) to authorize the creation of an identity assertion.

The "LOGINNEEDED" response includes a URL for a page where the authorization process can be completed. This URL is exposed to the application through the loginUrl attribute of the idpassertionerror event. This URL might be to a page where a user is able to enter their (IdP) username and password, or otherwise provide any information the IdP needs to authorize a assertion request.

An application can load the login URL in an IFRAME or popup; the resulting page then provides the user with an opportunity to provide information necessary to complete the authorization process.

Once the authorization process is complete, the page loaded in the IFRAME or popup sends a message using postMessage [webmessaging] to the page that loaded it (through the window.opener attribute for popups, or through window.parent for pages loaded in an IFRAME). The message MUST be the DOMString "LOGINDONE". This message informs the application that another attempt at generating an identity assertion is likely to be successful.

8.4.1 Attributes

onidentityresult of type EventHandler,

This event handler, of event handler event type identityresult, MUST be fired by all objects implementing the RTCPeerConnection interface. This event is fired when an identity assertion is successfully generated. Note: this event is fired when an identity assertion is generated during the creation of an offer or answer.

onidpassertionerror of type EventHandler,

This event handler of type idpassertionerror MUST be fired when an IdP encounters an error in generating an identity assertion.

onidpvalidationerror of type EventHandler,

This event handler of type idvalidationperror MUST be fired when an IdP encounters an error in validating an identity assertion.

onpeeridentity of type EventHandler,

This simple event handler of type peeridentity MUST be fired when a peer identity from a peer is successfully validated.

peerIdentity of type RTCIdentityAssertion, readonly , nullable

Contains the peer identity assertion information if an identity assertion was provided and verified. Once this value is set to a non-null value, it cannot change.

8.4.2 Methods

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 connection's RTCPeerConnection signalingState is closed, abort these steps.

2. Request an identity assertion from the IdP.

No parameters.

Return type: void

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 will be used to get an assertion.

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

1. If the connection's RTCPeerConnection signalingState is closed, throw an InvalidStateError exception and abort these steps.

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

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 the need to generate SDP with either createOffer or createAnswer.

Parameter	Type	Nullable	Optional	Description
provider	DOMString	✘	✘
protocol	DOMString	✘	✔
username	DOMString	✘	✔

Return type: void

8.5.1 Dictionary RTCIdentityAssertion Members

idp of type DOMString

A domain name representing the identity provider.

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].

8.6.1 Attributes

assertion of type DOMString,

A string containing the encoded identity assertion (the information that would be added to the "a=identity" line in SDP [RTCWEB-SECURITY-ARCH]).

8.7.1 Attributes

idp of type DOMString,

The domain name of the identity provider that is providing the error response.

loginUrl of type DOMString, , nullable

An IdP that is unable to generate an identity assertion due to a lack of sufficient user authentication information can provide a URL to a page where the user can complete authentication. If the IdP provides this URL, this attribute includes the value provided by the IdP.

protocol of type DOMString,

The IdP protocol that is in use.

9.2.1 id

The id attribute specified in MediaStream returns an id that is unique to this stream, so that streams can be recognized after they are sent through 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.

9.2.2 Events on MediaStream

A new media track may be associated with an existing MediaStream. For example, if a remote peer adds a new MediaStreamTrack object to a MediaStream that is being sent over an RTCPeerConnection, this is observed on the local user agent. If this happens for the reason exemplified, or for any other reason than the addTrack() method being invoked locally on a MediaStream or tracks being added as the stream is created (i.e. the stream is initialized with tracks), the user agent MUST run the following steps:

1. Let stream be the target MediaStream object.

2. Represent component with track: Run the following steps to create a track representing the incoming component:

   1. Create a MediaStreamTrack object track to represent the component.

   2. Initialize track's kind attribute to "audio" or "video" depending on the media type of the incoming component.

   3. Initialize track's id attribute to the component track id.

   4. Initialize track's label attribute to "remote audio" or "remote video" depending on the media type of the incoming component.

   5. Initialize track's readyState attribute to live and muted attribute to true.

   6. Add track to stream's track set.

3. Fire a track event named addtrack with the newly created MediaStreamTrack object at stream.

An existing media track may also be disassociated from a MediaStream. If this happens for any other reason than the removeTrack() method being invoked locally on a MediaStream or the stream being destroyed, the user agent MUST run the following steps:

1. Let stream be the target MediaStream object.

2. Let track be the MediaStreamTrack object representing the media component about to be removed.

3. Remove track from stream's track set.

4. Fire a track event named removetrack with track at stream.

The event source for the onended event in the networked case is the RTCPeerConnection object.

9.4.1 Constructors

MediaStreamEvent

readonly attribute MediaStream? stream

Parameter	Type	Nullable	Optional	Description
type	DOMString	✘	✘
eventInitDict	MediaStreamEventInit	✘	✘

9.4.2 Attributes

stream of type MediaStream, readonly , nullable

The stream attribute represents the MediaStream object associated with the event.

9.4.3 Dictionary MediaStreamEventInit Members

stream of type MediaStream

TODO

9.5.1 Dictionary MediaStreamConstraints Members

peerIdentity of type DOMString

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

9.5.2 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;
};

9.5.3 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.

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).

9.5.4 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.