Class WebRTC

Definition

Namespace:
Tizen.Multimedia.Remoting
Assembly:
Tizen.Multimedia.Remoting.dll

Provides the ability to control WebRTC.

public class WebRTC : IDisposable
Inheritance
object
WebRTC

Constructors

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WebRTC()

Initializes a new instance of the WebRTC class.

Declaration
public WebRTC()

Properties

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BundlePolicy

Gets or sets the bundle policy.
The default bundle policy is MaxBundle.

Declaration
public WebRTCBundlePolicy BundlePolicy { get; set; }
Property Value
Type Description
WebRTCBundlePolicy

The policy of bundle
View Source

IceConnectionState

Gets the ICE connection state of the WebRTC.

Declaration
public WebRTCIceConnectionState IceConnectionState { get; }
Property Value
Type Description
WebRTCIceConnectionState

The current ICE connection state of the WebRTC.
View Source

IceGatheringState

Gets the Ice gathering state of the WebRTC.

Declaration
public WebRTCIceGatheringState IceGatheringState { get; }
Property Value
Type Description
WebRTCIceGatheringState

The current Ice gathering state of the WebRTC.
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IceTransportPolicy

Gets or sets the ICE transport policy.

Declaration
public IceTransportPolicy IceTransportPolicy { get; set; }
Property Value
Type Description
IceTransportPolicy

The policy of ICE transport
View Source

PeerConnectionState

Gets the peer connection state of the WebRTC.

Declaration
public WebRTCPeerConnectionState PeerConnectionState { get; }
Property Value
Type Description
WebRTCPeerConnectionState

The current peer connection state of the WebRTC.
View Source

SignalingState

Gets the signaling state of the WebRTC.

Declaration
public WebRTCSignalingState SignalingState { get; }
Property Value
Type Description
WebRTCSignalingState

The current signaling state of the WebRTC.
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State

Gets the state of the WebRTC.

Declaration
public WebRTCState State { get; }
Property Value
Type Description
WebRTCState

The current state of the WebRTC.
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StunServer

Gets or sets the STUN server url.

Declaration
public string StunServer { get; set; }
Property Value
Type Description
string

The STUN server url

Methods

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AddIceCandidate(string)

Adds a new ICE candidate from the remote peer over its signaling channel.

Declaration
public void AddIceCandidate(string iceCandidate)
Parameters
Type Name Description
string iceCandidate

The ICE candidate.
Remarks

The WebRTC must be in the Negotiating.

View Source

AddIceCandidates(IEnumerable<string>)

Adds new ICE candidates from the remote peer over its signaling channel.

Declaration
public void AddIceCandidates(IEnumerable<string> iceCandidates)
Parameters
Type Name Description
IEnumerable<><string> iceCandidates

The ICE candidates.
Remarks

The WebRTC must be in the Negotiating.

View Source

AddSource(MediaSource)

Adds media source to the current WebRTC.

Declaration
public void AddSource(MediaSource source)
Parameters
Type Name Description
MediaSource source

The media sources to add.
Remarks

This method does not throw state exception anymore(Since API Level 12). It can be called in any state.
Each MediaSource requires different feature or privilege.
MediaCameraSource needs camera feature and privilege.
MediaMicrophoneSource needs microphone feature and recorder privilege.

See Also
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AddSources(params MediaSource[])

Adds media sources from the current WebRTC.

Declaration
public void AddSources(params MediaSource[] sources)
Parameters
Type Name Description
MediaSource[] sources

The media sources to add.
Remarks

This method does not throw state exception anymore(Since API Level 12). It can be called in any state.
Each MediaSource requires different feature or privilege.
MediaCameraSource needs camera feature and privilege.
MediaMicrophoneSource needs microphone feature and recorder privilege.

See Also
View Source

CreateAnswerAsync()

Creates SDP(Session Description Protocol) answer asynchronously with option to an offer received from a remote peer.

Declaration
public Task<string> CreateAnswerAsync()
Returns
Type Description
Task<><string>

The SDP answer.
Remarks

The WebRTC must be in the Negotiating or Playing(Since API Level 12)

See Also
View Source

CreateOfferAsync()

Creates SDP(Session Description Protocol) offer asynchronously to start a new WebRTC connection to a remote peer.

Declaration
public Task<string> CreateOfferAsync()
Returns
Type Description
Task<><string>

The SDP offer.
Remarks

The WebRTC must be in the Negotiating or Playing(Since API Level 12)

See Also
View Source

Dispose()

Releases all resources used by the current instance.

Declaration
public void Dispose()
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Dispose(bool)

Releases the unmanaged resources used by the WebRTC.

Declaration
protected virtual void Dispose(bool disposing)
Parameters
Type Name Description
bool disposing

true to release both managed and unmanaged resources; false to release only unmanaged resources.
View Source

GetLocalDescription()

Gets the session description for a local peer.

Declaration
public string GetLocalDescription()
Returns
Type Description
string

The local session description string
View Source

GetRemoteDescription()

Gets the offer or answer session description from the current remote peer.

Declaration
public string GetRemoteDescription()
Returns
Type Description
string

The remote session description string
See Also
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GetStatistics(WebRTCStatisticsCategory)

Retrieves the current statistics information.

Declaration
public ReadOnlyCollection<WebRTC.WebRTCStatistics> GetStatistics(WebRTCStatisticsCategory category)
Parameters
Type Name Description
WebRTCStatisticsCategory category

The category of statistics to get.
Returns
Type Description
ReadOnlyCollection<><WebRTCStatistics>

The WebRTC statistics informations.
Remarks

The WebRTC must be in the Negotiating(Since API Level 12) or Playing

View Source

GetTurnServer()

Retrieves all turn servers.

Declaration
public ReadOnlyCollection<string> GetTurnServer()
Returns
Type Description
ReadOnlyCollection<><string>

The turn server list.
View Source

RemoveSource(MediaSource)

Removes media source from the current WebRTC.

Declaration
public void RemoveSource(MediaSource source)
Parameters
Type Name Description
MediaSource source

The media source to remove.
Remarks

This method does not throw state exception anymore(Since API Level 12). It can be called in any state.
If user want to use removed MediaSource again, user should create new instance for it.

See Also
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RemoveSources(params MediaSource[])

Removes media sources from the current WebRTC.

Declaration
public void RemoveSources(params MediaSource[] sources)
Parameters
Type Name Description
MediaSource[] sources

The media source to remove.
Remarks

This method does not throw state exception anymore(Since API Level 12). It can be called in any state.
If user want to use removed MediaSource again, user should create new instance for it.

See Also
View Source

SetLocalDescription(string)

Sets the session description for a local peer.

Declaration
public void SetLocalDescription(string description)
Parameters
Type Name Description
string description

The local session description.
Remarks

The WebRTC must be in the Negotiating or Playing(Since API Level 12)

See Also
View Source

SetRemoteDescription(string)

Sets the offer or answer session description from the current remote peer.

Declaration
public void SetRemoteDescription(string description)
Parameters
Type Name Description
string description

The remote session description.
Remarks

The WebRTC must be in the Negotiating or Playing(Since API Level 12)

See Also
View Source

SetTurnServer(string)

Sets a turn server for signalling with remote peer which cannot be connected directly.

Declaration
public void SetTurnServer(string turnServer)
Parameters
Type Name Description
string turnServer
View Source

SetTurnServers(params string[])

Sets turn servers for signalling with remote peer which cannot be connected directly.

Declaration
public void SetTurnServers(params string[] turnServers)
Parameters
Type Name Description
string[] turnServers
View Source

Start()

Starts the WebRTC with specific media source.

Declaration
public void Start()
Remarks

The WebRTC must be in the Idle state.
The WebRTC state will be Negotiating state.
The user should check whether State is changed to Negotiating state or not.

View Source

StartAsync()

Starts the WebRTC asynchronously with specific media source.

Declaration
public Task StartAsync()
Returns
Type Description
Task
Remarks

The WebRTC must be in the Idle state.
The WebRTC state will be Negotiating state.
This ensures that State is changed to Negotiating state.

View Source

Stop()

Stops the WebRTC.

Declaration
public void Stop()
Remarks

The WebRTC must be in the Negotiating or Playing state.
The WebRTC state will be Idle state.
The user should check whether State is changed to Idle state or not.

Events

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AudioFrameEncoded

Occurs when each audio frame is ready to render.

Declaration
public event EventHandler<WebRTCFrameEncodedEventArgs> AudioFrameEncoded
Event Type
Type Description
EventHandler<><WebRTCFrameEncodedEventArgs>
View Source

DataChannel

Occurs when the data channel is created to the connection by the remote peer.

Declaration
public event EventHandler<WebRTCDataChannelEventArgs> DataChannel
Event Type
Type Description
EventHandler<><WebRTCDataChannelEventArgs>
View Source

ErrorOccurred

Occurs when any error occurs.

Declaration
public event EventHandler<WebRTCErrorOccurredEventArgs> ErrorOccurred
Event Type
Type Description
EventHandler<><WebRTCErrorOccurredEventArgs>
View Source

IceCandidate

Occurs when the WebRTC needs to send the ICE candidate message to the remote peer.

Declaration
public event EventHandler<WebRTCIceCandidateEventArgs> IceCandidate
Event Type
Type Description
EventHandler<><WebRTCIceCandidateEventArgs>
View Source

IceConnectionStateChanged

Occurs when the WebRTC ICE connection state is changed.

Declaration
public event EventHandler<WebRTCIceConnectionStateChangedEventArgs> IceConnectionStateChanged
Event Type
Type Description
EventHandler<><WebRTCIceConnectionStateChangedEventArgs>
View Source

IceGatheringStateChanged

Occurs when the WebRTC ICE gathering state is changed.

Declaration
public event EventHandler<WebRTCIceGatheringStateChangedEventArgs> IceGatheringStateChanged
Event Type
Type Description
EventHandler<><WebRTCIceGatheringStateChangedEventArgs>
View Source

NegotiationNeeded

Occurs when negotiation is needed.

Declaration
public event EventHandler<EventArgs> NegotiationNeeded
Event Type
Type Description
EventHandler<><EventArgs>
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PeerConnectionStateChanged

Occurs when the WebRTC peer connection state is changed.

Declaration
public event EventHandler<WebRTCPeerConnectionStateChangedEventArgs> PeerConnectionStateChanged
Event Type
Type Description
EventHandler<><WebRTCPeerConnectionStateChangedEventArgs>
View Source

SignalingStateChanged

Occurs when the WebRTC signaling state is changed.

Declaration
public event EventHandler<WebRTCSignalingStateChangedEventArgs> SignalingStateChanged
Event Type
Type Description
EventHandler<><WebRTCSignalingStateChangedEventArgs>
View Source

StateChanged

Occurs when WebRTC state is changed.

Declaration
public event EventHandler<WebRTCStateChangedEventArgs> StateChanged
Event Type
Type Description
EventHandler<><WebRTCStateChangedEventArgs>
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TrackAdded

Occurs when a new track has been added to the WebRTC.

Declaration
public event EventHandler<WebRTCTrackAddedEventArgs> TrackAdded
Event Type
Type Description
EventHandler<><WebRTCTrackAddedEventArgs>
View Source

VideoFrameEncoded

Occurs when each video frame is ready to render.

Declaration
public event EventHandler<WebRTCFrameEncodedEventArgs> VideoFrameEncoded
Event Type
Type Description
EventHandler<><WebRTCFrameEncodedEventArgs>

Extension Methods