PeerConnection 
    Java-land version of the PeerConnection APIs; wraps the C++ API http://www.webrtc.org/reference/native-apis, which in turn is inspired by the JS APIs: http://dev.w3.org/2011/webrtc/editor/webrtc.html and http://www.w3.org/TR/mediacapture-streams/
Types
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                  enum AdapterType
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                  enum BundlePolicy
Java version of PeerConnectionInterface.
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                  Java version of PeerConnectionInterface.
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                  Java version of PeerConnectionInterface.
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                  enum IceConnectionState
Tracks PeerConnectionInterface::IceConnectionState
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                  enum IceGatheringState
Tracks PeerConnectionInterface::IceGatheringState
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                  enum IceTransportsType
Java version of PeerConnectionInterface.
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                  enum PeerConnectionState
Tracks PeerConnectionInterface::PeerConnectionState
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                  enum PortPrunePolicy
Java version of webrtc::PortPrunePolicy
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                  Java version of PeerConnectionInterface.
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                  enum RtcpMuxPolicy
Java version of PeerConnectionInterface.
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                  enum SdpSemantics
Java version of webrtc::SdpSemantics.
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                  enum SignalingState
Tracks PeerConnectionInterface::SignalingState
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                  enum TcpCandidatePolicy
Java version of PeerConnectionInterface.
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                  enum TlsCertPolicy
Tracks PeerConnectionInterface::TlsCertPolicy
Functions
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                  Adds a new MediaStream to be sent on this peer connection.
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                  Adds a new media stream track to be sent on this peer connection, and returns the newly created RtpSender.
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                  Creates a new RtpTransceiver and adds it to the set of transceivers.
open fun addTransceiver(track: MediaStreamTrack, @Nullable init: RtpTransceiver.RtpTransceiverInit): RtpTransceiver
open fun addTransceiver(mediaType: MediaStreamTrack.MediaType, @Nullable init: RtpTransceiver.RtpTransceiverInit): RtpTransceiver
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                  Creates an RtpSender without a track.
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                  Gets stats using the new stats collection API, see webrtc/api/stats/.
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                  Removes the given media stream from this peer connection.
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                  Stops sending media from sender.
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                  Tells the PeerConnection that ICE should be restarted.
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                  Enables/disables playout of received audio streams.
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                  Enables/disables recording of transmitted audio streams.
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                  Limits the bandwidth allocated for all RTP streams sent by this PeerConnection.
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                  Starts recording an RTC event log.
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                  Stops recording an RTC event log.