Module webrtc::media::rtp [−][src]
Modules
Structs
rtcpfeedback signals the connection to use additional RTCP packet types. https://draft.ortc.org/#dom-rtcrtcpfeedback
RTPCapabilities represents the capabilities of a transceiver https://w3c.github.io/webrtc-pc/#rtcrtpcapabilities
RTPCodingParameters provides information relating to both encoding and decoding. This is a subset of the RFC since Pion WebRTC doesn’t implement encoding/decoding itself http://draft.ortc.org/#dom-rtcrtpcodingparameters
RTPReceiveParameters contains the RTP stack settings used by receivers
RTPSendParameters contains the RTP stack settings used by receivers
RTPTransceiverInit dictionary is used when calling the WebRTC function addTransceiver() to provide configuration options for the new transceiver.
Constants
TYPE_RTCP_FB_ACK ..
TYPE_RTCP_FB_CCM ..
TYPE_RTCP_FB_GOOG_REMB ..
TYPE_RTCP_FB_NACK ..
TYPE_RTCP_FBT_RANSPORT_CC ..
Type Definitions
PayloadType identifies the format of the RTP payload and determines its interpretation by the application. Each codec in a RTP Session will have a different PayloadType https://tools.ietf.org/html/rfc3550#section-3
RTPDecodingParameters provides information relating to both encoding and decoding. This is a subset of the RFC since Pion WebRTC doesn’t implement decoding itself http://draft.ortc.org/#dom-rtcrtpdecodingparameters
RTPEncodingParameters provides information relating to both encoding and decoding. This is a subset of the RFC since Pion WebRTC doesn’t implement encoding itself http://draft.ortc.org/#dom-rtcrtpencodingparameters
SSRC represents a synchronization source A synchronization source is a randomly chosen value meant to be globally unique within a particular RTP session. Used to identify a single stream of media. https://tools.ietf.org/html/rfc3550#section-3