RTCRtpEncodingParameters

An instance of the WebRTC API's RTCRtpEncodingParameters dictionary describes a single configuration of a codec. It's used in the RTCRtpParameters describing the configuration of an RTP sender or receiver.

Properties

active

If true, the described encoding is currently actively being used. That is, for RTP senders, the encoding is currently being used to send data, while for receivers, the encoding is being used to decode received data.

codecPayloadType

When describing a codec for an RTCRtpSender, codecPayloadType is a single 8-bit byte (or octet) specifying the codec to use for sending the stream.

dtx

Only used for RTCRtpSender, this property indicates whether or not to use discontinuous transmission (a feature by which a phone is turned off or the microphone muted automatically in the absence of voice activity). Defaults to true.

maxBitrate

An unsigned long integer indicating the maximum number of bits per second to allow for this encoding. Other parameters may further constrain the bit rate, such as the value of maxFramerate or transport or physical network limitations.

maxFramerate

A double-precision floating-point value specifying the maximum number of frames per second to allow for this encoding.

priority

A string from the RTCPriorityType enumerated type which indicates the encoding's priority. Default value is "low".

ptime

An unsigned long integer value which, for RTCRtpSender, indicates the preferred duration of a media packet in milliseconds. This is typically only relevant for audio encodings. The user agent will try to match this as well as it can.

rid

A DOMString which, if set, specifies an RTP stream ID (RID) to be sent using the RID header extension. This parameter cannot be modified using setParameters(). Its value can only be set when the transceiver is first created.

scaleResolutionDownBy

Only used for senders whose track's kind is "video", this is a double-precision floating-point value specifying a scaling factor to apply while encoding the video. The default value, 1.0, means that the sent video's size will be the same as the original. A value of 2.0 scales the video frames down by a factor of 2 in each dimension, resulting in a video 1/4 the size of the original. The value must not be less than 1.0.

See also

• Introduction to the Real-time Transport Protocol (RTP)
• RTCRtpSender, RTCRtpReceiver, and RTCRtpTransceiver