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    GObject
    ╰── GInitiallyUnowned
        ╰── GstObject
            ╰── GstElement
                ╰── GstRTPBasePayload
                    ╰── GstRTPBaseAudioPayload
                        ╰── GstRtpL16Pay
Payload raw audio into RTP packets according to RFC 3551. For detailed information see: http://www.rfc-editor.org/rfc/rfc3551.txt
| 1 | gst-launch-1.0 -v audiotestsrc ! audioconvert ! rtpL16pay ! udpsink | 
| plugin | rtp | 
| author | Wim Taymans <wim.taymans@gmail.com> | 
| class | Codec/Payloader/Network/RTP | 
| name | sink | 
| direction | sink | 
| presence | always | 
| details | audio/x-raw, format=(string)S16BE, layout=(string)interleaved, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ] | 
| name | src | 
| direction | source | 
| presence | always | 
| details | application/x-rtp, media=(string)audio, payload=(int)[ 96, 127 ], clock-rate=(int)[ 1, 2147483647 ], encoding-name=(string)L16, channels=(int)[ 1, 2147483647 ] | 
| application/x-rtp, media=(string)audio, encoding-name=(string)L16, payload=(int)10, clock-rate=(int)44100 | |
| application/x-rtp, media=(string)audio, encoding-name=(string)L16, payload=(int)11, clock-rate=(int)44100 |