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    GObject
    ╰── GInitiallyUnowned
        ╰── GstObject
            ╰── GstElement
                ╰── GstRTPBasePayload
                    ╰── GstRtpAMRPay
Payload AMR audio into RTP packets according to RFC 3267. For detailed information see: http://www.rfc-editor.org/rfc/rfc3267.txt
| 1 | gst-launch-1.0 -v audiotestsrc ! amrnbenc ! rtpamrpay ! udpsink | 
| plugin | rtp | 
| author | Wim Taymans <wim.taymans@gmail.com> | 
| class | Codec/Payloader/Network/RTP | 
| name | sink | 
| direction | sink | 
| presence | always | 
| details | audio/AMR, channels=(int)1, rate=(int)8000 | 
| audio/AMR-WB, channels=(int)1, rate=(int)16000 | 
| name | src | 
| direction | source | 
| presence | always | 
| details | application/x-rtp, media=(string)audio, payload=(int)[ 96, 127 ], clock-rate=(int)8000, encoding-name=(string)AMR, encoding-params=(string)1, octet-align=(string)1, crc=(string)0, robust-sorting=(string)0, interleaving=(string)0, mode-set=(int)[ 0, 7 ], mode-change-period=(int)[ 1, 2147483647 ], mode-change-neighbor=(string){ 0, 1 }, maxptime=(int)[ 20, 2147483647 ], ptime=(int)[ 20, 2147483647 ] | 
| application/x-rtp, media=(string)audio, payload=(int)[ 96, 127 ], clock-rate=(int)16000, encoding-name=(string)AMR-WB, encoding-params=(string)1, octet-align=(string)1, crc=(string)0, robust-sorting=(string)0, interleaving=(string)0, mode-set=(int)[ 0, 7 ], mode-change-period=(int)[ 1, 2147483647 ], mode-change-neighbor=(string){ 0, 1 }, maxptime=(int)[ 20, 2147483647 ], ptime=(int)[ 20, 2147483647 ] |