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rtpbin - GStreamer
https://gstreamer.freedesktop.org/documentation/rtpmanager/rtpbin.html
RTP bin combines the functions of , , and in one element. It allows for multiple RTP sessions that will be synchronized together using RTCP SR packets. is configured with a number of request pads that define the functionality that is activated, similar to the element. To use as an RTP receiver, request a recv_rtp_sink_%u pad.
rtpbin: GStreamer Good Plugins 1.0 Plugins Reference …
https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-good/html/gst-plugins-good-plugins-rtpbin.html
Encode and payload AMR audio generated from audiotestsrc. The video is sent to session 0 in rtpbin and the audio is sent to session 1. Video packets are sent on UDP port 5000 and audio packets on port 5002. The video RTCP packets for session 0 are sent on port 5001 and the audio RTCP packets for session 0 are sent on port 5003.
audio - Increase sync rate in Gstreamer rtpbin - Stack ...
https://stackoverflow.com/questions/69028917/increase-sync-rate-in-gstreamer-rtpbin
I am trying to synchronise audio data from two separate audio sources using Gstreamer rtpbin. NTP is disabled on the senders, I am hoping to use the RTCP packets to synchronise the two audio streams. On the receiving end I am using the audiointerleave plugin to save the received microphone data from the two different sources to a WAV file.
rtpbin - freedesktop.org
https://www.freedesktop.org/wiki/Software/Farstream/GstRtpDesign/
rtpbin. The idea is to have an element (rtpbin) that ensures all of the above features while offering a simple and flexible inte rface to the user. rtpbin is a dynamic element that is used for all RTP sessions. For each RTP session, the user needs t o connect to the following request pads : recv_rtp_sink_%d : Incoming RTP stream
gstrtpbin - freedesktop.org
https://www.freedesktop.org/software/gstreamer-sdk/data/docs/latest/gst-plugins-good-plugins-0.10/gst-plugins-good-plugins-gstrtpbin.html
Receive AMR on port 5002, send it through rtpbin in session 1, depayload, decode and play the audio. Receive server RTCP packets for session 0 on port 5001 and RTCP packets for session 1 on port 5003. These packets will be used for session management and synchronisation.
gstrtpbin - Massachusetts Institute of Technology
https://stuff.mit.edu/afs/athena/system/amd64_deb50/os/usr/share/gtk-doc/html/gst-plugins-good-plugins-0.10/gst-plugins-good-plugins-gstrtpbin.html
Receive AMR on port 5002, send it through rtpbin in session 1, depayload, decode and play the audio. Receive server RTCP packets for session 0 on port 5001 and RTCP packets for session 1 on port 5003. These packets will be used for session management and synchronisation.
No RTP retransmission when injecting video and audio …
https://stackoverflow.com/questions/70939802/no-rtp-retransmission-when-injecting-video-and-audio-from-gstreamer-to-mediasoup
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Why does rtpbin example from Gstreamer not work? - Unix ...
https://unix.stackexchange.com/questions/680984/why-does-rtpbin-example-from-gstreamer-not-work
Doing everything from C code seems straightforward but the rtpbin example uses gst-launch-1.0 (covered in one of the basic tutorials ). I couldn't get the rtpbin example to run without errors initially: ffenc_h263 and ffdec_h263 ( WARNING: erroneous pipeline: no element "ffenc_h263" ), so I replaced them with avenc_h263 and avdec_h263 ...
GStreamer RTP Streaming - NXP Community
https://community.nxp.com/t5/i-MX-Processors-Knowledge-Base/GStreamer-RTP-Streaming/ta-p/1126911
GStreamer RTP Streaming. First run the playback pipeline then the streaming pipeline. The above example streams H263 video and AMR audio data. Change codec format to your needs. In case where the iMX is the streaming machine, the audio encoder ' amrnbenc' must be installed before.
gstreamer-rtp-experiments/myRtpReceiver.c at master ...
https://github.com/cnotin/gstreamer-rtp-experiments/blob/master/scripts/myRtpReceiver.c
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