RTP retransmission mechanism

[Olivier Crête]

Hi,

Instead of using rtprtxqueue, you should use rtprtxsend &
rtprtxreceive, see the examples here:


https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-good/html/gst-plugins-good-plugins-rtprtxreceive.html#GstRtpRtxReceive

Olivier

On Tue, 2018-10-09 at 03:39 -0500, thxjd wrote:
> Hi, I am trying the rtp retransmission mechanism on my jetson tx2 and
> the
> pipelines are mentioned below: 
> 
> Sender: 
> gst-launch-1.0 -v rtpbin name=rtpbin audiotestsrc freq=1000 !
> audioconvert !
> alawenc ! rtppcmapay ! rtprtxqueue ! rtpbin.send_rtp_sink_0
> rtpbin.send_rtp_src_0 ! udpsink host=192.154.10.20 port=5000
> rtpbin.send_rtcp_src_0 ! udpsink host=192.154.10.20 port=5001
> sync=false
> udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 
> 
> Receiver: 
> gst-launch-1.0 -v rtpbin name=rtpbin do-retransmission=true udpsrc
> port=5000
> caps="application/x-rtp, media=(string)audio, clock-rate=(int)8000,
> encoding-name=(string)PCMA, payload=8" ! rtpbin.recv_rtp_sink_0
> rtpbin. !
> rtppcmadepay ! alawdec ! alsasink sync=false udpsrc port=5001 !
> rtpbin.recv_rtcp_sink_0 rtpbin.send_rtcp_src_0 ! udpsink
> host=192.154.10.21
> port=5005 sync=false
> 
> If I set 10% packet loss on network, I still hear audio glitch and
> can't
> receive any rtcp packet in sender side according wireshark. Is there
> something wrong? Thank you very much.
> 
> 
> 
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-- 
Olivier Crête
olivier.crete at collabora.com