RTP retransmission mechanism

[thxjd]

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-3]

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.
>
>
>
> --
> Sent from: http://gstreamer-devel.966125.n4.nabble.com/
> _______________________________________________
> gstreamer-devel mailing list
> [hidden email]
> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel

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Olivier Crête
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