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List: gstreamer-devel
Subject: Re: RTP retransmission mechanism in gstreamer 1.2.0
From: George Kiagiadakis <kiagiadakis.george () gmail ! com>
Date: 2013-12-11 15:31:18
Message-ID: CAFPG5sn9ju2fxdQUddAvRi2eD0GfD1ErmTajeib0Uqe+HWifLA () mail ! gmail ! com
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Hi,
On Wed, Dec 11, 2013 at 11:15 AM, abhilash <abhilashtuse@gmail.com> wrote:
> Hi,
>
> Thanks for your reply.
> As you suggested I tried using rtprtxqueue on receiver side(pipeline below):
>
> gst-launch-1.0 -v rtpbin name=rtpbin udpsrc port=5000
> caps="application/x-rtp, media=audio, clock-rate=8000, encoding-name=PCMA,
> payload=8" ! rtpbin.recv_rtp_sink_0 rtpbin. ! rtprtxqueue ! rtppcmadepay !
> alawdec !
>
> alsasink udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 rtpbin.send_rtcp_src_0
> ! udpsink host=192.154.10.21 port=5005 sync=false async=false
> --gst-debug=3
>
> Still I am not able to see any RTCP packets other than reports from receiver
> to sender in wireshark.
> Is above pipeline correct?
> As per my understanding there should be RTCP request from receiver to sender
> to send again lost packets.
> Is this correct?
>
You misunderstood what I said. I said you need to use the
do-retransmission=true on the receiver side and that rtprtxqueue needs
to be added after the *pay*loader, which lives on the sender side.
Regards,
George
On Wed, Dec 11, 2013 at 11:15 AM, abhilash <abhilashtuse@gmail.com> wrote:
> Hi,
>
> Thanks for your reply.
> As you suggested I tried using rtprtxqueue on receiver side(pipeline below):
>
> gst-launch-1.0 -v rtpbin name=rtpbin udpsrc port=5000
> caps="application/x-rtp, media=audio, clock-rate=8000, encoding-name=PCMA,
> payload=8" ! rtpbin.recv_rtp_sink_0 rtpbin. ! rtprtxqueue ! rtppcmadepay !
> alawdec !
>
> alsasink udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 rtpbin.send_rtcp_src_0
> ! udpsink host=192.154.10.21 port=5005 sync=false async=false
> --gst-debug=3
>
> Still I am not able to see any RTCP packets other than reports from receiver
> to sender in wireshark.
> Is above pipeline correct?
> As per my understanding there should be RTCP request from receiver to sender
> to send again lost packets.
> Is this correct?
>
> Thanks,
> Abhilash.
>
>
>
> On Wed, Dec 11, 2013 at 3:21 PM, George Kiagiadakis [via GStreamer-devel]
> <[hidden email]> wrote:
> >
> > On Wed, Dec 11, 2013 at 10:29 AM, abhilash <[hidden email]> wrote:
> >
> > > Hi,
> > >
> > > I am trying out a client-server based audio streaming application
> > > using
> > > gstreamer 1.2.0 version, running on 2 different Linux machines on
> > > network. I
> > > want to check the functionality of retransmissions as mentioned in the
> > > "rtpbin" element and I tried setting the property
> > > "do-retransmissions=true".
> > > I am simulating packet loss, packet delay & reorder on the server side.
> > > The
> > > observation is as follows -
> > > 1. I am able to see RTCP Sender and Receiver reports being sent between
> > > server and client on Wireshark.
> > > 2. I am able to see dropped RTP packet sequence numbers in Wireshark
> > > trace,
> > > however I am not able to see any RTP retransmissions event going from
> > > client
> > > to server.
> > >
> > > The pipelines at the server and client side are as mentioned below -
> > >
> > > Server:
> > > gst-launch-1.0 -v rtpbin name=rtpbin do-retransmission=true audiotestsrc
> > > freq=1000 ! audioconvert ! alawenc ! rtppcmapay ! 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
> > > async=false udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 --gst-debug=3
> > >
> > > Client:
> > > gst-launch-1.0 -v rtpbin name=rtpbin udpsrc port=5000
> > > caps="application/x-rtp, media=audio, clock-rate=8000,
> > > encoding-name=PCMA,
> > > payload=8" ! rtpbin.recv_rtp_sink_0 rtpbin. ! rtppcmadepay ! alawdec !
> > > alsasink udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0
> > > rtpbin.send_rtcp_src_0
> > > ! udpsink host=192.154.10.21 port=5005 sync=false async=false
> > > --gst-debug=3
> > >
> > > Can anyone please let me know if anything is missed out above and
> > > why I
> > > am not seeing any RTP retransmissions ?
> > >
> > Hi,
> >
> > do-retransmission=true should be set on the receiver (client) rtpbin.
> > On the sender it has no effect. Also, since you are on 1.2, you need
> > to add rtprtxqueue in your pipeline after the payloader. Note though
> > that this does not implement RFC4588-compliant retransmission packets.
> > An RFC4588-compliant implementation is under review in bugzilla at the
> > moment.
> >
> > Regards,
> > George
> > _______________________________________________
> > gstreamer-devel mailing list
> > [hidden email]
> > http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
> >
> >
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> > below:
> >
> > http://gstreamer-devel.966125.n4.nabble.com/RTP-retransmission-mechanism-in-gstreamer-1-2-0-tp4663965p4663966.html
> > To unsubscribe from RTP retransmission mechanism in gstreamer 1.2.0, click
> > here.
> > NAML
>
>
>
> ________________________________
> View this message in context: Re: RTP retransmission mechanism in gstreamer
> 1.2.0
>
> Sent from the GStreamer-devel mailing list archive at Nabble.com.
>
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