RTP retransmission mechanism in gstreamer 1.2.0

[abhilash]

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] <
ml-node+s966125n4663966h94 at n4.nabble.com> wrote:

> On Wed, Dec 11, 2013 at 10:29 AM, abhilash <[hidden email]<http://user/SendEmail.jtp?type=node&node=4663966&i=0>>
> 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
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