RTP retransmission mechanism in gstreamer 1.2.0

[abhilash]

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 ?



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