[gst-devel] H.264 content streaming

[Agarwal, Lomesh]

I am streaming H.264 content (in MPEG TS) between two Ubuntu machines using following pipelines -
Server -
gst-launch -v gstrtpbin name=rtpbin filesrc location=h264.ts ! \
                 queue2 max-size-buffers=65535 ! \
                 h264parse ! \
                 rtpmp2tpay ! \
                 queue2 max-size-buffers=65535 ! \
                 identity sync=true silent=true sleep-time=300 ! \
                 rtpbin.send_rtp_sink_0 \
                 rtpbin.send_rtp_src_0 ! udpsink host=192.168.1.2 port=5000 \
                 rtpbin.send_rtcp_src_0 ! udpsink host=192.168.1.2 port=5001 sync=false async=false \
                 udpsrc port=5002 ! rtpbin.recv_rtcp_sink_0
client -
gst-launch -v gstrtpbin name=rtpbin udpsrc buffer-size=200000 port=5000 caps="application/x-rtp,media=video,clock-rate=90000,encoding-name=mpegts" ! \
                 rtpbin.recv_rtp_sink_0 rtpbin. ! \
                 rtpmp2tdepay ! \
                 flutsdemux name=demuxer \
                 demuxer. ! queue2 max-size-buffers=0 max-size-time=0 ! fluh264dec ! autovideosink \
                 udpsrc port=5001 ! \
                 rtpbin.recv_rtcp_sink_0 \
                 rtpbin.send_rtcp_src_0 ! \
                 udpsink host=192.168.1.1 port=5002 sync=false async=false -t
If I don't use the identity element on server and don't increase the buffer-size on client (text in red) then on client side video is jittery and after some time client pipeline dies. Can someone clarify why does it work with the text in red?

Thanks,
Lomesh
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