[gst-devel] problems in streaming H.264 content using gstreamer

[Agarwal, Lomesh]

-----Original Message-----
From: gstreamer-devel-bounces at lists.sourceforge.net [mailto:gstreamer-devel-bounces at lists.sourceforge.net] On Behalf Of Wim Taymans
Sent: Friday, August 29, 2008 1:58 AM
To: Discussion of the development of GStreamer
Subject: Re: [gst-devel] problems in streaming H.264 content using gstreamer

On Thu, 2008-08-28 at 11:38 -0700, Agarwal, Lomesh wrote:
> I am trying to stream/render H.264 content using gstreamer on two Ubuntu machines. Here is what I run on the server (streaming) side -
> gst-launch -v gstrtpbin name=rtpbin filesrc location=h264.ts ! \
>                  queue2 max-size-buffers=65535 ! \
>                  rtpmp2tpay ! \
>                  queue2 max-size-buffers=65535 ! \
>                  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=5005 ! rtpbin.recv_rtcp_sink_0 }
> On the receiver side I run following command to render it -
> gst-launch -v gstrtpbin name=rtpbin udpsrc port=5000 caps="application/x-rtp,media=video,clock-rate=90000,encoding-name=mpegts" ! \
>                  rtpbin.recv_rtp_sink_0 rtpbin. ! \
>                  identity sync=true silent=true ! \
>                  rtpmp2tdepay ! \
>                  flutsdemux name=demuxer \
>                      demuxer. ! queue max-size-buffers=0 max-size-time=0 ! flumcaacdec ! audioconvert ! volume volume=10 ! autoaudiosink \
>                  demuxer. ! queue max-size-buffers=0 max-size-time=0 ! fluh264dec! autovideosink \
>                  udpsrc port=5001 ! \
>                  rtpbin.recv_rtcp_sink_0 \
>                  rtpbin.send_rtcp_src_0 ! \
>                  udpsink port=5005 sync=false async=false -t
> h264.ts is a MPEG TS file with aac audio format and H.264 video format. I am using commercial versions of Fluendo plugins.
> On the receiver side I see jerky video and after some time video stops. I am using a quad core machine on the receiver side.
> If I repeat the same experiment using tcpserversink (in place of rtp bin) plugin on the streamer side and tcpclientsrc on the receiver side, everything works ok.
> Am I missing setting some rtp bin property or is it a problem with rtp bin? Can someone suggest some experiment which can narrow down the problem?
>

2 problems, timestamps are not correctly set on the sender side (use
mpegtsparse in front of the payloader?).
[Agarwal, Lomesh] adding mpegtsparse doesn't help.
 The second problem is that the
demuxer might not correctly preserve timing information on the receiver.
Also: http://bugzilla.gnome.org/show_bug.cgi?id=548979
[Agarwal, Lomesh] any pointers on how to resolve or workaround this bug?

Wim


> Thanks,
> Lomesh
>
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