Need help to test RTP in client side

[Chuck Crisler]

It just seems like there are way too many pieces in your pipeline to me.
There are several things that you need to do.

1. What OS are you using? That is important.
2. You can turn on a lot of debug tracing. That helps you find where and
why it fails. Search on 'gstreamer debugging' and 'gstinfo'.
3. Upgrade to the 1.x GStreamer if you can. The online docs default to the
current release and it is different than the 0.10 code. Here is a link to
the 0.10.14 doc.

http://gstreamer.freedesktop.org/data/doc/gstreamer/stable/manual/html/index.html
4. Here is a Linux script that I use for this testing. I think that the
payload should be 33 (which you have). Please note that this does not
handle audio.

#!/bin/sh

gst-launch udpsrc port=$1 !
'application/x-rtp,media=video,payload=96,clock-rate=90000,encoding-name=H264'\
     ! rtph264depay byte-stream=true ! video/x-h264 ! ffdec_h264 \
    ! ffmpegcolorspace ! xvimagesink sync=false

The '$1' is the first parameter to the script which is the port. This
should be a good start. You can expand from here. You can also use
wireshark (or tcpdump or tshark) to capture and then examine the packets
generated to verify that they are in the format that you expect.



On Wed, Jul 23, 2014 at 6:46 AM, Ravikumar B <ravikumar.embed at gmail.com>
wrote:

> Dear experts.,
>
>      I am new to the gstreamer. i am trying to test / play video file
> using RTP on client side.
>
>      Following setup i have done on my Host machine:
>
>      Server:
>        Run RTP streaming using VLC player.
>
>      Client:
>        How to test RTP on client side..
>
>        I am using following pipeline on client side. for testing on RTP.
>
> *       gst-launch-0.10 -v gstrtpbin name=rtpbin latency=2000
> rtcp-sync=2         buffer-mode=0 udpsrc caps="application/x-rtp,
> media=(string)video, clock-rate=(int)90000, payload=(int)33,
> seqnum-base=(int)49316, clock-base=(int)470688351,
> encoder-name=(string)MP2T-ES " auto-multicast=1  port=1234
> buffer-size=524288 do-timestamp=true blocksize=8192 !
> rtpbin.recv_rtp_sink_0 rtpbin. ! typefind ! rtpmp2tdepay ! mpegtsdemux
> name=demux demux. ! queue2 ! mpegvideoparse ! ffdec_mpeg2video  !
> ffmpegcolorspace qos=true ! videoscale ! autovideosink  demux. ! queue2 !
> ffdec_ac3 ! autoaudiosink *
>        but this pipeline is not working>
>
>      Can you please suggest me how can i make pipeline or test application
> for testing RTP.
>
>
> --
> ...with regards
>     ravikumar
>
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