[gst-devel] Re : How can I get detailed information about those packets dropped by rtspsrc ?

[Aurelien Grimaud]

Hi, there is already a gstrtpjitterbuffer inside rtspsrc, so no need for a second one.

You can run your gst-launch with GST_DEBUG=3 or 4 or 5 to track the received buffers.
Are you sure RTP is sent correctly ? received correctly ? 
packets are lost in your network ? 
network card fails ...

Check sent packet with wireshark on sending host 10.38.164.205.
Check received packet with wireshark on receiving host.
wireshark will decode RTP, you will be able to see wether there are packet drops in your network or not.


Initial file (water_mpeg4.mp4) contains only video ?
How do you find out that a quarter of the stream is received ?
Could remaining 3/4  be audio ?

You can use "rtpmp4vdepay ! mpeg4videoparse ! ffmux_mov ! filesink" to get a viewable file and see if those dropped buffers are really dropped.

Aurelien


----- Message d'origine ----
De : Ting Wang <wangting at gmail.com>
À : gstreamer-devel at lists.sourceforge.net
Envoyé le : Mardi, 12 Août 2008, 11h17mn 54s
Objet : [gst-devel] How can I get detailed information about those packets dropped by rtspsrc ?


Hi, 

I'm using Gstreamer to setup a streaming video client using rtspsrc. Thanks to the help from Aurelien, I successfully setup the pipeline and it's rolling good. Now I find that with the packet drop is quite frequent even in locale area network. I have been tried to configure the parameter "latency" on rtspsrc/gstrtpjitterbuffer, but they didn't help. The video source rate is 788kbps, lasts 6 seconds. By dumping the received file to disk, it turned out that only a quarter of the stream was successfully received ("gst-launch -v rtspsrc location=rtsp://10.38.164.205/water_mpeg4.mp4 ! gstrtpjitterbuffer latency=20000 ! rtpmp4vdepay ! filesink location=./stream.m4v"). Most of them was dropped for some reason I don't know. How can I get information regarding those drooped packets? Thank you for your advice :)

Best regards

-felix



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