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

[Ting Wang]

Aurelien,

Thank you for your help. I enabled GST_DEBUG on gstrtpjitterbuffer and
udpsrc. From the log, I can see part of the UDP packets are missing (there
exists gap of the packet number between consecutive packets that are pushed
into jitterbuffer). If I force rtspsrc to use TCP for all transmissions,
then everything is perfect and no packet is dropped thanks to TCP's flow
control and retransmission. Since I'm using Darwin Streaming Server, I guess
the server is pretty aggresive in sending out UDP packet, my client board
cannot catch up the incoming UDP packets and packet drop happens. Maybe I
have to find out someway to control the sending rate at the server side when
using UDP.

Best regards,
Felix

On Wed, Aug 13, 2008 at 2:33 AM, Aurelien Grimaud <gstelzz at yahoo.fr> wrote:

> 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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