[gst-devel] Simple pipeline with rtp and udp

[Wim Taymans]

On Mon, 2008-10-06 at 23:37 -0700, Jon Dufresne wrote:
> > For theora to work over RTP, you need to carry the content of the
> > "configuration" value from the caps on the sender side to the receiver
> > side. That should preferably be done in your signalling. Or you can use
> > an older codec liek h263 that does not require transmitting a header.
> 
> Thank you for your response. I had a chance to try your suggestion. I
> am going to try using h263. I changed my two pipelines to the
> following:
> 
> server:
> gst-launch \
> 	videotestsrc ! \
> 	ffenc_h263 ! \
> 	rtph263pay ! \
> 	udpsink
> 
> client:
> gst-launch --verbose \
> 	udpsrc ! \
> 	rtph263depay ! \
> 	ffdec_h263 ! \
> 	xvimagesink
> 
> This is what I see when I first run the server in one terminal, and
> then the client in a different terminal.

This is not going to work, see below.

> 
> $ ./server
> 
> ** (gst-launch-0.10:2874): WARNING **: Unknown codec ID 86022, please add here
> Setting pipeline to PAUSED ...
> Pipeline is PREROLLING ...
> Pipeline is PREROLLED ...
> Setting pipeline to PLAYING ...
> New clock: GstSystemClock
> 
> 
> $ ./client
> 
> ** (gst-launch-0.10:2878): WARNING **: Unknown codec ID 86022, please add here
> Setting pipeline to PAUSED ...
> Pipeline is live and does not need PREROLL ...
> Setting pipeline to PLAYING ...
> New clock: GstSystemClock
> ERROR: from element /pipeline0/ffdec_h2630: Internal GStreamer error:
> negotiation problem.  Please file a bug at
> http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer.
> Additional debug info:
> gstffmpegdec.c(2313): gst_ffmpegdec_chain (): /pipeline0/ffdec_h2630:
> ffdec_h263: input format was not set before data start
> Execution ended after 29165444 ns.
> Setting pipeline to PAUSED ...
> Setting pipeline to READY ...
> Setting pipeline to NULL ...
> FREEING pipeline ...
> 
> 
> 
> My first question is, why am I getting this unknown coded ID error?
> And is this something to be concerned about?

You are building against a systemwide ffmpeg, which is not supported.

> 
> Second. Did I really come across a gstreamer bug and should I file it as such?

Very unlikely, your example pipeline all seems to function as designed.
Hint, you have no codec signaling going on between the sender and
receiver.

> 
> In the end my pipelines still do not produce the desired result, any
> idea how I can fix this?

You should read
http://webcvs.freedesktop.org/gstreamer/gst-plugins-good/gst/rtp/README?revision=1.13&view=markup

Go to the part 'usage with udp'

Wim

> 
> Thanks,
> Jon
> 
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