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[Paolo Bolzoni]

Dear gstreamer list,
I am writing a little application to send a video stream and an audio stream
via RTP using gstreamer. It should not be important, but I am using those Java
bindings http://code.google.com/p/gstreamer-java/ .

Reading about in the pipelines described here
http://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/tests/examples/rtp
I had almost no problem implementing sender and client for both audio and
video.

So the programs work, but:
- I completely missed the idea of CAPS negotiation, it is not a great deal
because at the moment both sender and client are programs of mine so
I can just set the caps in the client, but I am curious how it should work.
What can I read? Where it is documented?

- More importantly, I would like to change the bitrate dynamically according
the number of packets lost during the transmission.
I did not tried yet, but I guess it is not a problem changing ffenc_h263 bitrate

parameter while the pipeline is running, right?

The RTCP protocol do specify a SR packet with the information I need
( http://freesoft.org/CIE/RFC/1889/19.htm ) but I am missing how I should
query the sender's rtpbin to get it. Or I should parse the packet manually?

Thanks,
Paolo