[gst-devel] video on rtp with gstreamer

[Philippe Khalaf]

HI,


On Sat, 06 May 2006 11:02:36 +0800
"ye nan" <bucketxp at hotmail.com> wrote:

> Hi,
>   I try to build up a video stream with rtpbin in gst-plugins-farsight.
>   Sender:
>   gst-launch-0.10 v4lsrc ! video/x-raw-yuv,width=320,height=240 ! 
> ffmpegcolorspace ! video/x-raw-yuv,width=320,height=240 ! ffenc_h263p ! 
> rtph263ppay ! rtpbin localport=10000 destinations=127.0.0.1:10002
>   Receiver:
>   gst-launch-0.10 rtpbin localport=10002 ! rtph263pdepay ! ffdec_h263 ! 
> ffmpegcolorspace ! ximagesink
>   When I started the receiver, I got a error:
>   ** (gst-launch-0.10:29147): CRITICAL **: gst_base_rtp_depayload_chain: 
> assertion `filter->clock_rate > 0' failed
>   I also try to ff***_mpeg4 codecs and rtpmp4v** payloader, but got the 
> same failure.
>   I use the gst-inspect to check more information about rtpbin and then add 
> a capsfilter between rtpbin and rtph263pday in receiving side: 
> application/x-rtp, clock-rate=????. 
Unfortunately rtpbin cannot really be used from gst-launch except for
payloads that are in the default pt table (RFC 3551). Since I can't
define the other payloads that are dynamic they have to be done by the
application through the pt-map property on rtp-bin. This problem is due
to the fact that you cannot determine the caps coming from the network
as rtp packets, all you can read is the pt. Maybe I could add a
property that allows you to specify just one caps as a string for one pt
on rtpbin for quick testing from gst-launch, but for the moment you have
to pass in a pt-map table.

Regards,
Philippe Kalaf