A couple of RTP questions

[David Jaggard]

GStreamer 1.6.1 on Windows

Here is a test pipeline:
gst-launch-1.0.exe rtpbin name=rtpbinout udpsrc
caps=application/x-rtp,media=video,clock-rate=(int)90000
address=239.192.65.121 port=5000 ! rtpbin ! queue2 ! rtpmp2tdepay ! tsdemux
name=demux ! queue2 ! video/x-h264 ! decodebin ! deinterlace fields=1
method=3 ! x264enc ! video/x-h264,profile=baseline ! queue2 ! rtph264pay !
rtpbinout.send_rtp_sink_0 rtpbinout.send_rtp_src_0 ! multiudpsink clients=
227.0.0.82:50090 ttl-mc=100 demux. ! queue2 ! audio/mpeg ! decodebin !
audioconvert ! audio/x-raw,channels=1 ! audioresample !
audio/x-raw,rate=48000 ! voaacenc bitrate=128000 ! audio/mpeg,mpegversion=4
! queue2 ! rtpmp4gpay pt=97 ! rtpbinout.send_rtp_sink_1
rtpbinout.send_rtp_src_1 ! multiudpsink clients=227.0.0.82:50092 ttl-mc=100

The source mpeg2 ts containing h264 video and aac audio - the transcoding
is necessary for a particular client application.

Question 1:
Why does it generate a constant stream of these warnings?
rtpsource rtpsource.c:950:calculate_jitter: cannot get clock-rate for pt 33

Question 2:
Why does the pipeline use a GstSystemClock? I would have though it would
clock off the live source. Is this related to question 1?

gst-launch crashes after a few minutes
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