A couple of RTP questions

[Sebastian Dröge]

On Mi, 2015-12-02 at 13:34 +0000, David Jaggard wrote:
> 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

That means you get packets from pt=33 somehow but those are not seen as
the one you are setting caps for on udpsrc. Just guessing here though,
needs at least a debug log with "2,rtp*:6" to be sure.

> 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?

There is nothing else in the pipeline that would provide a clock, so we
use the system clock instead.

> gst-launch crashes after a few minutes

How does it crash? That would be a separate issue. Do you have a
backtrace?

-- 
Sebastian Dröge, Centricular Ltd · http://www.centricular.com

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