Issue while decoding an RTP stream

[Graham Leggett]

On 10 Jan 2017, at 10:45 PM, Jon Mandrell <Jon.Mandrell at zii.aero> wrote:

> I am using gstreamer 1.10.2, with the following pipeline:
> gst-launch-1.0 -v udpsrc uri=udp://239.192.16.11:52416 <udp://239.192.16.11:52416> do-timestamp=true ! application/x-rtp,media=video,clock-rate=90000 ! rtpsession ! rtpssrcdemux ! rtpjitterbuffer latency=1000 do-lost=true ! rtpptdemux ! rtpmp2tdepay ! tsdemux ! audio/mpeg ! queue ! avdec_aac ! audioconvert ! alsasink sync=true
> 
> The stream I am tuning to has both audio and video, but for the sake of this email I've simplified this to just the audio portion of the pipeline.
> About 35 seconds into the stream there is a group of dropped packets (I do see a discontinuity indicated coming out of the tsdemux element). However, there is some issue later on in the decoder where it just stops. The end result is that the audio stops at the dropped packets and doesn't pick up again. This also occurs with the video stream, but I figure the audio gives a fairly simplified example.
> 
> I have done the same thing with a 0.10 pipeline and it recovers like I would expect.
> 
> I'm at a loss here of where to go with this - I have a requirement that I be able to play this stream (with video/audio glitches) and not just stopping like it currently does. Any suggestions would be very welcome!

I have suffered similar behaviour on live streams, and recently this bug got fixed:

https://bugzilla.gnome.org/show_bug.cgi?id=774654

While the above bug affected gst-omx, this same bug might exist elsewhere.

Regards,
Graham
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