Using rtpjitterbuffer to handle RTP packets based on their timestamp only (mode 0)

[Andrea Lo Pumo]

I have a dump of RTP traffic (a single ssrc stream) and I would like to
decode it with gstreamer.
What I am trying works, but it fails if there are jumps in the timestamps
because there is silence in between (no packets sent during the silence).
The result is an audio without the right silence.
Note that the RTP timestamp advances correctly, so one knows how much time
passed before the jump and after it.
Thus, I was trying "rtpjitterbuffer mode=0" however, when I use mode=0, I
get the following:
....
INFO         rtpjitterbuffer
gstrtpjitterbuffer.c:2559:calculate_expected:<rtpjitterbuffer0> lost
packets (1, #23086->#23086) duration too large 0:00:00.800000000 >
0:00:00.200000000, consider 0 lost (0:00:00.200000000)
INFO         rtpjitterbuffer
gstrtpjitterbuffer.c:2559:calculate_expected:<rtpjitterbuffer0> lost
packets (2, #23112->#23113) duration too large 0:00:00.740000000 >
0:00:00.200000000, consider 1 lost (0:00:00.293333332)
...
INFO                 basesrc
gstbasesrc.c:2839:gst_base_src_loop:<rtpfilesrc0> pausing after
gst_base_src_get_range() = eos
INFO                    task gsttask.c:316:gst_task_func:<rtpfilesrc0:src>
Task going to paused

and at this point it remains blocked.

This is the pipeline I am using:

    gst-launch-1.0 rtpfilesrc location=/tmp/input.rtp ! "application/x-rtp,
media=(string)audio, clock-rate=(int)8000, encoding-name=(string)PCMA" !
rtpjitterbuffer mode=0 max-rtcp-rtp-time-diff=-1 ! decodebin ! wavenc !
filesink location=/tmp/out.wav

Rtpfilesrc is a plugin I have written that reads an RTP dump and becomes a
source of UDP packets. I used PushSrc as the base class and have overridden
"fill".
In init() I am using the following configuration:
  gst_base_src_set_live (GST_BASE_SRC (rtpfilesrc), TRUE);
  gst_base_src_set_format (GST_BASE_SRC (rtpfilesrc), GST_FORMAT_TIME);
  // do not make rtpfilesrc set timestamps on outgoing buffers based on the
running_time when they were captured
  gst_base_src_set_do_timestamp (GST_BASE_SRC (rtpfilesrc), FALSE);

What should I do?

Thank you in advance.
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