[Bug 730039] rtpjitterbuffer: jitter buffer stalls if first seqnum is different than seqnum-base

[GStreamer (bugzilla.gnome.org)]

https://bugzilla.gnome.org/show_bug.cgi?id=730039
  GStreamer | gst-plugins-good | git

Aleix Conchillo Flaqué <aleix> changed:

           What    |Removed                     |Added
----------------------------------------------------------------------------
             Status|UNCONFIRMED                 |RESOLVED
         Resolution|                            |NOTABUG

--- Comment #10 from Aleix Conchillo Flaqué <aleix at oblong.com> 2014-05-14 17:41:04 UTC ---
(In reply to comment #9)

> > 
> > In any case, what happens if packet with seqnum-base is lost?
> 
> It should probably estimate when it should be received and wait until the last
> moment before
> generating a lost-packet event.
> 
> We don't do anything like this, it would involve estimating a time in the past
> using seqnum-base+1
> timestamp and seqnum-base+2 packet spacing. not sure it is worth it. What
> happens now is that
> seqnum-base+1 is delayed as much as possible (waiting for seqnum-base) but
> there is never a lost-packet 
> event generated for seqnum-base.

Right. Thanks for the comment.

I just verified that this bug is not an issue anymore and that it was a
consequence of bug 730078.

I verfied it by sending a sequence number a bit lower than the first packet in
RTP-Info
(gst-rtsp-server/rtsp-stream-transport.c::gst_rtsp_stream_transport_get_rtpinfo).

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