[Bug 746479] rtsp: Only two second of playback with rtpsrc and test-mp4 (rtsp-server)

[GStreamer (GNOME Bugzilla)]

https://bugzilla.gnome.org/show_bug.cgi?id=746479

--- Comment #14 from Nicolas Dufresne (stormer) <nicolas.dufresne at collabora.co.uk> ---
This fixes the skew at start, which would confuse the RTP jitter buffer. Though
a skew and an overflow still occurs on seeks.

(gstrtpjitterbuffer.c:2104):calculate_jitter: runtime check failed:
(priv->last_rtptime == -1 || rtptime >= priv->last_rtptime)

>From the spec, after a seek, some buffers of the old segment might be
travelling over the network and we are supposed to drop it base on the initial
rtptime and seq we got from the RTPInfo. I'm not sure yet if we have code for
that, e.g. we have trusty start seq + rtptime, please trust and drop mode. Any
input about how this is supposed to work would be appreciated. I'm not sure how
this could have worked in the past.

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