It's related to the commit http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/gst/rtp/gstrtph264depay.c?id=ef7863355c7c7eca24fbd92d6457a91c02f042ac

In gst_rtp_h264_set_src_caps, in case that byte_stream is true and sps and pps is NULL, this logic just does as below except for sps, pps insertion logic

----------------------------------------------
codec_data = gst_buffer_new_and_alloc (len);
gst_buffer_map (codec_data, &map, GST_MAP_WRITE);
data = map.data;

gst_buffer_unmap (codec_data, &map);
if (rtph264depay->codec_data)
  gst_buffer_unref (rtph264depay->codec_data);
rtph264depay->codec_data = codec_data;
---------------------------------------------
In this case, codec_data is gst buffer, which has NULL. (because len is 0)
This causes failure when playing.


Reproduce step : gst-rtsp-server / gst-launch playbin.