Weird bugs on gstreamer 1.6 and rtp*depay quesion

[Krutskikh Ivan]

Hi everyone,

I'm developing a video surveillance server based on gstreamer. Yesterday we
had a first shipment of h265 ip camera and I considered upgrading our
server software. It turned out that rtph265(pay)depay is available only on
gstreamer 1.6.X ( I had 1.4.X before from opensuse official repo). I
updated my gstreamer stack, but then I stumbled upon some strange bugs:

1) rtspsrc with default protocols fails with

gstrtspsrc.c(6856): gst_rtspsrc_setup_streams ():
/GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0:
Could not setup transport.

This can be fixed with protocols=tcp option on pipeline

2) mp4mux element produces file with 0 duration. When i use ffprobe on new
recorded rtsp stream I get 00:00:00 duration, yet the size is definitely
non zero. This can be fixed if I use matroskamux instread.


Is this a broken repo/library issue or am I doing wrong here?


And other big question: how can I automatically define rtp pay/depay time
for each camera type when recording live video streams? I have several
types: mjpeg over rtp, h264 over rtp, h265 over rtp and possibly mpeg-ts
over http and multipart mjpeg over http.  Right now I have command line
switched for each codec, witch looks awkward as hell.

Thanks in advance!
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