Please help, I need to convert multiple H.264 rtsp live stream to mpegts rtp simultaneously without a big latency

[Sebastian Dröge]

On Fr, 2016-02-26 at 08:16 -0800, ismail wrote:
> Hi I hope you have a good day and that tomorrow will be better 
> 
> Please help : 
> 
> I need your help to do a multiconversion please and i am a beginer with
> gstreamer : 
> 
> I have 8 camera connected to a debian calculator i have access to my
> calculator via putty/winscp, each camera have an IP and send H264 rtsp
> stream so i can see my H264 videos by opening
> "rtsp://192.168.50.4X:8601/wxHdSdiStreaming/2?videoCodecType=H.264" using
> vlc or ffplay on my computer. Now i need to make my calculator convert all
> my cameras live stream to mpegts and send it using rtp protocol for each
> camera. So i installed gstreamer on my linux calculator using sudo aptitude
> build-dep  python-gst0.10 gstreamer0.10-plugins-good
> gstreamer0.10-plugins-ugly.
> 
> My first question is : do i have to install more gstreamer libraries,
> packages ?

You should use an 1.x version of GStreamer, which is included in the
last stable release of Debian too. 0.10 is no longer maintained since
3+ years now.

>  It is better if i program in c++ or to use command line ? and how
> can i do please?

gst-launch is only a debugging tool and not meant to be used for actual
applications. You can write your code in C or C++, or many other
languages (e.g. Python). Whatever language you prefer.

> I tried other converter but same problems then an expert told me that i wont
> find better than gstreamer to do what i need to do today because i have a
> lot of stream to convert at the same time.

You say you want to convert an RTSP stream containing H264 to MPEG-TS
and send it via RTP. Do you want to send it really via RTP only or do
you want to provide another RTSP stream with it?

But that basic data flow would be to use something like
  rtspsrc location=rtsp://... ! rtph264depay !
    rtph264pay ! ...

where the last ... depends on whether you want plain RTP, RTP with
RTCP, RTSP or whatever else.


Here you can find an example that does RTP with RTCP with VP8 and Opus,
but as input uses videotestsrc and audiotestsrc. In your case you would
omit the audio part and place the videotestsrc and encoding with the
rtspsrc and rtph264depay:
https://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/tests/examples/rtp/server-VTS-VP8-ATS-OPUS.sh


Here you can find an example that takes a gst-launch style pipeline
description and exposes an RTSP server with it:
https://cgit.freedesktop.org/gstreamer/gst-rtsp-server/tree/examples/test-launch.c

-- 
Sebastian Dröge, Centricular Ltd · http://www.centricular.com
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 949 bytes
Desc: This is a digitally signed message part
URL: <https://lists.freedesktop.org/archives/gstreamer-devel/attachments/20160227/ad2a2da5/attachment.sig>