Failed in RTP and RTSP over HTTP

s_kamiya at toa.co.jp s_kamiya at toa.co.jp
Wed Mar 30 15:01:31 UTC 2016


Hi, Nicola

Thank you for your reply.

I would like to try in gstreamer ver.1.8.0.
I am using the MSYS2, but I could only upgrade to Ver.1.6.3 the gstreamer 
in the next command.

  update-core
  pacman -Syu

If you know how to upgrade the gstreamer to ver.1.8.0 in MSYS2, please 
tell me.

If you can not achieve in the current version of MSYS2, please tell me the 
other ways that you can run the gst-launch of gstreamer ver.1.8.0 in 
Windows.

Thank you for your attention.


"gstreamer-devel" <gstreamer-devel-bounces at lists.freedesktop.org> wrote on 
2016/03/30 16:06:51:

> 
> Hi,

> this is a know bug in 1.6.x, gstreamer is unable to authenticate when
> using rtsp over http, it works with 1.6.x if you disable authentication
> on the camera and it works with and without authentication using 1.8.x

> so I suggest to upgrade your gstreamer installation to 1.8 and retry,

> Nicola

> Il 30/03/2016 07:04, s_kamiya at toa.co.jp ha scritto:
> > My name is geysee.
> > Nice to meet you.
> >
> > I am trying to streaming playback of network camera using the 
gst-launch.
> > I have succeeded in "RTSP", however, I failed in "RTP and RTSP over 
HTTP".
> > On the other hand, if I used the VLC, I was successful even "RTP and 
RTSP
> > over HTTP".
> > So that I can succeed "RTP and RTSP over HTTP" in the gst-launch, 
please
> > your advice.
> > Details are described below.
> > Thank you for your attention.
> >
> > ### Successful ###
> >
> > gst-launch command
> >
> >    gst-launch-1.0.exe -v rtspsrc location =
> > rtsp://root:root@10.107.14.2/axis-media/media.amp videocodec = h264
> > latency = 100 rtph264depay avdec_h264 autovideosink
> >
> > VLC
> >    1. VLC setting : Check the "HTTP over tunnel RTSP and RTP".
> >    2. VLC setting : Set network URL to
> > "rtsp://root:root@10.107.14.2/axis-media/media.amp.
> >    3. Play.
> >
> >
> > ## failed ###
> >
> > gst-launch command
> >
> >    gst-launch-1.0.exe -v rtspsrc debug = TRUE protocol = 20 location =
> > rtsph://root:root@10.107.14.2/axis-media/media.amp videocodec = h264
> > latency = 100 rtph264depay avdec_h264 autovideosink
> >
> >    gst-launch-1.0.exe -v rtspsrc debug = TRUE protocol = 31 location =
> > rtsph://root:root@10.107.14.2:?!! 80 / axis-media / media.amp 
videocodec =
> > h264 latency = 100 rtph264depay avdec_h264 ! autovideosink
> >
> >    gst-launch-1.0.exe -v rtspsrc debug = TRUE protocols = 31 location 
=
> > rtsph://root:root@10.107.14.2/axis-media/media.amp videocodec = h264 
proxy
> > = http: //10.107.14.23 :!!! 80 proxy-id = admin proxy-pw = guest 
latency =
> > 100 rtph264depay avdec_h264 autovideosink
> >
> > *I tried set protocols option to 1,2,4,8,15,16,20,31,32. but failed.
> >
> >
> > ### Environment ###
> >
> > gstreamer
> >    local / mingw-w64-i686-clutter-gst 3.0.14-1
> >    local / mingw-w64-i686-gst-editing-services 1.4.0-1
> >    local / mingw-w64-i686-gst-libav 1.6.3-1
> >    local / mingw-w64-i686-gst-plugins-bad 1.6.3-1
> >    local / mingw-w64-i686-gst-plugins-base 1.6.3-1
> >    local / mingw-w64-i686-gst-plugins-good 1.6.3-1
> >    local / mingw-w64-i686-gst-plugins-ugly 1.6.3-1
> >    local / mingw-w64-i686-gstreamer 1.6.3-1
> >
> > PC
> >    Windows 7 Professional SP1 32bit
> >
> > camera
> >    AXIS P3301
> > _______________________________________________
> > gstreamer-devel mailing list
> > gstreamer-devel at lists.freedesktop.org
> > https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel

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