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Failed in RTP and RTSP over HTTP

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s_kamiya

Failed in RTP and RTSP over HTTP

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
[hidden email]
https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel

Mailing List SVR

Re: Failed in RTP and RTSP over HTTP

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, [hidden email] 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
> [hidden email]
> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel

_______________________________________________
gstreamer-devel mailing list
[hidden email]
https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel

s_kamiya

Re: Failed in RTP and RTSP over HTTP

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" <[hidden email]> 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, [hidden email] 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
> > [hidden email]
> > https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel

> _______________________________________________
> gstreamer-devel mailing list
> [hidden email]
> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
_______________________________________________
gstreamer-devel mailing list
[hidden email]
https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel

Mailing List SVR

Re: Failed in RTP and RTSP over HTTP

Il 30/03/2016 17:01, [hidden email] ha scritto:

> 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.

Hi,

give a try to the official windows binaries:

https://gstreamer.freedesktop.org/data/pkg/windows/1.8.0/

Nicola

>
>
> "gstreamer-devel" <[hidden email]> 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, [hidden email] 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
>>> [hidden email]
>>> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>> _______________________________________________
>> gstreamer-devel mailing list
>> [hidden email]
>> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
> _______________________________________________
> gstreamer-devel mailing list
> [hidden email]
> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel

_______________________________________________
gstreamer-devel mailing list
[hidden email]
https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel

s_kamiya

Re: Failed in RTP and RTSP over HTTP

Hi, Nicola

Thank you for your reply.

I was able to run the gst-launch in gstreamer ver.1.8.0 of official
windows binaries.

And, I was able to receive the "RTP and RTSP over HTTP" in the next
command.

.\gst-launch-1.0.exe -v rtspsrc debug=TRUE
location=rtsph://root:root@10.107.14.2:80/axis-media/media.amp?videocodec=h264
latency=100 ! rtph264depay ! avdec_h264 ! autovideosink

Thank you very much !

"gstreamer-devel" <[hidden email]> wrote on
2016/03/31 00:09:25:

> > 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.

> Hi,

> give a try to the official windows binaries:

> https://gstreamer.freedesktop.org/data/pkg/windows/1.8.0/

> Nicola

> >
> >
> > "gstreamer-devel" <[hidden email]>
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, [hidden email] 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
> >>> [hidden email]
> >>> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
> >> _______________________________________________
> >> gstreamer-devel mailing list
> >> [hidden email]
> >> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
> > _______________________________________________
> > gstreamer-devel mailing list
> > [hidden email]
> > https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel

s_kamiya

Re: Failed in RTP and RTSP over HTTP

Hi, Nicola

I have gotten the advice previously, I succeeded in the reception of the
"RTP and RTSP over HTTP" in gstreamer ver.1.8.0.

I used the following command at that time.

.\Gst-launch-1.0.exe -v rtspsrc debug = TRUE location =
rtsph://root:root@10.107.14.2:80 / axis-media / media.amp videocodec =
h264 latency = 100 ! rtph264depay ! avdec_h264 ! autovideosink

However, depending on the camera manufacturers, it does not accept the
user name and password is embedded URL.
For example, TOA company's Onvif camera "N-C3120" does not accept this
URL.

Please tell me how to tell a user name and password to the camera in other
ways that are not URL using the gst-launch.

I tried the following command, but failed both AXIS and TOA.

. \ Gst-launch-1.0.exe -v rtspsrc debug = TRUE location =
rtsph://10.107.14.2/axis-media/media.amp Videocodec = h264 user-id = admin
user-pw = guest latency = 100 ! rtph264depay ! avdec_h264 ! autovideosink

. \ Gst-launch-1.0.exe -v rtspsrc debug = TRUE location =
rtsph://axis-media/media.amp Videocodec = h264 proxy =
http://10.107.14.2:80 proxy-id = root proxy- pw = root latency = 100 !
rtph264depay ! avdec_h264! autovideosink

Thank you for your attention.

"gstreamer-devel" <[hidden email]> wrote on
2016/03/31 12:34:57:

>
> Hi, Nicola

> Thank you for your reply.

> I was able to run the gst-launch in gstreamer ver.1.8.0 of official
> windows binaries.

> And, I was able to receive the "RTP and RTSP over HTTP" in the next
> command.

> .\gst-launch-1.0.exe -v rtspsrc debug=TRUE
>
location=rtsph://root:root@10.107.14.2:80/axis-media/media.amp?videocodec=h264
> latency=100 ! rtph264depay ! avdec_h264 ! autovideosink

> Thank you very much !

>
> "gstreamer-devel" <[hidden email]> wrote
on
> 2016/03/31 00:09:25:

> > > 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.

> > Hi,

> > give a try to the official windows binaries:

> > https://gstreamer.freedesktop.org/data/pkg/windows/1.8.0/

> > Nicola

> > >
> > >
> > > "gstreamer-devel" <[hidden email]>
> 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, [hidden email] 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
> > >>> [hidden email]
> > >>> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
> > >> _______________________________________________
> > >> gstreamer-devel mailing list
> > >> [hidden email]
> > >> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
> > > _______________________________________________
> > > gstreamer-devel mailing list
> > > [hidden email]
> > > https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel

> > _______________________________________________
> > gstreamer-devel mailing list
> > [hidden email]
> > https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
> _______________________________________________
> gstreamer-devel mailing list
> [hidden email]
> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
_______________________________________________
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Sebastian Dröge-3

Re: Failed in RTP and RTSP over HTTP

On Mo, 2016-04-18 at 16:04 +0900, [hidden email] wrote:

> Hi, Nicola
>
> I have gotten the advice previously, I succeeded in the reception of the
> "RTP and RTSP over HTTP" in gstreamer ver.1.8.0.
>
> I used the following command at that time.
>
> .\Gst-launch-1.0.exe -v rtspsrc debug = TRUE location =
> rtsph://root:root@10.107.14.2:80 / axis-media / media.amp videocodec =
> h264 latency = 100 ! rtph264depay ! avdec_h264 ! autovideosink
>
> However, depending on the camera manufacturers, it does not accept the
> user name and password is embedded URL.
> For example, TOA company's Onvif camera "N-C3120" does not accept this
> URL.
>
> Please tell me how to tell a user name and password to the camera in other
> ways that are not URL using the gst-launch.
>
> I tried the following command, but failed both AXIS and TOA.

How are they failing? And what kind of authorization scheme are those
two cameras implementing?

--
Sebastian Dröge, Centricular Ltd · http://www.centricular.com

_______________________________________________
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s_kamiya

Re: Failed in RTP and RTSP over HTTP

Hi, Sebastian

Thank you for reply.

I've described the log. I am sorry in the long reply.
AXIS camera failed to authentication, TOA camera did not respond to the
RTSP (OPTION request).

Please give me advice.

###### AXIS Camera ######

[Command]

.\gst-launch-1.0.exe -v rtspsrc debug=TRUE
location=rtsph://10.107.14.2:80/axis-media/media.amp?videocodec=h264
user-id=admin user-pw=guest latency=100 ! rtph264depay ! avdec_h264 !
autovideosink

[Log]

Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Progress: (open) Opening Stream
Progress: (connect) Connecting to
rtsph://10.107.14.2:80/axis-media/media.amp?videocodec=h264
Progress: (connect) Connecting to
rtsph://10.107.14.2:80/axis-media/media.amp?videocodec=h264
ERROR: from element /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0: Could not
open resource for reading and writing.
Additional debug info:
gstrtspsrc.c(6743): gst_rtspsrc_retieve_sdp():
/GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0:
Failed to connect. (Generic error).
ERROR: pipeline doesn't want to preroll.
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
Setting pipeline to NULL ...
Freeing pipeline ...

[Packet]
First gst-launch was a Get request in the Basic authentication, but has
failed to authenticate.
send1.GET /axis-media/media.amp?videocodec=h264 HTTP/1.0
recv1.401 Unauthorized

Next gst-launch was a Get request in the Digest authentication, but has
failed to authenticate.
send2.GET /axis-media/media.amp?videocodec=h264 HTTP/1.0
recv2.401 Unauthorized

###### TOA Camera ######

[Command]

.\gst-launch-1.0.exe -v rtspsrc debug=TRUE
location=rtsph://10.107.14.82:80/http/unicast/Profile_H264_1 user-id=xxx
user-pw=xxx latency=100 ! rtph264depay ! avdec_h264 ! autovideosink

[Log]

Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Progress: (open) Opening Stream
Progress: (connect) Connecting to
rtsph://10.107.14.82:80/http/unicast/Profile_H264_1
RTSP request message 09E5FD08
request line:
method: 'OPTIONS'
uri: 'rtsp://10.107.14.82:80/http/unicast/Profile_H264_1'
version: '1.0'
headers:
key: 'User-Agent', value: 'GStreamer/1.8.0'
key: 'User-Agent', value: 'RealMedia Player Version 6.0.9.1235
(linux-2.0-libc6-i386-gcc2.95)'
Progress: (open) Retrieving server options
key: 'ClientChallenge', value: '9e26d33f2984236010ef6253fb1887f7'
key: 'CompanyID', value: 'KnKV4M4I/B2FjJ1TToLycw=='
key: 'GUID', value: '00000000-0000-0000-0000-000000000000'
key: 'RegionData', value: '0'
key: 'PlayerStarttime', value: '[28/03/2003:22:50:23 00:00]'
key: 'ClientID', value: 'Linux_2.4_6.0.9.1235_play32_RN01_EN_586'
body:
ERROR: from element /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0: Could not
read from resource.
Additional debug info:
gstrtspsrc.c(5448): gst_rtspsrc_try_send():
/GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0:
Could not receive message. (Timeout while waiting for server response)
ERROR: pipeline doesn't want to preroll.
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
Setting pipeline to NULL ...
Freeing pipeline ...

[Packet]
send1.GET /http/unicast/Profile_H264_1 HTTP/1.0
recv1.200 OK
send1.POST /http/unicast/Profile_H264_1 HTTP/1.0
body → OPTIONS rtsp://10.107.14.82:80/http/unicast/Profile_H264_1
RTSP/1.0 ...
recv2.No response

> > Hi, Nicola
> >
> > I have gotten the advice previously, I succeeded in the reception of
the

> > "RTP and RTSP over HTTP" in gstreamer ver.1.8.0.
> >
> > I used the following command at that time.
> >
> > .\Gst-launch-1.0.exe -v rtspsrc debug = TRUE location =
> > rtsph://root:root@10.107.14.2:80 / axis-media / media.amp videocodec =
> > h264 latency = 100 ! rtph264depay ! avdec_h264 ! autovideosink
> >
> > However, depending on the camera manufacturers, it does not accept the
> > user name and password is embedded URL.
> > For example, TOA company's Onvif camera "N-C3120" does not accept this
> > URL.
> >
> > Please tell me how to tell a user name and password to the camera in

other
> > ways that are not URL using the gst-launch.
> >
> > I tried the following command, but failed both AXIS and TOA.

> How are they failing? And what kind of authorization scheme are those
> two cameras implementing?

> --
> Sebastian Dr?ge, Centricular Ltd ? http://www.centricular.com
> [添付ファイル "signature.asc" は神谷茂治/TOA が削除しました]
> _______________________________________________
> gstreamer-devel mailing list
> [hidden email]
> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
_______________________________________________
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[hidden email]
https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel

s_kamiya

Re: Failed in RTP and RTSP over HTTP

In reply to this post by Sebastian Dröge-3