rtspsrc can't connect to network camera

[Kocsis Tibor]

Hi,

I have a network camera and i wan't to use rtspsrc to watch live, but
it can't connect to the camera. The last few lines are the following
before I have the "not linked (-1)" error message:

0:00:00.857076440  7086  0x9ab0800 DEBUG                rtspsrc
gstrtspsrc.c:2197:request_pt_map:<rtspsrc0> getting pt map for pt 100
in session 0
0:00:00.857117205  7086  0x9ab0800 DEBUG                rtspsrc
gstrtspsrc.c:2197:request_pt_map:<rtspsrc0> getting pt map for pt 111
in session 0
0:00:00.857424306  7086  0x9ad2508 DEBUG                rtspsrc
gstrtspsrc.c:2129:new_session_pad:<rtspsrc0> got new session pad
<rtpbin0:recv_rtp_src_0_708331394_111>
0:00:00.857531911  7086  0x9ad2508 DEBUG                rtspsrc
gstrtspsrc.c:2137:new_session_pad:<rtspsrc0> stream: 0, SSRC
708331394, PT 111
0:00:00.857818625  7086  0x9ad2508 DEBUG                rtspsrc
gstrtspsrc.c:2161:new_session_pad:<rtspsrc0> stream 0x9aa7b48,
container 0, disabled 0, added 1
0:00:00.857919871  7086  0x9ad2508 DEBUG                rtspsrc
gstrtspsrc.c:2161:new_session_pad:<rtspsrc0> stream 0x9a9b310,
container 0, disabled 0, added 0
WARNING: from element /GstPipeline:pipeline0/GstFakeSink:fakesink0:
Internal data flow problem.
Additional debug info:
gstbasesink.c(3446): gst_base_sink_chain_unlocked ():
/GstPipeline:pipeline0/GstFakeSink:fakesink0:
Received buffer without a new-segment. Assuming timestamps start from 0.
0:00:00.858316430  7086  0x9a9ed78 DEBUG                rtspsrc
gstrtspsrc.c:1991:gst_rtspsrc_handle_src_query:<rtspsrc0> pad
rtspsrc0:recv_rtp_src_0_708331394_111 received query latency
0:00:00.858416389  7086  0x9a9ed78 DEBUG                rtspsrc
gstrtspsrc.c:1875:gst_rtspsrc_handle_src_event:<rtspsrc0> pad
rtspsrc0:recv_rtp_src_0_708331394_111 received event latency
0:00:02.856541066  7086  0x9ad2508 DEBUG                rtspsrc
gstrtspsrc.c:2129:new_session_pad:<rtspsrc0> got new session pad
<rtpbin0:recv_rtp_src_0_708331394_100>
0:00:02.856568323  7086  0x9ad2508 DEBUG                rtspsrc
gstrtspsrc.c:2137:new_session_pad:<rtspsrc0> stream: 0, SSRC
708331394, PT 100
0:00:02.856657855  7086  0x9ad2508 DEBUG                rtspsrc
gstrtspsrc.c:2161:new_session_pad:<rtspsrc0> stream 0x9aa7b48,
container 0, disabled 0, added 1
0:00:02.856669701  7086  0x9ad2508 DEBUG                rtspsrc
gstrtspsrc.c:2161:new_session_pad:<rtspsrc0> stream 0x9a9b310,
container 0, disabled 0, added 0
0:00:02.857156131  7086  0x9ab0800 DEBUG                rtspsrc
gstrtspsrc.c:6060:gst_rtspsrc_handle_message:<rtspsrc0> got error from
udpsrc0
0:00:02.857172483  7086  0x9ab0800 DEBUG                rtspsrc
gstrtspsrc.c:6074:gst_rtspsrc_handle_message:<rtspsrc0> combined
flows: not-linked
ERROR: from element
/GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0/GstUDPSrc:udpsrc0: Internal
data flow error.
Additional debug info:
gstbasesrc.c(2563): gst_base_src_loop ():
/GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0/GstUDPSrc:udpsrc0:
streaming task paused, reason not-linked (-1)
Execution ended after 2704644757 n

VLC and mplayer can work with the camera, only gstreamer fails. It is
possible to have some problem with the "PT 100"? For h264 the payload
is usually 96, but this camera sends the following sdp back:

 medias:
  media 0:
   media:       'video'
   port:        '0'
   num_ports:   '4294967295'
   proto:       'RTP/AVP'
   formats:
    format  '100'
   information: '(NULL)'
   bandwidths:
    type:         'AS'
    bandwidth:    '1800'
   key:
    type:       '(NULL)'
    data:       '(NULL)'
   attributes:
    attribute 'framerate' : '12.5'
    attribute 'quality' : '8'
    attribute 'control' : 'trackID=1'
    attribute 'rtpmap' : '100 H264/90000'
    attribute 'fmtp' : '100 profile-level-id=42e01f; Reso=320:240;
Scanning=0; TVSystem=pal; CameraMode=standard; BitRateMode=vbr;
Gop=30; AspectRatio=4:3; packetization-mode=1;
sprop-parameter-sets=Z0LwFJGwUH7AW4KAgKAAAH0gAAw1EIAAAAAAAAA=,aM44gA=='
  media 1:
   media:       'data'
   port:        '0'
   num_ports:   '4294967295'
   proto:       'RTP/AVP'
   formats:
    format  '111'
   information: '(NULL)'
   key:
    type:       '(NULL)'
    data:       '(NULL)'
   attributes:
    attribute 'rtpmap' : '111 octet-stream/1'
    attribute 'fmtp' : '111 Mac=08007b889ebe; Model=VCC-HD2300P;
TargetBitRate=900; FirmVer=010103; CameraSeries=2;'


Any ideas?

Thanks
Tibor

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[Marco Ballesio]

Hi,

On Wed, Dec 8, 2010 at 4:08 PM, Kocsis Tibor <[hidden email]> wrote:
> Hi,
>
> I have a network camera and i wan't to use rtspsrc to watch live, but
> it can't connect to the camera. The last few lines are the following
> before I have the "not linked (-1)" error message:

it appears some of the elements in the pipeline are not negotiating
proper caps. Maybe you're just missing an ffmpegcolorspace element
between the video decoder and the sink, maybe the pipeline is
completely wrong or maybe you're missing a plugin, but you should get
an explicit message from gst-launch in such a case -if you're using
gst-launch-.

You can understand more about caps negotiation adding the -v
command-line option to gst-launch.

..snip..

>
> VLC and mplayer can work with the camera, only gstreamer fails. It is
> possible to have some problem with the "PT 100"? For h264 the payload

all the dynamic payload types are supported as specified in rfc1890,
table 2. The value "100" does not have any particular meaning wrt
that.

> is usually 96, but this camera sends the following sdp back:
>
>  medias:
>  media 0:
>   media:       'video'
>   port:        '0'
>   num_ports:   '4294967295'
>   proto:       'RTP/AVP'
>   formats:
>    format  '100'
>   information: '(NULL)'
>   bandwidths:
>    type:         'AS'
>    bandwidth:    '1800'
>   key:
>    type:       '(NULL)'
>    data:       '(NULL)'
>   attributes:
>    attribute 'framerate' : '12.5'
>    attribute 'quality' : '8'
>    attribute 'control' : 'trackID=1'
>    attribute 'rtpmap' : '100 H264/90000'
>    attribute 'fmtp' : '100 profile-level-id=42e01f; Reso=320:240;
> Scanning=0; TVSystem=pal; CameraMode=standard; BitRateMode=vbr;
> Gop=30; AspectRatio=4:3; packetization-mode=1;
> sprop-parameter-sets=Z0LwFJGwUH7AW4KAgKAAAH0gAAw1EIAAAAAAAAA=,aM44gA=='
>  media 1:
>   media:       'data'
>   port:        '0'
>   num_ports:   '4294967295'
>   proto:       'RTP/AVP'
>   formats:
>    format  '111'
>   information: '(NULL)'
>   key:
>    type:       '(NULL)'
>    data:       '(NULL)'
>   attributes:
>    attribute 'rtpmap' : '111 octet-stream/1'
>    attribute 'fmtp' : '111 Mac=08007b889ebe; Model=VCC-HD2300P;
> TargetBitRate=900; FirmVer=010103; CameraSeries=2;'

Nothing awfully wrong here imo. Can you post the pipeline you're
using? Have you tried with a simple:

gst-launch playbin2 uri=rtsp://camera_address

?

Regards

>
>
> Any ideas?
>
> Thanks
> Tibor
>
> ------------------------------------------------------------------------------
> What happens now with your Lotus Notes apps - do you make another costly
> upgrade, or settle for being marooned without product support? Time to move
> off Lotus Notes and onto the cloud with Force.com, apps are easier to build,
> use, and manage than apps on traditional platforms. Sign up for the Lotus
> Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d
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>

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[Kocsis Tibor]

Hi,

I used a very simple pipeline:
gst-launch rtspsrc location="..." ! fakesink sync=true

As far as I know this means that the negotiation problem is at one of
the rtspsrc's elements:

0:00:02.857172483  7086  0x9ab0800 DEBUG                rtspsrc
gstrtspsrc.c:6074:gst_rtspsrc_handle_message:<rtspsrc0> combined
flows: not-linked
ERROR: from element
/GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0/GstUDPSrc:udpsrc0: Internal
data flow error.
Additional debug info:
gstbasesrc.c(2563): gst_base_src_loop ():
/GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0/GstUDPSrc:udpsrc0:
streaming task paused, reason not-linked (-1)

I'll make a test tomorrow with playbin2, but I guess the result will
be the same as above.

Regards
Tibor



On Wed, Dec 8, 2010 at 3:52 PM, Marco Ballesio <[hidden email]> wrote:

> Hi,
>
> On Wed, Dec 8, 2010 at 4:08 PM, Kocsis Tibor <[hidden email]> wrote:
>> Hi,
>>
>> I have a network camera and i wan't to use rtspsrc to watch live, but
>> it can't connect to the camera. The last few lines are the following
>> before I have the "not linked (-1)" error message:
>
> it appears some of the elements in the pipeline are not negotiating
> proper caps. Maybe you're just missing an ffmpegcolorspace element
> between the video decoder and the sink, maybe the pipeline is
> completely wrong or maybe you're missing a plugin, but you should get
> an explicit message from gst-launch in such a case -if you're using
> gst-launch-.
>
> You can understand more about caps negotiation adding the -v
> command-line option to gst-launch.
>
> ..snip..
>
>>
>> VLC and mplayer can work with the camera, only gstreamer fails. It is
>> possible to have some problem with the "PT 100"? For h264 the payload
>
> all the dynamic payload types are supported as specified in rfc1890,
> table 2. The value "100" does not have any particular meaning wrt
> that.
>
>> is usually 96, but this camera sends the following sdp back:
>>
>>  medias:
>>  media 0:
>>   media:       'video'
>>   port:        '0'
>>   num_ports:   '4294967295'
>>   proto:       'RTP/AVP'
>>   formats:
>>    format  '100'
>>   information: '(NULL)'
>>   bandwidths:
>>    type:         'AS'
>>    bandwidth:    '1800'
>>   key:
>>    type:       '(NULL)'
>>    data:       '(NULL)'
>>   attributes:
>>    attribute 'framerate' : '12.5'
>>    attribute 'quality' : '8'
>>    attribute 'control' : 'trackID=1'
>>    attribute 'rtpmap' : '100 H264/90000'
>>    attribute 'fmtp' : '100 profile-level-id=42e01f; Reso=320:240;
>> Scanning=0; TVSystem=pal; CameraMode=standard; BitRateMode=vbr;
>> Gop=30; AspectRatio=4:3; packetization-mode=1;
>> sprop-parameter-sets=Z0LwFJGwUH7AW4KAgKAAAH0gAAw1EIAAAAAAAAA=,aM44gA=='
>>  media 1:
>>   media:       'data'
>>   port:        '0'
>>   num_ports:   '4294967295'
>>   proto:       'RTP/AVP'
>>   formats:
>>    format  '111'
>>   information: '(NULL)'
>>   key:
>>    type:       '(NULL)'
>>    data:       '(NULL)'
>>   attributes:
>>    attribute 'rtpmap' : '111 octet-stream/1'
>>    attribute 'fmtp' : '111 Mac=08007b889ebe; Model=VCC-HD2300P;
>> TargetBitRate=900; FirmVer=010103; CameraSeries=2;'
>
> Nothing awfully wrong here imo. Can you post the pipeline you're
> using? Have you tried with a simple:
>
> gst-launch playbin2 uri=rtsp://camera_address
>
> ?
>
> Regards
>
>>
>>
>> Any ideas?
>>
>> Thanks
>> Tibor
>>
>> ------------------------------------------------------------------------------
>> What happens now with your Lotus Notes apps - do you make another costly
>> upgrade, or settle for being marooned without product support? Time to move
>> off Lotus Notes and onto the cloud with Force.com, apps are easier to build,
>> use, and manage than apps on traditional platforms. Sign up for the Lotus
>> Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d
>> _______________________________________________
>> gstreamer-devel mailing list
>> [hidden email]
>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
>>
>
> ------------------------------------------------------------------------------
> What happens now with your Lotus Notes apps - do you make another costly
> upgrade, or settle for being marooned without product support? Time to move
> off Lotus Notes and onto the cloud with Force.com, apps are easier to build,
> use, and manage than apps on traditional platforms. Sign up for the Lotus
> Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d
> _______________________________________________
> gstreamer-devel mailing list
> [hidden email]
> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
>

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[Marco Ballesio]

On 12/8/10, Kocsis Tibor <[hidden email]> wrote:
> Hi,
>
> I used a very simple pipeline:
> gst-launch rtspsrc location="..." ! fakesink sync=true
>
> As far as I know this means that the negotiation problem is at one of
> the rtspsrc's elements:

that may not be true. A caps negotiation issue may trigger an error
from the source element as the one you're seeing.

I suggest you to add the -v option o gst-launch in order to see how
the negotiations occur.

Regards

>
> 0:00:02.857172483  7086  0x9ab0800 DEBUG                rtspsrc
> gstrtspsrc.c:6074:gst_rtspsrc_handle_message:<rtspsrc0> combined
> flows: not-linked
> ERROR: from element
> /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0/GstUDPSrc:udpsrc0: Internal
> data flow error.
> Additional debug info:
> gstbasesrc.c(2563): gst_base_src_loop ():
> /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0/GstUDPSrc:udpsrc0:
> streaming task paused, reason not-linked (-1)
>
> I'll make a test tomorrow with playbin2, but I guess the result will
> be the same as above.
>
> Regards
> Tibor
>
>
>
> On Wed, Dec 8, 2010 at 3:52 PM, Marco Ballesio <[hidden email]> wrote:
>> Hi,
>>
>> On Wed, Dec 8, 2010 at 4:08 PM, Kocsis Tibor <[hidden email]> wrote:
>>> Hi,
>>>
>>> I have a network camera and i wan't to use rtspsrc to watch live, but
>>> it can't connect to the camera. The last few lines are the following
>>> before I have the "not linked (-1)" error message:
>>
>> it appears some of the elements in the pipeline are not negotiating
>> proper caps. Maybe you're just missing an ffmpegcolorspace element
>> between the video decoder and the sink, maybe the pipeline is
>> completely wrong or maybe you're missing a plugin, but you should get
>> an explicit message from gst-launch in such a case -if you're using
>> gst-launch-.
>>
>> You can understand more about caps negotiation adding the -v
>> command-line option to gst-launch.
>>
>> ..snip..
>>
>>>
>>> VLC and mplayer can work with the camera, only gstreamer fails. It is
>>> possible to have some problem with the "PT 100"? For h264 the payload
>>
>> all the dynamic payload types are supported as specified in rfc1890,
>> table 2. The value "100" does not have any particular meaning wrt
>> that.
>>
>>> is usually 96, but this camera sends the following sdp back:
>>>
>>>  medias:
>>>  media 0:
>>>   media:       'video'
>>>   port:        '0'
>>>   num_ports:   '4294967295'
>>>   proto:       'RTP/AVP'
>>>   formats:
>>>    format  '100'
>>>   information: '(NULL)'
>>>   bandwidths:
>>>    type:         'AS'
>>>    bandwidth:    '1800'
>>>   key:
>>>    type:       '(NULL)'
>>>    data:       '(NULL)'
>>>   attributes:
>>>    attribute 'framerate' : '12.5'
>>>    attribute 'quality' : '8'
>>>    attribute 'control' : 'trackID=1'
>>>    attribute 'rtpmap' : '100 H264/90000'
>>>    attribute 'fmtp' : '100 profile-level-id=42e01f; Reso=320:240;
>>> Scanning=0; TVSystem=pal; CameraMode=standard; BitRateMode=vbr;
>>> Gop=30; AspectRatio=4:3; packetization-mode=1;
>>> sprop-parameter-sets=Z0LwFJGwUH7AW4KAgKAAAH0gAAw1EIAAAAAAAAA=,aM44gA=='
>>>  media 1:
>>>   media:       'data'
>>>   port:        '0'
>>>   num_ports:   '4294967295'
>>>   proto:       'RTP/AVP'
>>>   formats:
>>>    format  '111'
>>>   information: '(NULL)'
>>>   key:
>>>    type:       '(NULL)'
>>>    data:       '(NULL)'
>>>   attributes:
>>>    attribute 'rtpmap' : '111 octet-stream/1'
>>>    attribute 'fmtp' : '111 Mac=08007b889ebe; Model=VCC-HD2300P;
>>> TargetBitRate=900; FirmVer=010103; CameraSeries=2;'
>>
>> Nothing awfully wrong here imo. Can you post the pipeline you're
>> using? Have you tried with a simple:
>>
>> gst-launch playbin2 uri=rtsp://camera_address
>>
>> ?
>>
>> Regards
>>
>>>
>>>
>>> Any ideas?
>>>
>>> Thanks
>>> Tibor
>>>
>>> ------------------------------------------------------------------------------
>>> What happens now with your Lotus Notes apps - do you make another costly
>>> upgrade, or settle for being marooned without product support? Time to
>>> move
>>> off Lotus Notes and onto the cloud with Force.com, apps are easier to
>>> build,
>>> use, and manage than apps on traditional platforms. Sign up for the Lotus
>>> Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d
>>> _______________________________________________
>>> gstreamer-devel mailing list
>>> [hidden email]
>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
>>>
>>
>> ------------------------------------------------------------------------------
>> What happens now with your Lotus Notes apps - do you make another costly
>> upgrade, or settle for being marooned without product support? Time to
>> move
>> off Lotus Notes and onto the cloud with Force.com, apps are easier to
>> build,
>> use, and manage than apps on traditional platforms. Sign up for the Lotus
>> Notes Migration Kit to learn more. http://p.sf.net/sfu/salesforce-d2d
>> _______________________________________________
>> gstreamer-devel mailing list
>> [hidden email]
>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
>>
>
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