[gst-devel] rtspsrc can't connect to network camera

Kocsis Tibor ikt011 at gmail.com
Wed Dec 8 18:49:32 CET 2010


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 <gibrovacco at gmail.com> wrote:
> Hi,
>
> On Wed, Dec 8, 2010 at 4:08 PM, Kocsis Tibor <ikt011 at gmail.com> 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
>>
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> 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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