create simple pipeline receive udpsrc rtp stream and send rtsp

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create simple pipeline receive udpsrc rtp stream and send rtsp

morrisford
In my search to create this pipeline it would seem to me that I could connect the depay and pay elements together such as:

./test-launch "( udpsrc port=5000 caps=\"application/x-rtp, media=\(string\)video, clock-rate=\(int\)90000, encoding-name=\(string\)H264,  payload=\(int\)96, ssrc=\(guint\)2396357661, clock-base=\(guint\)2297066863, seqnum-base=\(guint\)49439\" ! rtph264depay ! rtph264pay name=pay0 pt=96 )"

This doesn't work but it seems like it should. What do I need between the depay and pay to make it fly? Is there an even simpler pipe that would achieve the desired results?

Morris

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Re: create simple pipeline receive udpsrc rtp stream and send rtsp

Mailing List SVR


Il giorno lun, 28/02/2011 alle 15.43 -0500, Morris Ford ha scritto:
> In my search to create this pipeline it would seem to me that I could
> connect the depay and pay elements together such as:
>
>
> ./test-launch "( udpsrc port=5000 caps=\"application/x-rtp, media=
> \(string\)video, clock-rate=\(int\)90000, encoding-name=\(string
> \)H264,  payload=\(int\)96, ssrc=\(guint\)2396357661, clock-base=
> \(guint\)2297066863, seqnum-base=\(guint\)49439\" ! rtph264depay !
> rtph264pay name=pay0 pt=96 )"

this should work, please post the debug messages (export
GST_DEBUG="*:5")


>
>
> This doesn't work but it seems like it should. What do I need between
> the depay and pay to make it fly? Is there an even simpler pipe that
> would achieve the desired results?
>
>
> Morris
> _______________________________________________
> gstreamer-devel mailing list
> [hidden email]
> http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel

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Re: create simple pipeline receive udpsrc rtp stream and send rtsp

Marco Ballesio
In reply to this post by morrisford
Hi,

On Mon, Feb 28, 2011 at 10:43 PM, Morris Ford <[hidden email]> wrote:
> In my search to create this pipeline it would seem to me that I could
> connect the depay and pay elements together such as:
> ./test-launch "( udpsrc port=5000 caps=\"application/x-rtp,
> media=\(string\)video, clock-rate=\(int\)90000,
> encoding-name=\(string\)H264,  payload=\(int\)96, ssrc=\(guint\)2396357661,
> clock-base=\(guint\)2297066863, seqnum-base=\(guint\)49439\" ! rtph264depay
> ! rtph264pay name=pay0 pt=96 )"

looks like you're missing sink elements. Depending on what you want
you mean for "making it fly" it might already be flying or not ;).

Regards

> This doesn't work but it seems like it should. What do I need between the
> depay and pay to make it fly? Is there an even simpler pipe that would
> achieve the desired results?
> Morris
> _______________________________________________
> gstreamer-devel mailing list
> [hidden email]
> http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>
>
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Re: create simple pipeline receive udpsrc rtp stream and send rtsp

morrisford
I did gst-inspect on the pay and depay elements and the depay sources exactly the same thing that the pay sinks so I assumed that the two should talk. My understanding is that the gst-rtsp-server code in test-launch.c has the rest of what is needed to make the pipeline work.
Morris

On Fri, Mar 4, 2011 at 3:02 AM, Marco Ballesio <[hidden email]> wrote:
Hi,

On Mon, Feb 28, 2011 at 10:43 PM, Morris Ford <[hidden email]> wrote:
> In my search to create this pipeline it would seem to me that I could
> connect the depay and pay elements together such as:
> ./test-launch "( udpsrc port=5000 caps=\"application/x-rtp,
> media=\(string\)video, clock-rate=\(int\)90000,
> encoding-name=\(string\)H264,  payload=\(int\)96, ssrc=\(guint\)2396357661,
> clock-base=\(guint\)2297066863, seqnum-base=\(guint\)49439\" ! rtph264depay
> ! rtph264pay name=pay0 pt=96 )"

looks like you're missing sink elements. Depending on what you want
you mean for "making it fly" it might already be flying or not ;).

Regards

> This doesn't work but it seems like it should. What do I need between the
> depay and pay to make it fly? Is there an even simpler pipe that would
> achieve the desired results?
> Morris
> _______________________________________________
> gstreamer-devel mailing list
> [hidden email]
> http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>
>
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Re: create simple pipeline receive udpsrc rtp stream and send rtsp

Marco Ballesio
Very late (and short) reply. I'm sorry I didn't remember that you were
using the rtsp server. Could you please post the debug log for the
failing cases?

Regards

On Fri, Mar 4, 2011 at 2:40 PM, Morris Ford <[hidden email]> wrote:

> I did gst-inspect on the pay and depay elements and the depay sources
> exactly the same thing that the pay sinks so I assumed that the two should
> talk. My understanding is that the gst-rtsp-server code in test-launch.c has
> the rest of what is needed to make the pipeline work.
> Morris
>
> On Fri, Mar 4, 2011 at 3:02 AM, Marco Ballesio <[hidden email]> wrote:
>>
>> Hi,
>>
>> On Mon, Feb 28, 2011 at 10:43 PM, Morris Ford <[hidden email]>
>> wrote:
>> > In my search to create this pipeline it would seem to me that I could
>> > connect the depay and pay elements together such as:
>> > ./test-launch "( udpsrc port=5000 caps=\"application/x-rtp,
>> > media=\(string\)video, clock-rate=\(int\)90000,
>> > encoding-name=\(string\)H264,  payload=\(int\)96,
>> > ssrc=\(guint\)2396357661,
>> > clock-base=\(guint\)2297066863, seqnum-base=\(guint\)49439\" !
>> > rtph264depay
>> > ! rtph264pay name=pay0 pt=96 )"
>>
>> looks like you're missing sink elements. Depending on what you want
>> you mean for "making it fly" it might already be flying or not ;).
>>
>> Regards
>>
>> > This doesn't work but it seems like it should. What do I need between
>> > the
>> > depay and pay to make it fly? Is there an even simpler pipe that would
>> > achieve the desired results?
>> > Morris
>> > _______________________________________________
>> > gstreamer-devel mailing list
>> > [hidden email]
>> > http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>> >
>> >
>> _______________________________________________
>> gstreamer-devel mailing list
>> [hidden email]
>> http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>
>
> _______________________________________________
> gstreamer-devel mailing list
> [hidden email]
> http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>
>
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Re: create simple pipeline receive udpsrc rtp stream and send rtsp

morrisford
Here is the debug output:

morrisford@ubuntu:~/gst-rtsp-server/gst-rtsp-0.10.7/examples$ ./test-launch --gst-debug=2 "( udpsrc port=5000 caps=\"application/x-rtp, media=\(string\)video, clock-rate=\(int\)90000, encoding-name=\(string\)H264,  payload=\(int\)96, ssrc=\(guint\)2396357661\" ! rtph264depay ! rtph264pay name=pay0 pt=96 )"
0:00:11.641343170  5193  0x8667008 WARN               rtspmedia rtsp-media.c:806:alloc_udp_ports: multiudpsink version found without send-duplicates property
0:00:11.645912928  5193  0x8667008 WARN                     bin gstbin.c:2330:gst_bin_do_latency_func:<media-pipeline> failed to query latency

** (lt-test-launch:5193): WARNING **: ignoring stream 0 without media type


On Thu, Mar 10, 2011 at 1:28 AM, Marco Ballesio <[hidden email]> wrote:
Very late (and short) reply. I'm sorry I didn't remember that you were
using the rtsp server. Could you please post the debug log for the
failing cases?

Regards

On Fri, Mar 4, 2011 at 2:40 PM, Morris Ford <[hidden email]> wrote:
> I did gst-inspect on the pay and depay elements and the depay sources
> exactly the same thing that the pay sinks so I assumed that the two should
> talk. My understanding is that the gst-rtsp-server code in test-launch.c has
> the rest of what is needed to make the pipeline work.
> Morris
>
> On Fri, Mar 4, 2011 at 3:02 AM, Marco Ballesio <[hidden email]> wrote:
>>
>> Hi,
>>
>> On Mon, Feb 28, 2011 at 10:43 PM, Morris Ford <[hidden email]>
>> wrote:
>> > In my search to create this pipeline it would seem to me that I could
>> > connect the depay and pay elements together such as:
>> > ./test-launch "( udpsrc port=5000 caps=\"application/x-rtp,
>> > media=\(string\)video, clock-rate=\(int\)90000,
>> > encoding-name=\(string\)H264,  payload=\(int\)96,
>> > ssrc=\(guint\)2396357661,
>> > clock-base=\(guint\)2297066863, seqnum-base=\(guint\)49439\" !
>> > rtph264depay
>> > ! rtph264pay name=pay0 pt=96 )"
>>
>> looks like you're missing sink elements. Depending on what you want
>> you mean for "making it fly" it might already be flying or not ;).
>>
>> Regards
>>
>> > This doesn't work but it seems like it should. What do I need between
>> > the
>> > depay and pay to make it fly? Is there an even simpler pipe that would
>> > achieve the desired results?
>> > Morris
>> > _______________________________________________
>> > gstreamer-devel mailing list
>> > [hidden email]
>> > http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>> >
>> >
>> _______________________________________________
>> gstreamer-devel mailing list
>> [hidden email]
>> http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>
>
> _______________________________________________
> gstreamer-devel mailing list
> [hidden email]
> http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>
>
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Re: create simple pipeline receive udpsrc rtp stream and send rtsp

amartin
Dear all,

I have the same problem.

It is not possible to publish an RTSP stream from an udpsrc without transcoding/reencoding the rtp source stream?

Best Regards,

Angel


2011/3/10 Morris Ford <[hidden email]>
Here is the debug output:

morrisford@ubuntu:~/gst-rtsp-server/gst-rtsp-0.10.7/examples$ ./test-launch --gst-debug=2 "( udpsrc port=5000 caps=\"application/x-rtp, media=\(string\)video, clock-rate=\(int\)90000, encoding-name=\(string\)H264,  payload=\(int\)96, ssrc=\(guint\)2396357661\" ! rtph264depay ! rtph264pay name=pay0 pt=96 )"
0:00:11.641343170  5193  0x8667008 WARN               rtspmedia rtsp-media.c:806:alloc_udp_ports: multiudpsink version found without send-duplicates property
0:00:11.645912928  5193  0x8667008 WARN                     bin gstbin.c:2330:gst_bin_do_latency_func:<media-pipeline> failed to query latency

** (lt-test-launch:5193): WARNING **: ignoring stream 0 without media type


On Thu, Mar 10, 2011 at 1:28 AM, Marco Ballesio <[hidden email]> wrote:
Very late (and short) reply. I'm sorry I didn't remember that you were
using the rtsp server. Could you please post the debug log for the
failing cases?

Regards

On Fri, Mar 4, 2011 at 2:40 PM, Morris Ford <[hidden email]> wrote:
> I did gst-inspect on the pay and depay elements and the depay sources
> exactly the same thing that the pay sinks so I assumed that the two should
> talk. My understanding is that the gst-rtsp-server code in test-launch.c has
> the rest of what is needed to make the pipeline work.
> Morris
>
> On Fri, Mar 4, 2011 at 3:02 AM, Marco Ballesio <[hidden email]> wrote:
>>
>> Hi,
>>
>> On Mon, Feb 28, 2011 at 10:43 PM, Morris Ford <[hidden email]>
>> wrote:
>> > In my search to create this pipeline it would seem to me that I could
>> > connect the depay and pay elements together such as:
>> > ./test-launch "( udpsrc port=5000 caps=\"application/x-rtp,
>> > media=\(string\)video, clock-rate=\(int\)90000,
>> > encoding-name=\(string\)H264,  payload=\(int\)96,
>> > ssrc=\(guint\)2396357661,
>> > clock-base=\(guint\)2297066863, seqnum-base=\(guint\)49439\" !
>> > rtph264depay
>> > ! rtph264pay name=pay0 pt=96 )"
>>
>> looks like you're missing sink elements. Depending on what you want
>> you mean for "making it fly" it might already be flying or not ;).
>>
>> Regards
>>
>> > This doesn't work but it seems like it should. What do I need between
>> > the
>> > depay and pay to make it fly? Is there an even simpler pipe that would
>> > achieve the desired results?
>> > Morris
>> > _______________________________________________
>> > gstreamer-devel mailing list
>> > [hidden email]
>> > http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>> >
>> >
>> _______________________________________________
>> gstreamer-devel mailing list
>> [hidden email]
>> http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>
>
> _______________________________________________
> gstreamer-devel mailing list
> [hidden email]
> http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>
>
_______________________________________________
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--
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Investigador / Researcher
Televisión Digital y Servicios Multimedia / Digital TV & Multimedia Services

Vicomtech - Visual Interaction Communication Technologies
Mikeletegi Pasealekua, 57 - Parque Tecnológico
20009 Donostia - San Sebastián - Spain
Tel: +[34] 943 30 92 30
Fax: +[34] 943 30 93 93
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Re: create simple pipeline receive udpsrc rtp stream and send rtsp

mattias-12
Hombre, Angel!
K tal?

Un abrazo,

Mattias

Sent from my iPhone

On May 13, 2011, at 13:53, "Angel Martin" <[hidden email]> wrote:

Dear all,

I have the same problem.

It is not possible to publish an RTSP stream from an udpsrc without transcoding/reencoding the rtp source stream?

Best Regards,

Angel


2011/3/10 Morris Ford <[hidden email][hidden email]>
Here is the debug output:

morrisford@ubuntu:~/gst-rtsp-server/gst-rtsp-0.10.7/examples$ ./test-launch --gst-debug=2 "( udpsrc port=5000 caps=\"application/x-rtp, media=\(string\)video, clock-rate=\(int\)90000, encoding-name=\(string\)H264,  payload=\(int\)96, ssrc=\(guint\)2396357661\" ! rtph264depay ! rtph264pay name=pay0 pt=96 )"
0:00:11.641343170  5193  0x8667008 WARN               rtspmedia rtsp-media.c:806:alloc_udp_ports: multiudpsink version found without send-duplicates property
0:00:11.645912928  5193  0x8667008 WARN                     bin gstbin.c:2330:gst_bin_do_latency_func:<media-pipeline> failed to query latency

** (lt-test-launch:5193): WARNING **: ignoring stream 0 without media type


On Thu, Mar 10, 2011 at 1:28 AM, Marco Ballesio <[hidden email][hidden email]> wrote:
Very late (and short) reply. I'm sorry I didn't remember that you were
using the rtsp server. Could you please post the debug log for the
failing cases?

Regards

On Fri, Mar 4, 2011 at 2:40 PM, Morris Ford <[hidden email][hidden email]> wrote:
> I did gst-inspect on the pay and depay elements and the depay sources
> exactly the same thing that the pay sinks so I assumed that the two should
> talk. My understanding is that the gst-rtsp-server code in test-launch.c has
> the rest of what is needed to make the pipeline work.
> Morris
>
> On Fri, Mar 4, 2011 at 3:02 AM, Marco Ballesio <[hidden email][hidden email]> wrote:
>>
>> Hi,
>>
>> On Mon, Feb 28, 2011 at 10:43 PM, Morris Ford <[hidden email][hidden email]>
>> wrote:
>> > In my search to create this pipeline it would seem to me that I could
>> > connect the depay and pay elements together such as:
>> > ./test-launch "( udpsrc port=5000 caps=\"application/x-rtp,
>> > media=\(string\)video, clock-rate=\(int\)90000,
>> > encoding-name=\(string\)H264,  payload=\(int\)96,
>> > ssrc=\(guint\)2396357661,
>> > clock-base=\(guint\)2297066863, seqnum-base=\(guint\)49439\" !
>> > rtph264depay
>> > ! rtph264pay name=pay0 pt=96 )"
>>
>> looks like you're missing sink elements. Depending on what you want
>> you mean for "making it fly" it might already be flying or not ;).
>>
>> Regards
>>
>> > This doesn't work but it seems like it should. What do I need between
>> > the
>> > depay and pay to make it fly? Is there an even simpler pipe that would
>> > achieve the desired results?
>> > Morris
>> > _______________________________________________
>> > gstreamer-devel mailing list
>> > [hidden email][hidden email]
>> > http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>> >
>> >
>> _______________________________________________
>> gstreamer-devel mailing list
>> [hidden email][hidden email]
>> http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>
>
> _______________________________________________
> gstreamer-devel mailing list
> [hidden email][hidden email]
> http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>
>
_______________________________________________
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--
Ángel Martín Navas
Investigador / Researcher
Televisión Digital y Servicios Multimedia / Digital TV & Multimedia Services

Vicomtech - Visual Interaction Communication Technologies
Mikeletegi Pasealekua, 57 - Parque Tecnológico
20009 Donostia - San Sebastián - Spain
Tel: +[34] 943 30 92 30
Fax: +[34] 943 30 93 93
e-mail: [hidden email][hidden email]
www.vicomtech.org

*** member of IK4 Research Alliance ****
www.ik4.es
*** member of GraphicsMedia.net ****
www.graphicsmedia.net



-----------------------------------------------------
Vicomtech is an ISO 9001:2000 certified institute
-----------------------------------------------------

Este mensaje se dirige exclusivamente a su destinatario.
La información incluida en el presente correo es confidencial sometida a
secreto profesional, especialmente en lo que respecta a los datos de
carácter personal, cuya divulgación está prohibida, en virtud de la
legislación vigente. Si usted no es el destinatario legítimo y lo ha
recibido por error o tiene conocimiento del mismo por cualquier motivo,
le rogamos que nos lo comunique por este medio y proceda a destruirlo o
borrarlo. En todo caso abstengase de utilizar, reproducir, alterar,
archivar o comunicar a terceros el presente mensaje así como los ficheros
anexos, todo ello bajo pena de incurrir en responsabilidades legales.
Cualquier opinión contenida en este correo es exclusiva de su autor y no
representa necesariamente la opinión de ASOCIACIÓN CENTRO DE TECNOLOGÍAS
DE INTERACCIÓN VISUAL Y COMUNICACIONES VICOMTECH (en adelante Vicomtech-IK4)
El emisor no garantiza la integridad, rapidez o seguridad del presente correo,
ni se responsabiliza de posibles perjuicios derivados de la captura,
incorporaciones de virus o cualesquiera otras manipulaciones efectuadas por terceros.
Con motivo de la entrada en vigor de la Ley 34/2002, de 11 de julio, de
Servicios de la Sociedad de la Información y de Comercio Electrónico, le
informamos que pueden revocar en cualquier momento, de forma sencilla y gratuita,
el consentimiento para la recepción de mensajes de vicomtech.org en [hidden email][hidden email].

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Re: create simple pipeline receive udpsrc rtp stream and send rtsp

Felipe Mogollon
Hi to all.

We have solved this issue modifying gstreamer rtsp server:

We have a rtsp server which streams video that is received from rtp
and saves it simultaneously to a temp file:

           gst_rtsp_media_factory_set_launch (factory,     "gstrtpbin
name=rtpbin udpsrc port=5000 !
application/x-rtp,media=video,clock-rate=90000,encoding-name=H264 !
rtpbin.recv_rtp_sink_0   rtpbin. ! rtph264depay byte-stream=true
name=videodepay ! queue ! tee name=videotee ! queue ! rtph264pay
name=pay0 pt=96  udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0
rtpbin.send_rtcp_src_0 ! udpsink port=5005 host=127.0.0.1 sync=false
async=false udpsrc name=audioudpsrc port=5002 ! application/x-rtp,
media=(string)audio, clock-rate=(int)22050,
encoding-name=(string)MPEG4-GENERIC, encoding-params=(string)1,
streamtype=(string)5, profile-level-id=(string)2,
mode=(string)AAC-hbr, config=(string)1388, sizelength=(string)13,
indexlength=(string)3, indexdeltalength=(string)3,
ssrc=(uint)3831044409, payload=(int)96, clock-base=(uint)3295317201,
seqnum-base=(uint)50289 ! rtpbin.recv_rtp_sink_1 rtpbin. !
rtpmp4gdepay ! queue ! tee name=audiotee ! queue ! rtpmp4gpay
name=pay1 pt=97 audiotee. ! decodebin ! lame ! flvmux name=muxer !
filesink location=/tmp/video.flv videotee. ! queue ! muxer. ");

We didn't get the caps when we made  the  rtph264depay  ! rtph264pay
name=pay0 pt=96

We solved it hardcoding the  rtsp-sdp.c file with our rtp received caps:

I mean:
    if(stream->caps==NULL)
    {
        g_warning("NO CAPS");

        stream->caps=gst_caps_from_string("application/x-rtp,
media=(string)video, clock-rate=(int)90000,
encoding-name=(string)H264, ssrc=(uint)3142824526, payload=(int)96,
clock-base=(uint)1706805505, seqnum-base=(uint)54578");

}

Hope it helps.

Regards




On Fri, May 13, 2011 at 2:04 PM, MATTIAS BARTHEL <[hidden email]> wrote:

> Hombre, Angel!
> K tal?
> Un abrazo,
> Mattias
>
> Sent from my iPhone
> On May 13, 2011, at 13:53, "Angel Martin" <[hidden email]> wrote:
>
> Dear all,
>
> I have the same problem.
>
> It is not possible to publish an RTSP stream from an udpsrc without
> transcoding/reencoding the rtp source stream?
>
> Best Regards,
>
> Angel
>
>
> 2011/3/10 Morris Ford <[hidden email]>
>>
>> Here is the debug output:
>> morrisford@ubuntu:~/gst-rtsp-server/gst-rtsp-0.10.7/examples$
>> ./test-launch --gst-debug=2 "( udpsrc port=5000 caps=\"application/x-rtp,
>> media=\(string\)video, clock-rate=\(int\)90000,
>> encoding-name=\(string\)H264,  payload=\(int\)96, ssrc=\(guint\)2396357661\"
>> ! rtph264depay ! rtph264pay name=pay0 pt=96 )"
>> 0:00:11.641343170  5193  0x8667008 WARN               rtspmedia
>> rtsp-media.c:806:alloc_udp_ports: multiudpsink version found without
>> send-duplicates property
>> 0:00:11.645912928  5193  0x8667008 WARN                     bin
>> gstbin.c:2330:gst_bin_do_latency_func:<media-pipeline> failed to query
>> latency
>> ** (lt-test-launch:5193): WARNING **: ignoring stream 0 without media type
>>
>> On Thu, Mar 10, 2011 at 1:28 AM, Marco Ballesio <[hidden email]>
>> wrote:
>>>
>>> Very late (and short) reply. I'm sorry I didn't remember that you were
>>> using the rtsp server. Could you please post the debug log for the
>>> failing cases?
>>>
>>> Regards
>>>
>>> On Fri, Mar 4, 2011 at 2:40 PM, Morris Ford <[hidden email]>
>>> wrote:
>>> > I did gst-inspect on the pay and depay elements and the depay sources
>>> > exactly the same thing that the pay sinks so I assumed that the two
>>> > should
>>> > talk. My understanding is that the gst-rtsp-server code in
>>> > test-launch.c has
>>> > the rest of what is needed to make the pipeline work.
>>> > Morris
>>> >
>>> > On Fri, Mar 4, 2011 at 3:02 AM, Marco Ballesio <[hidden email]>
>>> > wrote:
>>> >>
>>> >> Hi,
>>> >>
>>> >> On Mon, Feb 28, 2011 at 10:43 PM, Morris Ford <[hidden email]>
>>> >> wrote:
>>> >> > In my search to create this pipeline it would seem to me that I
>>> >> > could
>>> >> > connect the depay and pay elements together such as:
>>> >> > ./test-launch "( udpsrc port=5000 caps=\"application/x-rtp,
>>> >> > media=\(string\)video, clock-rate=\(int\)90000,
>>> >> > encoding-name=\(string\)H264,  payload=\(int\)96,
>>> >> > ssrc=\(guint\)2396357661,
>>> >> > clock-base=\(guint\)2297066863, seqnum-base=\(guint\)49439\" !
>>> >> > rtph264depay
>>> >> > ! rtph264pay name=pay0 pt=96 )"
>>> >>
>>> >> looks like you're missing sink elements. Depending on what you want
>>> >> you mean for "making it fly" it might already be flying or not ;).
>>> >>
>>> >> Regards
>>> >>
>>> >> > This doesn't work but it seems like it should. What do I need
>>> >> > between
>>> >> > the
>>> >> > depay and pay to make it fly? Is there an even simpler pipe that
>>> >> > would
>>> >> > achieve the desired results?
>>> >> > Morris
>>> >> > _______________________________________________
>>> >> > gstreamer-devel mailing list
>>> >> > [hidden email]
>>> >> > http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>>> >> >
>>> >> >
>>> >> _______________________________________________
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>
>
> --
>
> Ángel Martín Navas
> Investigador / Researcher
> Televisión Digital y Servicios Multimedia / Digital TV & Multimedia Services
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