gst-rtsp-server: rebroadcasting existing RTP streams

[Martin Jackson]

Dear gstreamer developers,

I'm trying to use the gst-rtsp-server to slurp in existing RTP streams in various formats and make these available to external clients with the RTSP protocol.

However, the general design of the gst-rtsp-server seems to be that you have some media streams e.g. h264 + aac, for which you create payloader elements, and the server does all the rest for you. This is great, unless you just want to route some existing RTP streams, in which case it gets in the way a bit! I'm really impressed with the rtsp protocol handling and RTP infrastructure of gstreamer, so I think there's a lot of gain to be had here from gst-rtsp-server.

Does anyone have any advice on how to do such a thing? The best I have come up with so far is to use identity elements as the payloaders, which is obviously less than ideal!

Best regards, many thanks,

Martin Jackson

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[Thornton, Keith]

Hi, I do something similar by routing my already existing rtp stream via appsink to an rtsp_server.

The server is created using gst_rtsp_media_factory_launch and looks like this

Appsrc name=MySrc min-latency=2000000000 is-live=true ! queue ! rtphpay name=pay0 config-interval= 10 pt=96

I create the rtsp server and get a handle to appsrc using element=gst_rtsp_media_get_element(media) followed by gst_bin_get_by_name_recurse_up(element, MySrc);

There are probably better ways but this one works for me.

 

Von: gstreamer-devel [mailto:[hidden email]] Im Auftrag von Martin Jackson
Gesendet: Freitag, 30. Juni 2017 09:51
An: Discussion of the development of and with GStreamer <[hidden email]>
Betreff: gst-rtsp-server: rebroadcasting existing RTP streams

 

Dear gstreamer developers,

 

I'm trying to use the gst-rtsp-server to slurp in existing RTP streams in various formats and make these available to external clients with the RTSP protocol.

 

However, the general design of the gst-rtsp-server seems to be that you have some media streams e.g. h264 + aac, for which you create payloader elements, and the server does all the rest for you. This is great, unless you just want to route some existing RTP streams, in which case it gets in the way a bit! I'm really impressed with the rtsp protocol handling and RTP infrastructure of gstreamer, so I think there's a lot of gain to be had here from gst-rtsp-server.

 

Does anyone have any advice on how to do such a thing? The best I have come up with so far is to use identity elements as the payloaders, which is obviously less than ideal!

 

Best regards, many thanks,

 

Martin Jackson

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gstreamer-devel mailing list
[hidden email]
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[Martin Jackson]

Hi Keith,

Many thanks for your reply.

On Fri, Jun 30, 2017 at 10:37 AM Thornton, Keith <[hidden email]> wrote:

The server is created using gst_rtsp_media_factory_launch and looks like this

Appsrc name=MySrc min-latency=2000000000 is-live=true ! queue ! rtphpay name=pay0 config-interval= 10 pt=96

I don't quite understand this; what is the rtphpay element? I can track this element down neither in google nor gst-inspect ... do you mean rtph264pay?

Best regards,

Martin Jackson 

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