rtspserver debugging

[Thornton, Keith]

Hi, I’m having difficulty debugging my pipeline using rtspserver.

It is implemented along the lines of the appsrc example. I am pushing packets to the appsrc and am getting status = OK back. I have wireshark running on the server side and I see the RTSP negotiation but no rtp packets go out onto the network. The packets being pushed are h264 byte-stream au packets. The appsrc pipeline is just

“ ( appsrc name=mysrc ! rtph264pay name=pay0 config-interval=10 pt96 )”

I have verified that the pipeline is built and negotiated by generating a .dot file. I am running with GST_DEBUG=rtsp*:6,rtp*:6,udp=6 but I don’t see any error messages in the resulting log.

I am running on windows 64 with gstreamer 1.7 (a 2 month old copy from git master)

Does anyone have any suggestions what the problem might be or how I can get mor useful debugging information?

Regards Keith

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[Venkatesh Adiga]

Not sure you may have some ports are behind the iptables etc.

Sent from Yahoo Mail on Android

On Wed, Mar 9, 2016 at 9:10 PM, Thornton, Keith

<[hidden email]> wrote:

Hi, I’m having difficulty debugging my pipeline using rtspserver.

It is implemented along the lines of the appsrc example. I am pushing packets to the appsrc and am getting status = OK back. I have wireshark running on the server side and I see the RTSP negotiation but no rtp packets go out onto the network. The packets being pushed are h264 byte-stream au packets. The appsrc pipeline is just

“ ( appsrc name=mysrc ! rtph264pay name=pay0 config-interval=10 pt96 )”

I have verified that the pipeline is built and negotiated by generating a .dot file. I am running with GST_DEBUG=rtsp*:6,rtp*:6,udp=6 but I don’t see any error messages in the resulting log.

I am running on windows 64 with gstreamer 1.7 (a 2 month old copy from git master)

Does anyone have any suggestions what the problem might be or how I can get mor useful debugging information?

Regards Keith

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[Thornton, Keith]

No, that’s not the problem, If I replace the appsrc with a videotestsrc the packets are sent out over the network. I suspect it has something to do with timestamps.

 

Von: gstreamer-devel [mailto:[hidden email]] Im Auftrag von Venkatesh Adiga
Gesendet: Mittwoch, 9. März 2016 19:37
An: Discussion of the development of and with GStreamer
Betreff: Re: rtspserver debugging

 

Not sure you may have some ports are behind the iptables etc.

 

Sent from Yahoo Mail on Android

 

On Wed, Mar 9, 2016 at 9:10 PM, Thornton, Keith

<[hidden email]> wrote:

Hi, I’m having difficulty debugging my pipeline using rtspserver.

It is implemented along the lines of the appsrc example. I am pushing packets to the appsrc and am getting status = OK back. I have wireshark running on the server side and I see the RTSP negotiation but no rtp packets go out onto the network. The packets being pushed are h264 byte-stream au packets. The appsrc pipeline is just

“ ( appsrc name=mysrc ! rtph264pay name=pay0 config-interval=10 pt96 )”

I have verified that the pipeline is built and negotiated by generating a .dot file. I am running with GST_DEBUG=rtsp*:6,rtp*:6,udp=6 but I don’t see any error messages in the resulting log.

I am running on windows 64 with gstreamer 1.7 (a 2 month old copy from git master)

Does anyone have any suggestions what the problem might be or how I can get mor useful debugging information?

Regards Keith

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[Sebastian Dröge-3]

On Do, 2016-03-10 at 06:50 +0000, Thornton, Keith wrote:
> No, that’s not the problem, If I replace the appsrc with a
> videotestsrc the packets are sent out over the network. I suspect it
> has something to do with timestamps.

How are you producing timestamps? You must ensure that your timestamps
make the first buffer to start at running time 0. If it would e.g. be
10s, then the sink would wait 10s before sending out the first packet.

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


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[Thornton, Keith]

This is most probably the problem. My pipeline runs, providing a live image on the screen and perhaps one or two recondings in parallel. When a streaming client connects, a new branch is added to a tee dynamically so the running time is not 0

-----Ursprüngliche Nachricht-----
Von: gstreamer-devel [mailto:[hidden email]] Im Auftrag von Sebastian Dröge
Gesendet: Donnerstag, 10. März 2016 10:02
An: Discussion of the development of and with GStreamer; Venkatesh Adiga
Betreff: Re: AW: rtspserver debugging

On Do, 2016-03-10 at 06:50 +0000, Thornton, Keith wrote:
> No, that’s not the problem, If I replace the appsrc with a
> videotestsrc the packets are sent out over the network. I suspect it
> has something to do with timestamps.

How are you producing timestamps? You must ensure that your timestamps make the first buffer to start at running time 0. If it would e.g. be 10s, then the sink would wait 10s before sending out the first packet.

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

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[Sebastian Dröge-3]

On Do, 2016-03-10 at 10:36 +0000, Thornton, Keith wrote:
> This is most probably the problem. My pipeline runs, providing a live
> image on the screen and perhaps one or two recondings in parallel.
> When a streaming client connects, a new branch is added to a tee
> dynamically so the running time is not 0

You need to translate it then :)

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


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