specifying the RTP packetization with an RTSP server

[Chuck Crisler]

There are three packetization modes for H264/RTP.

0. SIngle NAL
1. Non-interleaved
2. Interleaved.

I have 1 video source that is supplying RTP input with packetization mode
1. None of the others specify the packetization mode. My RTSP clients all
fail to decode/display the RTP from this source, but succeed with other
sources. My RTSP server has dynamic sources.

I don't know the details of what the packetization mode means and effects.
I have read about it in RFC 3984 and other places. I suspect that it
affects the actual macroblock encoding, hence the decoding, and cannot be
determined from the bitstream. That would cause my clients to fail since my
RTSP SDP isn't specifying the packetization and the default is 0 (single
NAL).

The packetization mode can be specified in an SDP. Since the GStreamer RTSP
server uses a supplied cap filter with an sprop-parameter-set, it seems
like you should be able to also specify the packetization mode. GStreamer
0.10.7 ignores it. :-(

I don't know if this is the source of my problem. To be honest, I really
don't know the details about what I am talking about. I am thinking of
modifying the GStreamer RTSP server code to hardcode the packetization mode
in the SDP to see if that allows clients to decode that video stream.

But since it seems that the GStreamer RTSP server 0.10.7 doesn't support
this option (that is based on my testing), the question is no longer
relevant.


On Fri, Jan 17, 2014 at 12:12 PM, Andrey Utkin <
andrey.krieger.utkin at gmail.com> wrote:

> 2014/1/17 Chuck Crisler <ccrisler at mutualink.net>:
> > How do you specify the packetization mode with an RTSP server?
>
> What exactly do you mean?
>
> --
> Andrey Utkin
> _______________________________________________
> gstreamer-devel mailing list
> gstreamer-devel at lists.freedesktop.org
> http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>
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