[gst-devel] RTP Caps,base payloader and jrtplib

[Thomas Vander Stichele]

On Wed, 2005-11-02 at 14:27 +0200, Kai Vehmanen wrote:
> On Wed, 2 Nov 2005, Thomas Vander Stichele wrote:
> 
> >> gst-plugins-good/gst/rtp/README and I see ssrc, timestamp start and
> >> sequence number start all defined in the caps. They are ssrc,
> > not everyone uses rtpbin - it should be possible to retrieve the random
> > values chosen by the payload encoder so the application using it can
> > create the necessary session descriptions.
> 
> Do you really need these values (ssrc, timestamp, sequence), in session 
> descriptions? These are by nature random, and should be handled within the 
> RTP stack.
> 
> Hmm, ssrc might be useful in some cases, but that can potentially
> change at any time (due to collisions), so the it would be best that RTP 
> stack would just signal about the ssrc to upper layers.

Take for example an RTSP server.  The transport header in the RTSP reply
to a SETUP command should list the ssrc that is going to be used for
this RTP stream.  The RTP-Info header in the RTSP reply to a PLAY
command should list seq and rtptime at which the stream for that client
starts.  The RTP spec suggests that rtptime and seqnum start at a
randomized value.

Thomas


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