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--></style><title>Re: [Live-devel] RTSP server/client
application</title></head><body>
<blockquote type="cite" cite><font face="Arial" size="-1">Thanks for
the information. If I got it correctly for the server side the new
"RTPSink" subclass handless the interface to the livemedia
library and to the dedicated hardware interface for controlling the
hardware RTP packetization (payloadtype,
...). "OnDemandServerMediaSubsession" control's
the streaming (init, start, stop, ...). It is not clear how it
works with the SDP parameters that are needed during RSTP init if the
data stream isn't available at all on the
host.</font></blockquote>
<div><br></div>
<div>Only some of the SDP lines (namely: "a=fmtp:") require
reading data from the stream, and - in any case - these lines are
often the same for every stream (if the codec stays the same), so you
may be able to figure this data out in advance, and reuse it for each
stream, rather than actually reading it from the stream each
time.</div>
<div><br></div>
<blockquote type="cite" cite><font face="Arial" size="-1"> The
hardware is controlled by a very simple interface (start/stop RTP
packetization with parameters x, y, z).</font></blockquote>
<blockquote type="cite" cite><font face="Arial" size="-1">Just to have
a tought, since you know the livemedia library better then me, how
many hours/days/month would you think is needed to do the
changes?</font></blockquote>
<div><br></div>
<div>I don't know nearly enough about your system to estimate this
(and I'm usually bad at such estimates anyway).</div>
<div><br></div>
<div>Good luck!</div>
<x-sigsep><pre>--
</pre></x-sigsep>
<div><br>
Ross Finlayson<br>
Live Networks, Inc.<br>
http://www.live555.com/</div>
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