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<div class="moz-cite-prefix">Il 17/05/2015 18:13, Sebastian Dröge ha
scritto:<br>
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<blockquote cite="mid:1431879213.1700.1.camel@centricular.com"
type="cite">
<pre wrap="">On So, 2015-05-17 at 17:51 +0200, Mailing List SVR wrote:
</pre>
<blockquote type="cite">
<pre wrap="">Hi,
Sebastian, based on my experience, when there is low bandwidth between
client and server some rtsp server drop something in tcp mode (I think
it is similar to what multifdsink do when a client is late).
Please take a look at the attacched capture (rtsp over tcp), you can see
several gaps in sequence number, for example 60-61 ... 63 or
65...67...70 or 79-80...85 and so on, however this stream is properly
handled by gstreamer, so I don't fully understand your answer, I know
that tcp is a reliable protocol but probably the sender can dedice to
drop something probably based on client sending queue size,
</pre>
</blockquote>
<pre wrap="">That makes sense, yes. We would have to implement something to resync
with TCP after a gap then. The main problem here is that for TCP we only
measure the capture time at the very beginning once, not for following
packets (as TCP generally has a lot of buffering everywhere and we can't
assume that packets arrive in real time, like we do with UDP).</pre>
</blockquote>
<br>
are you sure? Without looking at the code this seems not true, I
have several files saved from rtsp over tcp with gap properly setted,<br>
<br>
the only difference from 1.4 and git master seems the new error
about packet without DTS after gap that seems to happen only
sometime,<br>
<br>
Nicola<br>
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<blockquote cite="mid:1431879213.1700.1.camel@centricular.com"
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