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<DIV dir=ltr align=left><SPAN class=339492909-30052012><FONT color=#0000ff
size=2 face=Arial>Hello Ross,</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=339492909-30052012><FONT color=#0000ff
size=2 face=Arial></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=339492909-30052012><FONT color=#0000ff
size=2 face=Arial>does it mean that we don't have any means to handle
tool-low-bandwidth network case? I can imagine for instance for the live view
the following scenario - we monitor the output queue length and when it's
overflown or close to the top - start to skip all incoming frames till the next
key frame (or till the queue is lowering)? That makes video not ideal, but at
least customer see something (with gaps).</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=339492909-30052012><FONT color=#0000ff
size=2 face=Arial>For the playback usually it's possible to ask the source part
to hold on frame delivery and I guess, again I can monitor output queue length
and tame my backend server stream.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=339492909-30052012><FONT color=#0000ff
size=2 face=Arial></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=339492909-30052012><FONT color=#0000ff
size=2 face=Arial>Does it make sense in live555?</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=339492909-30052012><FONT color=#0000ff
size=2 face=Arial></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=339492909-30052012><FONT color=#0000ff
size=2 face=Arial>Nikolai</FONT></SPAN></DIV><BR>
<DIV dir=ltr lang=en-us class=OutlookMessageHeader align=left>
<HR tabIndex=-1>
<FONT size=2 face=Tahoma><B>From:</B> live-devel-bounces@ns.live555.com
[mailto:live-devel-bounces@ns.live555.com] <B>On Behalf Of </B>Ross
Finlayson<BR><B>Sent:</B> Saturday, May 26, 2012 11:56 AM<BR><B>To:</B> LIVE555
Streaming Media - development & use<BR><B>Subject:</B> Re: [Live-devel] RTSP
Client TCP mode<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV>
<BLOCKQUOTE type="cite">
<DIV>I have a live555 RTSP server serving a 3mbps mpeg2ts file. The server is
hosted with public internet connection.<BR><BR>When openRTSP is used in
RTP/UDP mode its able to receive the file properly. <BR></DIV></BLOCKQUOTE>
<DIV><BR></DIV>Yes, but if your network's capacity is less than 3 Mbps, then you
will lose data, but the losses will occur on Transport Stream 'sync byte'
boundaries.</DIV>
<DIV><BR></DIV>
<DIV><BR>
<BLOCKQUOTE type="cite">
<DIV>However when RTP/TCP mode, openRTSP prints "Missing sync byte!" and
stops getting data. <BR>On the server the Receiver Report shows lot of packet
loss. <BR><BR>For low bitrate content TCP mode works fine.
<BR></DIV></BLOCKQUOTE>
<DIV><BR></DIV></DIV>Your stream is exceeding the bandwidth of the network.
Therefore, you are going to lose data. End of story. Note that
- in this case - RTP-over-TCP streaming will not save you. If you try to
stream over a too-low-bandwidth network using TCP, you'll still get data loss,
but it'll occur at the sender (because of its OS buffer overflowing), rather
than at the receiver.
<DIV><BR></DIV>
<DIV>This is the important distinction between live media streaming, and
on-demand streaming (e.g., using the World-Wide Web). If you use our
software, it's important to understand this distinction.</DIV><BR><BR>
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class=Apple-style-span>Ross Finlayson<BR>Live Networks, Inc.<BR><A
href="http://www.live555.com/">http://www.live555.com/</A></SPAN></SPAN>
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