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<DIV dir=ltr align=left><SPAN class=122551310-12072010><FONT face=Arial
color=#0000ff size=2>I just tried to modify the fDurationInMicroseconds but it
doesn't work.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=122551310-12072010><FONT face=Arial
color=#0000ff size=2>Actually, It seems the task sends packet through
"sendPacketIfNecessary" as soon as the packet is prepared when I would like to
wait for 3 or 4 packets ready before to send them on the
network.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=122551310-12072010></SPAN><FONT
face=Arial><FONT color=#0000ff><FONT size=2>I<SPAN class=122551310-12072010>
tried to find a way to skip to send data but when I do that, it seems also than
the packet is lost...</SPAN></FONT></FONT></FONT></DIV>
<DIV><FONT face=Arial color=#0000ff size=2></FONT><FONT face=Arial color=#0000ff
size=2></FONT> </DIV>
<DIV><SPAN class=122551310-12072010><FONT face=Arial color=#0000ff size=2>It
looks like the output buffer gathers frames and when there are enough frames to
do a packet the RTP packet is done and sent, could I let prepare many packets
and wait for few to effectively send them ?</FONT></SPAN></DIV>
<DIV><SPAN class=122551310-12072010><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=122551310-12072010><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><BR></DIV>
<DIV class=OutlookMessageHeader lang=fr dir=ltr align=left>
<HR tabIndex=-1>
<FONT face=Tahoma size=2><B>De :</B> live-devel-bounces@ns.live555.com
[mailto:live-devel-bounces@ns.live555.com] <B>De la part de</B> BONNEAU
Guy<BR><B>Envoyé :</B> dimanche 11 juillet 2010 05:51<BR><B>À :</B>
LIVE555 Streaming Media - development & use<BR><B>Objet :</B> Re:
[Live-devel] RTP packet lost when Live555MediaServerstreamsforward Dektec
DTE3114<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV class=WordSection1>
<P class=MsoNormal><SPAN
style="FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: 'Calibri','sans-serif'">This
problem seems to be very similar to one I have had about 1 year ago while
trying to stream a Mpeg2 TS with the live555 library with a very high datarate.
Unfortunately I do not remember all the details but I eventually pinpointed the
problem to the scheduler of Windows. (My testing application was Windows based).
If a remember the library would compute a delay to callback a routine to
schedule a new RTP packet to be sent over the network. I believe that was done
through a select or sleep() function. The computing of the delay was well done
by the library. But the timer resolution of Windows is 15 or 10 ms by default
and the requested delay in my case was much lower given the requested bitrate
from the Mpeg2 TS stream. The jitter of the packets over the network was
terrible and the library much of the time was trying to catch-up from the
misbehavior of the Windows Timer default resolution of 15 ms by bursting the RTP
packets over the network.<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN
style="FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: 'Calibri','sans-serif'"><o:p> </o:p></SPAN></P>
<P class=MsoNormal><SPAN
style="FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: 'Calibri','sans-serif'">I
solved that problem by using the TimeBeginPeriod(1) to change the Windows period
of the timer to 1 ms. The callback function would then wakeup much faster and
the jitter of the packets went down to an acceptable
level.<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN
style="FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: 'Calibri','sans-serif'"><o:p> </o:p></SPAN></P>
<P class=MsoNormal><SPAN
style="FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: 'Calibri','sans-serif'">Guy
Bonneau<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN
style="FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: 'Calibri','sans-serif'"><o:p> </o:p></SPAN></P>
<P class=MsoNormal><SPAN
style="FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: 'Calibri','sans-serif'"><o:p> </o:p></SPAN></P>
<DIV
style="BORDER-RIGHT: medium none; PADDING-RIGHT: 0in; BORDER-TOP: medium none; PADDING-LEFT: 4pt; PADDING-BOTTOM: 0in; BORDER-LEFT: blue 1.5pt solid; PADDING-TOP: 0in; BORDER-BOTTOM: medium none">
<P class=MsoNormal><SPAN
style="FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: 'Arial','sans-serif'">Just to
complet my explaination, please find the 2 IO Graphs Qvidium vs Live555 rtp
streaming.</SPAN><o:p></o:p></P>
<P class=MsoNormal> <o:p></o:p></P>
<P class=MsoNormal><SPAN
style="FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: 'Arial','sans-serif'">Regards</SPAN><o:p></o:p></P>
<P class=MsoNormal><o:p> </o:p></P>
<DIV class=MsoNormal style="TEXT-ALIGN: center" align=center><SPAN lang=FR>
<HR align=center width="100%" SIZE=2>
</SPAN></DIV>
<P class=MsoNormal style="MARGIN-BOTTOM: 12pt"><B><SPAN lang=FR
style="FONT-SIZE: 10pt; FONT-FAMILY: 'Tahoma','sans-serif'">De :</SPAN></B><SPAN
lang=FR style="FONT-SIZE: 10pt; FONT-FAMILY: 'Tahoma','sans-serif'"> Lambert
Marc <BR><B>Envoyé :</B> vendredi 9 juillet 2010 15:35<BR><B>À :</B>
'live-devel@lists.live555.com'<BR><B>Objet :</B> RTP packet lost when
Live555MediaServer streams forward Dektec DTE3114</SPAN><SPAN
lang=FR><o:p></o:p></SPAN></P>
<DIV>
<P class=MsoNormal><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">Hello,</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=MsoNormal> <o:p></o:p></P></DIV>
<DIV>
<P class=MsoNormal><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">First of all, I'm
certainly not the first to say that but thank you for this very nice
software.</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=MsoNormal><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">I use
live555MediaServer in the context of my job and I compilled it with the option
"RTSP_ALLOW_CLIENT_DESTINATION_SETTING" to make me able to control the server
from a device connected on the network and to stream forward an IPtoTS modulator
(DTE-3114 from Dektec) providing a DVB-C signal to a
TV.</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=MsoNormal><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">My client is working
well and everything is working perfectly with VLC
but...</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=MsoNormal><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">When I replace VLC by
the DTE-3114, the outgoing signal is not good with many RTP packet
loss.</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=MsoNormal><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">Then, I tryed to
change many parameters on each sides but nothing is fixing
that.</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=MsoNormal> <o:p></o:p></P></DIV>
<DIV>
<P class=MsoNormal><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">I also used the
DTE-3114 with the QVidium file streamer (it doesn't have the rtsp server, just
the rtp streamer) and, in that case, I succeed to stream perfectly until
25Mbps...</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=MsoNormal><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">I captured with
Wireshark and there is a difference between the 2 streaming methodes, the
Qvidium seems to allow more time before to send the next frame and the
frame is composed of many UDP packet when Live555 try to gather perfectly the
packets during the time and I guess it is stressing too much the DTE-3114 to
treat this bandwidth on the fly.</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=MsoNormal><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">Then, I would like to
know if it is possible to change the way to send packet on the network to try to
make my test bench working.</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=MsoNormal> <o:p></o:p></P></DIV>
<DIV>
<P class=MsoNormal><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">Thank's in
advance</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=MsoNormal> <o:p></o:p></P></DIV></DIV></DIV></BODY></HTML>