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<DIV><SPAN class=497235419-06062005><FONT face=Arial size=2>Dear
all.</FONT></SPAN></DIV>
<DIV><SPAN class=497235419-06062005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=497235419-06062005><FONT face=Arial size=2>Pardon the
ignorance, but I'm wondering about the relationship between RTP and the (for
example) Speex jitter buffer. This may not relate directly to speex as such, but
I hope you'll take the time to reply anyway should you have any details for
me.</FONT></SPAN></DIV>
<DIV><SPAN class=497235419-06062005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=497235419-06062005><FONT face=Arial size=2>If I would implement
my VoIP application using RTP to transmit data between recording-point and
playback-point, would I still use the jitter buffer or does RTP actually take
care of the jitter buffer tasks?</FONT></SPAN></DIV>
<DIV><SPAN class=497235419-06062005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=497235419-06062005><FONT face=Arial size=2>If I use the Speex
jitter buffer implementation (or any other jitter buffer implementation for that
matter), should I expect to get a decent voice data transfer/decoding by sending
the speex encoded data (along with a simple time-stamp of when the frame/packet
was prepared) via UDP and relying on the jitter buffer implementation to supply
a steady stream of ready-to-use packets?</FONT></SPAN></DIV>
<DIV><SPAN class=497235419-06062005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=497235419-06062005><FONT face=Arial
size=2>Respectfully,</FONT></SPAN></DIV>
<DIV><SPAN class=497235419-06062005><FONT face=Arial
size=2>Baldvin</FONT></SPAN></DIV>
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size=2></FONT></SPAN> </DIV></BODY></HTML>