[Speex-dev] RTP and jitter buffer relationship

[Baldvin Hansson]

Dear all.
 
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.
 
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?
 
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?
 
Respectfully,
Baldvin
 
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