[Speex-dev] Jitter buffer usage

[Baldvin Hansson]

Dear all.
 
Questions regarding VoIP implementation and the use of the Speex jitter
buffer, if I may:
 
Am I right in my understanding that the Speex jitter buffer implementation
is used only on the receiving end of a network VoIP stream?
 
1)
The sender would sample+encode+timestamp packets/frames of speex data and
send via UDP to receiver.
 
UDP packet would be constructed as: [TIMESTAMP][Speex data].
 
2)
The receiver would receive UDP data and call speex_jitter_put(...) for each
packet received.
 
3)
The decoder would call speex_jitter_get(...) to get the next packet to
decode and play.
 
Far off, or close to how the big boys would do it?
 
Any information is greatly appreciated.
 
Respectfully,
Baldvin
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