[asterisk-dev] asterisk-dev Digest, Vol 104, Issue 34

[Dan Austin]

> I tested the module in the following scenario: p1 = 20ms and p2= 40ms

>> 0.00 - start conversation
>> 0.02 - p1(1) arrives, not enough data to assemble p2(1)
>> 0.04 - p1(2) arrives, p2(1) assembled, buffer is freed
>> 0.08 - repeat

> Why will then be zitter according to your simulation? Here no buffer is left
> behind for merging a packet apparently, so the packet delivery should be
> harmonic! But it is not. Can you suggest something here?

If there are no errors assembling the audio payload and the packets
are transmitted on a consistent 40ms interval, then you are correct
it should be clear sounding.  A packet capture of the RTP stream 
from the receiving end is the only way to see if the timestamps,
delivery interval and payload are consistent with the ptime selected.

I should also mention the codec used for testing may play a factor,
particularly the very low bandwidth codecs that can use a ptime that
is not a multiple of 10ms.

Dan