[Asterisk-Users] RTP Packetization

[Patrick Neubauer]

Hi all,

I need to be able to adjust packet sizes and found the patch at 
http://bugs.digium.com/view.php?id=5162

Thus, I checked out and compiled 
http://svn.digium.com/view/asterisk-old/team/group/5162_rtp_packetization

I added the line "packetization = 30" for one peer in my sip.conf and 
started asterisk with the "-I" switch for async RTP.
That's all it takes according to the 5162 issue page. Nevertheless, 
asterisk still keeps sending it 20ms packets, even though a "sip show 
peer foobar" shows Packetization: 30.

What could be wrong? What about that ztdummy thing for internal timing? 
Is this necessary to run asterisk properly? Is it important for 
packetization?

Regards, Patrick