[Asterisk-Dev] Packetization discussion

[Dan Austin]

I've been forward porting the packetization patch that anthm posted to
bugid 5162, as it
fills a distinct need my orginazation has.

I recall that when it was posted and announced on this list, there we a
number of objections
to the way it was implimented, but not so much to the concept.  I'd like
to try and spark
another discussion.  

Forward porting it is trivial, so I don't mind maintaining it for our
needs, but it seems like
a very useful feature.

[Summary]
The patch allows channels (sip and ooh323c) to send RTP packets of a
selected size,
based on desired ms of audio per packet.  Asterisk currently accepts and
deals with
RTP packets up to 60ms (G729A), but can only send 20ms sized packets.
This patch,
and the related patch for chan_ooh323, allow setting a desired size on a
global, per peer
and per user basis.  This patch DOES NOT currently allow a per codec
setting, which is
what I think the original objections concerned.

Dan