Although this probably isn't the "right" way of doing it, you can
change in the source code, globally for all calls using a codec:
See the "smooter" creation statement in the function ast_rtp_write:
rtp->smoother = ast_smoother_new(4 * 50);
(I changed mine to 50 ms for G726 which did wonders for those slooooow
DSL users to reduce the number of packet/sec, and the latency increase
is virtually not noticeable to me).
I'm sure we could make a patch to set it on a per-call basis from the
dialplan... if someone cares to do so.
--Luki