[Asterisk-Dev] (Re)Invite for codec change

[Andrew Lindh]

Is there any thought to enable/support/force a (re)invite to only change
codecs on "bridged" channels that no longer have matching codecs? This
would reduce asterisk transcoding overhead while still allowing asterisk
to stay in the middle of a call for billing/hold/etc...It would be nice
to support lots of devices/users on the same system without killing it
or scaling with lots of systems.

I'm not clear if SIP allows this and how well devices will deal with it.

The idea is if a PRI is on a gateway (say cisco 5300) and it uses g.711 to
the asterisk for IVR and then gets transfered to an off-site location
that wants to use g.729. There is no reason asterisk should have to do
all the work. The gateway has DSPs that can do G.729......I know one answer
is to just use G.729 or GSM for everything, but it's not a usable answer.
The only thing that everything supports and does not eat CPU time is G.711

Yes I know if it's a PRI on TDM card in the asterisk box then something has
to do compression anyway, but there are a lot of external gateways that
have lot's of DSP time waiting to be used.

  Andrew