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

[Karl Brose]

Don't know if I grasp the issue correctly, but I think if Asterisk where 
smarter about SIP media
setup it probably wouldn't have to stay in the media path just for DTMF 
or transfers in many cases.
To do so, it only needs a one-way channel from the end of the call from 
where DTMF or other
inband signaling is expected and doesn't have to stay in the path for 
the translation of the entire
call. That means your gateway could do all the hard work, and * would 
just sit there and listen
for any signaling.  It should be possible with some work on the SDP 
offer/answer model in SIP.
Is this what you would like to address?

Andrew Lindh wrote:

>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
>  
>
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