[Asterisk-Dev] Change codec on running SIP channel

[BJ Weschke]

 If you wanted to do it with some framework that's already in chan_sip, I'd
imagine the way to do it would be to invoke a re-invite back to the same
endpoints with different codec preferences, and once you've got a succesful
connection up, transfer the bridge to the newly established channels. I'm
making this sound alot more simple than it really is though because you're
also going to have to try and work around the configuration values for
codecs on these endpoints that are going to want to use the same codecs
they've already negotiated for the first bridge.
  I believe one of the goals of chan_sip ("chan_sip3") that is being planned
for post 1.2 is to make codec switches while a channel is still "hot" alot
easier to do than what it would be through a programming effort now.

 On 10/17/05, Michael Manousos <manousos at inaccessnetworks.com> wrote:
>
>
> I'm not looking for a configuration-based method.
> I would like a programmer's approach on that.
>
> Michael.
>
> BJ Weschke wrote:
> > It's not presently possible via specific configuration of Asterisk.
> >
> > On 10/17/05, *Michael Manousos* <manousos at inaccessnetworks.com
> > <mailto:manousos at inaccessnetworks.com>> wrote:
> >
> >
> > Anyone knows what is the proper way to change the codec
> > of an established SIP channel?
> >
> > Michael.
> >
> >
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