[Freeswitch-users] Changing codec during calls

[Daniel Ivanov]

Freeswitch is turning into a mean, mean event-driven voip panacea...
Congrats on the new commit.
On Feb 22, 2013 6:49 AM, "Emrah" <lists at kavun.ch> wrote:

> Hi guys,
>
> Sorry was disconnected from the list for a while.
> Indeed it looks like the latest commit fixed it.
> The issue I see now is that the channel info are not updated and a show
> channels will only list the initial codecs.
>
> Great job and thanks a bunch,
> Emrah
> On Feb 21, 2013, at 8:42 PM, Dmitry Lysenko <dvl36.ripe.nick at gmail.com>
> wrote:
>
> > Now this cool feature is fully working! Today Anthony committed the
> bugfix!
> > Tested with 2 different SIP UA<->FS<->Callcentric. Both legs.
> > Now possible to write application that will switch codecs on the fly,
> regarding on any event, such as switching to low bandwidth backup internet
> line or too high cpu load, etc.
> > BTW, is there way to get API access to statistic of jitter buffer?
> > Thanks.
> >
> > P.S. As I know, asterisk can't do such thing.
> > _________________________________________________________________________
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> > 
> > 
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>
> 
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