[Freeswitch-users] Changing codec during calls

Dmitry Lysenko dvl36.ripe.nick at gmail.com
Mon Feb 25 10:53:34 MSK 2013


Emrah, please file a bug to jira. Btw, are you checked changing of codec
ptime  during call?
Thanks.


2013/2/25 Emrah <lists at kavun.ch>

> Is it necessary to file a bug on the channel info issue?
> The channel data does not get updated when changing codecs and a "show
> channel" always shows the first negotiated codec.
>
> This is an incredible feature and ever so useful, thanks a bunch.
>
> E
> On Feb 22, 2013, at 11:59 AM, Daniel Ivanov <sertys at gmail.com> wrote:
>
> > 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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