[Freeswitch-users] Changing codec during calls

[Dmitry Lysenko]

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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