[Freeswitch-users] New codec integration and concurrent calls limit questions

[Fernando Hernandez]

Hello everyone,

we are working on a videoconferece platform with freeSWITCH integrated. Now
we have a the possibility to try a new codec, proprietary, and we have
never gone too deep with freeSWITCH. I have tried to take a look at the
wiki but I didn´t find the answer to the next question,

Is it possible to integrate that new codec into freeSWITCH? according to
the codec provider it is a G.279 implementation, with several improvements.
Any licence issue? difficulty to achieve the integration?

On the other hand, we also have another issue. I have checked freeSWITCH
wiki trying to find any possible limit for concurrent calls, and according
to it, it doesn't seem to be a limit of SIP but purely the RTP. But with
the open source video conference platform we are working now this is
limited to 25, and according to the developers it is due to the audio. Do
you have any idea about this issue? We have run some test and we go over 25
people in the meeting freeSWITCH uses 125% of CPU (ref. 8 cores machine and
each core represent 100% capacity => 800%), is this a normal behavior?

Thank you very much,

Fernando Hernandez
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