Hello everyone,<div><br></div><div>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 didnīt find the answer to the next question,</div>
<div><br></div><div>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? </div>
<div><br></div><div>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,<font face="arial, helvetica, sans-serif"> it <span style="font-size:13px;line-height:19.046875px;background-color:rgb(255,255,255)">doesn't seem to be a limit of SIP but purely the RTP</span></font>. 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?</div>
<div><br></div><div>Thank you very much,</div><div><br></div><div>Fernando Hernandez</div>