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

I put the Who? in Mishehu mishehu at freeswitch.org
Thu May 16 20:44:01 MSD 2013


See inline responses.

On 05/16/2013 02:40 AM, Fernando Hernandez wrote:
> 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?
>

Yes, it is possible.  You have to look at the code for another codec's 
module for basic reference.  Yes, there *can* be licensing issues, 
especially if you wish to distribute the code in any fashion.  You need 
to read both the MPL 1.1 as well as the license terms for whatever 
libraries you are linking your new module with. But before I go any 
further, the issue with g729 isn't that the source code is necessarily 
restrictively licensed - it's that the codec itself is protected under 
patents and thus subject to per-channel licensing.  I do find myself 
curious to know what this implementation of g729 that you are planning 
to work with is, and what are the improvements that it offers over the 
licensed g729 in the FreeSWITCH tree.  I am curious in part because I 
know the guys put in a lot of effort to be able to provide legal 
licensing for g729 in FreeSWITCH, and if there's a viable way to improve 
on the code, there might be the option to provide a bounty for it or 
make the contribution yourself back to the FreeSWITCH code (without the 
use of the proprietary code, of course).

> 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?

Are you transcoding audio on these calls?  That's where most of the CPU 
power would go to if the audio is indeed the issue.

>
> Thank you very much,
>
> Fernando Hernandez
>
>
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