Battle Titans FreeSwitch vs. Asterisk - Performance Test

Not so long ago I switched to FreeSwitch, the system is really interesting and gives very great opportunities, even commercial counterparts are lagging behind in their capabilities. Gradually I am going away from using Asterisk in my daily activities.
Many have a question: what is the best FreeSwitch of the tested and tested Asterisk ???
Why to understand sometimes not in the most simple dialplan in the XML language ?? When there is an Asterisk!
I think FS is no longer inferior to Asterisk unless there is a web interface that is inconsistent in functionality as FreePBX ...
The rest only surpasses it!
Description of the merits can be found here.
There is the same opengsm analog chan_dongle, freetdm analog Zap and others.
Most of all, I was struck by the sound quality of the FS, even on a simple G.711, which is somehow better than that of Asterisk.
And what about HD codecs (CELT, Siren, Silk, ARM-WB), and there are already a lot of softphones that work with them. Asterisk also somehow supports these codecs but they sound disgusting, not in any comparison!
An example of Bria or Linphone for Android that can work with them.



The article itself is written to tell the results of the test!
')
There is such an interesting utility of HP SIPP which allows you to generate VoIP traffic and create various scenarios, thereby testing softswitches.

As a server for testing, the PC performed with the parameters:
Intel Core i3 3200
4 Gg DDR3
500 Gb HDD
something like this.

Preloaded with Debian 6.0.4

Testing method:
sipp made 10 calls per second (in fact, you can choose others as well), in response to the call, musiconhold was included, plus rtp echo was made to everything, as a full conversation with the system.

sipp 192.168.1.200 -s 110 -i 192.168.1.4 -d 2h -l 1 -aa -mi 192.168.1.4 -rtp_echo -nd -r 10

The first subject was Asterisk 1.8.13 compiled from sources:







Aster kept 168 connections in real time and then either fell at all or tightly hung the system.

As a little, I thought, it was thought that somehow did not correctly collected it! I put the system from Asterisk 1.8.11 from the Digium repository from scratch, I did the same with Asterisk 10.5 and the result is the same!

It can be assumed that in real life calls would be even less so this is not a full call but a call from one side.

The following was FreeSwitch 1.2:







It can be seen that at half CPU load FS holds 1000 calls !!! As it turned out, there is a software limit for 1000 calls in the FS itself, although I haven’t yet found where to remove it.

We can assume that another 1000 he would have pulled))

And the last for the test was the build of Elastix 2.3, it was very interesting how it works already in the boxed solution. And the results struck me! Essentially the same Asterisk:





Elastix kept about 500 calls and without suspending the system at 100% loading of the process!

It seems that CentOS is somehow more adapted to VoIP, although perhaps the guys from Elastix themselves have somehow turned Aster))

Even if so, FS bypasses Asterisk 4 times at least while working normally and without loading the system !!!
The test was conducted by the community of the VoIPLab group.