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After a VoIP server is installed, it is always an important thing to do a stress test. This will allow us to know the aproximate number of simultaneous calls (aka CPS) the server can support before start having issues. This kind of tests allows you to do a metric about availability, as you can know how tolerant is your server to high loads of traffic.

The test result is not always exact; there are many factors involved, such as:

  • network conditions
  • destination to test
  • transcoding
  • other processes executing in the same server
  • XML handler configuration
  • database configuration (if apply)
  • authentication type and
  • others

So, remember the number you will get here, it won't be an absolute true but it will give you a good idea about what to expect. Another thing to have in mind is that production and laboratory are very different, just in case if something doesn't make sense.

To stress your PBX do the following steps:

  1. Install sipp tool (from your Linux distribution or accessing http://sipp.sourceforge.net/)
  2. Download scenario 1 and 2 files from the download area. This files will only work with IP authentication
  3. Edit acl.conf.xml file and add the IP you will use as LOCAL_IP_ADDRESS then, execute fs_cli -x 'reloadacl'
  4. Edit the downloaded files and do these substitutions:
    1. MY_DESTINATION: where you want to call to, for FusionPBX you can use any destination specified in the inbound routes if you are using a vanilla Freeswitch select any destination in the public context. Please read your dial plans before.
    2. MY_DOMAIN: the IP or domain (depends if installation is multi-tenant or not)
    3. PCAP file routes
  5. Execute echo -e "SEQUENTIAL\nXXX" > data.txt where XXX is a SIP extension or a DID where you will generate call from
  6. Execute sipp FS_IP_ADDRESS -i LOCAL_IP_ADDRESS -sf scenario1.xml -inf data.txt -r 2 -m 4000 -trace_rtt -trace_err -trace_screen -l 200 where FS_IP_ADDRESS is the Freeswitch server IP and LOCAL_IP_ADDRESS is a local IP local (or public IP). Please read sipp manual to customize options for your specific case.
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You will be able to send more calls to the PBX, pay attention when the tool starts to report missing packets. 

Good Luck!

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