The call simulator was designed to fully mimic the footprint on the network of a phone call (with the codec specified), yet use ICMP (ping) as the protocol for communication. That way, you can point to any IP address and test to/from that location easily. We send out the exact same size packet, inter-packet-gapping, and transmit frequency as a RTP data stream.

There are some possibilities that will reduce the accuracy of testing. For example:

  • Don’t run the call simulator on a virtual computer. A virtual machine is time-sliced to give resources to other computers on the same physical hardware. As a result, we cannot be accurate when running the call simulator when we are not running against physical hardware.
  • Don’t point the call simulator to a virtual machine as the destination computer. This is for the same reasons as above.
  • Don’t simulate more than 1 call and point it to an actual VoIP handset. You can easily overload the small CPU on the phone if you attempt to send more than 1 call to a VoIP phone (the phone is typically designed to handle only 2 calls at the same time. Additional calls conferenced in are usually handled by a centralized conference server.)
  • If you point the call simulator to a router that is overloaded with traffic, it may show packet loss. Routers are designed to principally route traffic, and respond to ICMP requests as a lower priority request. If you see packet loss to/from a router, check to see if the router is overloaded (high CPU utilization) which could mean that you should upgrade the router.