I am interested to see if y'all have recommendations for putting together a
SIP load testing platform using general purpose hardware and open-source
(or inexpensive) software. We are aware of Empirix Hammer and similar
solutions, and we are looking to see if there is an alternative option.
- Generate somewhere on the order of 20k phone calls with real SIP and RTP.
- Route the flows through our VoIP infrastructure to test performance
- Receive and analyze the SIP and RTP on the other end to find out at what
load the signaling and/or media start to break down.
- SIPp spread across many servers. Here the limiting factor seemed to be
the CPU load from the interrupts from each packet. The CPU on the servers
sending and receiving the phone calls got bogged down before the VoIP core.
- We have dabbled with interrupt moderation in the NIC drivers, but this
has not seemed to help very much.
- Has anyone had success using PF_RING with Direct NIC Access and libzero
from the folks at ntop? Has anyone been able to use this with SIPp or some
other SIP and RTP generator?