[VoiceOps] Homegrown SIP load testing platform

[Jon Chleboun]

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.

Goals:
- Generate somewhere on the order of 20k phone calls with real SIP and RTP.
- Route the flows through our VoIP infrastructure to test performance limits.
- 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.

Attempted already:
- 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.

Looks interesting:
- 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?


Many thanks,

Jon Chleboun  




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