On Fri, 21 Feb 2014, Georgi Stefanov wrote:

Thanks for the suggestion Claus

But of course I have tried with one call - no problem In matter of fact no problem is seen up to 25 CPS and 250 total calls (10 sec call duration) Somewhere between 25 and 40 CPS the problem is rarely shown. The issue become more obvious around 100 CPS I thought it could be an issue, but I need to know I have missed some setting or not. It sounds like a bug to me

SIP usually fares much better than RTP and I would expect you to see problems with the RTP streams before seeing issues with SIP traffic. Have you verified the "voice" quality at high call rates? Are you actaully able to push the limits without dropping packets? I assume that you have seen: https://wiki.freeswitch.org/wiki/Performance_testing_and_configurations Of particular intereset is bandwidth calculator. What can you actually expect (best case) with you current setup (LAN, codec etc). And if you are still running 32 bit is suggest that you are running on older hardware. As you can see from: https://wiki.freeswitch.org/wiki/Real-world_results You should be able to push 250 calls with 20 CPS with 15% CPU on an EC2 instance with (2,4 Ghz and 4 GB RAM). I do not have enough experience pushing over the hardware limits so I do not know what to expect of Freeswitch. Proper stress testing is hard enough in itself. But my immediate reactions was that I am very surprised that you are not seeing anything in the Freeswitch log on the offending server not sending the final OK. If you are pushing over some limits you would normally see something. My first bet would be to turn up logging in sofia: http://wiki.freeswitch.org/wiki/Sofia-SIP#Debugging_Sofia-SIP Try comparing the log output for a successfull call with a tainted call under load. I would be very very surprised if this turns out to be a bug where Freeswitch silently drops actions without having logged it anywhere. With the info supplied so far it sounds more like a setup issue. Kind Regards, Claus Andersen