[asterisk-dev] Regarding SIP performance

[Chirag Vaishnav]

Hi !

I am chirag from india and I am working with SIP part of asterisk.  
Currently I am doing testing with SIPP tool. Now the thing is that asterisk 
is not able to handle more then 550 concurrent calls with call rate 100 
call/sec (on my amd 64bit dual core machine with 2 gb ram) which is to less. 
I am doing this with canreinvite mode so there is no issue of RTP traffic. I 
just want to astablish peer to peer call thru asterisk. As i have said doing 
550 call with 100 call/sec ( I hang up calls after some time so this is 
including all invite and hangup related massages) it consume full processing 
power and retransmission starts if i increase number of calls.

If i keep firing  100 calls/sec without hangup retrans starts after 3200 
calls.

Can any one tell me why it is so? is there any way to increase number of 
concurrent calls with same or more fire rate?

- Chirag

_________________________________________________________________
Spice up your IM conversations. New, colorful and animated emoticons. Get 
chatting! http://server1.msn.co.in/SP05/emoticons/