[asterisk-dev] Regarding SIP performance

[Chirag Vaishnav]

Sorry to disturb you friends as you are diverted on different problem.

I know asterisk is not a proxy, and I'm not looking for proxy. I am not 
saying that we should remove some functionality from it and the peer to peer 
scenario was just for an example.

As you are discussing now if both channels supports same codec, RTP stream 
should be direct between them without re-invite = yes option so load on  
asterisk is less.

But the thing is that performance is limited by the the sip channel 
creation, before we deal with RTP. If we fire 100 calls per second when the 
number of calls reach to 500-600 retransmission starts for comming invites 
means asterisk is not able to handle that much call at the same time.

And this is not because of the process asterisk do for a call establishment, 
this because of the way asterisk do it. At a given scenario a single thread 
consume 100% processing power so you can't  take advantage of dual-core or 
quad processer.

So my question is are you sure the way it process the call is the only way 
to do it, or it is well optimized? can't we improve on that?

Ok let's forget about we can do it or not. Let me know  this issue is 
important for you or not.

- Chirag

_________________________________________________________________
One and only Ash. Find out all about her. Only on MSN Search 
http://server1.msn.co.in/profile/aishwarya.asp