[Asterisk-Dev] H323 processing limit

[Abdul Hakeem]

Hello,
Has anyone done any Sip simulations ?. I need to know how many
concurrent Sip calls that can be handled with and without codec-ing.
Regards,
Abdul Hakeem
-----Original Message-----
From: asterisk-dev-admin at lists.digium.com
[mailto:asterisk-dev-admin at lists.digium.com] On Behalf Of Jeremy
McNamara
Sent: Wednesday, April 16, 2003 9:52 PM
To: asterisk-dev at lists.digium.com
Subject: Re: [Asterisk-Dev] H323 processing limit



Unfortunately, it depends on which channel driver your running.
chan_h323 is setup so that you can take advantage of the codec
pass-through feature of Asterisk.   We have simulated 125 concurrent
calls setup in random (sub-15 second) intervals, using passthru, on a
500mhz P4 without causing more than a 5 second delay in call setup time.
We have a few separate production systems (single 2.53 ghz,  P4, 2gig
ram) with running pass-thru and have never had a single compliant about
audio problems, but at this point we've only pushed about 3 T-1s of
concurrent real traffic on any one system.

Now, if you have to do the actual codec-ing on the machine then you are
going to lose overall number of calls due to the processor time it takes
to encode and decode whatever codec your running.   On that same 500mhz
P4 test machine, we were able to simulate 45 concurrent calls using GSM,
again without a long delay in call setup time.

With that other H.323 driver we couldn't make10 concurrent calls work
reliably, so we wrote chan_h323.


Jeremy McNamara


Xara Developement wrote:


Hi.
I need to know aproximately, how many H323 calls coming from a ethernet
10/100 card , i can manage with Asterisk running on a Pentium III 750
MHZ machine with Linux RH9.
Any source of bechmarks ?
 
Thank you.
 
Rodrigo.




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