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<b>From</b>: "Patrick Lists" <asterisk-list@puzzled.xs4all.nl><br />
<b>Sent</b>: Thursday, February 09, 2012 10:42 AM<br />
<b>To</b>: asterisk-users@lists.digium.com<br />
<b>Subject</b>: Re: [asterisk-users] Asterisk V/s FreeSwitch</span><br />
<br />
On 09-02-12 14:52, Stefan Schmidt wrote:<br />
> Am 09.02.12 14:19, schrieb Bryant Zimmerman:<br />
>> Stefan<br />
>><br />
>> This is on target with my configuration I am working on. What kind of<br />
>> dialplan were you using when running the tests.<br />
>> Were you doing database lookups or just answering the calls and playing<br />
>> hold music. Any example would be appreciated so we can quantify your test<br />
>> results. I look forward to your response.<br />
>><br />
>> Thanks<br />
>> Bryant<br />
><br />
> the dialplan is quite simple:<br />
><br />
> for the signaling up to 13500 CC i use this wait and for the 10000CC i<br />
> enable the musiconhold<br />
><br />
> exten => monitor,1,Noop(PERFORMANCE TESTS)<br />
> exten => monitor,n,Answer<br />
> ;exten => monitor,n,MusicOnHold(806,45)<br />
> exten => monitor,n,Wait(45)<br />
> exten => monitor,n,Hangup<br />
<br />
Iirc a long time ago there was a discussion about load testing by <br />
playing MoH was not a realistic test. Something about all MoH music <br />
getting streamed synchronized so basically Asterisk only has to stream <br />
one file and sorta multiplex that single output to all the established <br />
calls (legs).<br />
<br />
[snip]<br />
<br />
> btw my normal production machines which are just the same virtual<br />
> machines like this test system. i also had 330 concurrent calls, some<br />
> with transcoding, many database lookups, musiconhold, pickup ... and the<br />
> sysload was around 1.0 ;)<br />
<br />
The difference (13500 with MoH versus 330 with a real dialplan) shows <br />
that it makes sense to mimic your dialplan in your test scenario as much <br />
as possible to see how far you can realistically push the box and still <br />
keep things stable and sound quality good.<br />
<br />
Regards,<br />
Patrick<br />
<br />
<br />
-------------------------------------<br />
Patrick<br />
<br />
I agree with you but it looks like these test show that asterisk 10 could handle a very high volume switch style application if you remove the rtp and media handling from the dialplan. For me this opens up the question as to why would I need freeswitch for high volume switching if my RTP is being handled else where. I would have a different dialplan code set for this kind of application anyway. This is why I asked Stefan to share his dialplan and testing matrix. It allows us to see raw performance what is the asterisk code base cable of and for his test case it is quite impressive. For other test cases it may not be. I would like to see someone test a full PBX implementation with independent audio files playing not just MOH and see what kind of load it can handle there. That would give me a better idea of how the two would compare. I wounder how the new 10.0 conference application would stack against freeswitches conference. How would we best design a test for that?<br />
<br />
This is a great conversation. Very productive for me at least. <br />
<br />
Thanks<br />
Bryant</span>