[Freeswitch-users] Asterisk vs. Freeswitch - added question

[Ken Rice]

Oh Oh! Shit... I made that sound like 10 FS boxes for 1 Asterisk box... I
have my ratios backwards... Its 1 FS box for 10 Asterisk boxes... Sorry for
the confusion this will invariably generate

K


> From: Ken Rice <krice at suspicious.org>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Wed, 16 Apr 2008 10:04:06 -0500
> To: <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] Asterisk vs. Freeswitch - added question
> 
> Hmmm,
> 
> I have done testing with FS and Asterisk in similar configurations on the
> same hardware, avg ratio of call handling FreeSwitch:Asterisk is 10:1
> 
> Simple test
> 
> SIPP -> (FreeSwitch|Asterisk) Run SIPP to bring up channels doing media
> playback, then call in with a SIP device to listen  to the quality of the
> audio on the call. On the particular hardware we did this testing on,
> Asterisk was about to handle about 250 call legs before starting to sound
> choppy or it couldn't keep up with the media...  FreeSwitch was able to
> handle 2500 sessions and still play media.
> 
> Current in production we have a stack of Asterisk based Call Generators for
> a specific call center appliction, these call generator are deployed in an
> approximate 10:1 ratio to FS with the FS boxes handling Rating and Routing.
> 
> Needless to say, need to load test your system? Drop me a line, we can
> generate extremely high call volumes (both concurrent and calls/sec)
> 
> That being said it would be nice for someone to come along and help out with
> a real load testing document. That is duplicatable... Any takes on that(from
> the rest of the list)?
> 
> K
> 
>> From: Marek Górecki <marek at telesave.pl>
>> Reply-To: <freeswitch-users at lists.freeswitch.org>
>> Date: Wed, 16 Apr 2008 15:40:06 +0200 (CEST)
>> To: <freeswitch-users at lists.freeswitch.org>
>> Subject: Re: [Freeswitch-users] Asterisk vs. Freeswitch - added question
>> 
>> Hi FreeSWecialists,
>> I'd like to extend title question for more details:
>> assuming Asterisk efficiency (concurrent 'normal' calls processing) for given
>> hardware
>> (meaning any, but fixed configuration server) as 1
>> [ normal call understood as no codec translation, no tricks, nothing special
>> -
>> just one
>> SIP call comes in and is connected according to dialplan to other SIP B#. I
>> try to
>> explain maybe to simple, but I'd like to learn the simplest. ]
>> 
>> what should be expected FS efficiency on same hardware ?
>> 
>> [ it was already few times mentioned similar data on this list, but never in
>> clear way,
>> so I'd like to ask current, experienced users ]
>> 
>> TkX in advance.
>> 
>> Regards,
>> /\/\arekg
>> 
>> 
>>> Hi,
>>> This question may have come up a few times already.   I am working on
>>> a application to provide IVR, voicemail, and tailored call routing
>>> services.  The SIP registration will be handled by Openser, and
>>> Asterisk is only doing the media function.   We are talking about over
>>> 100 users.
>>> 
>>> Is Freeswitch better than Asterisk in terms of functionality,
>>> ease-of-maintain, and ease-of-use?
>>> 
>>> Thanks alot in advance for your inputs.
>>> 
>>> Pete
>>> 
>>> _______________________________________________
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>>> 
>> 
>> 
>> 
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> 
> 
> 
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