[Freeswitch-users] Asterisk vs. Freeswitch - added question
krice at suspicious.org
Wed Apr 16 08:04:06 PDT 2008
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
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)?
> 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
> (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.
>> 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.
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
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> Freeswitch-users at lists.freeswitch.org
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