For Scalability, I'd choose Option B to limit the sip messages to increase another x fold from a subscriber base without putting processing power into simple overhead tasks such as options polling - responses for nat. For Fun, both can run on the same box -- make sure your gigabit port isn't utilized much and slam it with a lot of calls to see for yourself. Think about redundancy. To increase, scale it, also consider getting like Brian said, using xml_curl to obtain db output for current subscribers for registration, feature allowance, etc scheduled. Openser is a good proxy and serves a good border controller purpose, no need to touch that item, even though it's doable and being used in production as such. FS and openser do work in harmony. my two cents On Thu, 17 Apr 2008, Pete Kay wrote:
Hi,
One scalability question:
Can someone provide input to the following options?
Option A: Asterisk ( for IVR, Voicemail, Conference) + Openser ( for SIP registration ) Option B: Freeswitch + Openser Option C: Freeswitch
So, it is definite that Asterisk is not as scalable as Freeswitch. Among the above three options, which one is more preferable in terms of scalability? If Freeswitch is used, is Operser still needed?
Thanks alot in advance for your input.
Thanks, Pete
On Wed, Apr 16, 2008 at 11:29 PM, Daniel Hefti <dhefti@metropark.com> wrote:
It's ok, I had an idea from what you meant, but that may have been because I was looking elsewhere and already had an idea what to expect.
I originally thought you were comparing the amount of calls per some value... 10 calls in FS for every 1 call from asterisk for the same amount of value of some value. (some value being time/processing power/etc.)
-Dan
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Ken Rice Sent: Wednesday, April 16, 2008 10:09 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Asterisk vs. Freeswitch - added question
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@suspicious.org> Reply-To: <freeswitch-users@lists.freeswitch.org> Date: Wed, 16 Apr 2008 10:04:06 -0500 To: <freeswitch-users@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@telesave.pl> Reply-To: <freeswitch-users@lists.freeswitch.org> Date: Wed, 16 Apr 2008 15:40:06 +0200 (CEST) To: <freeswitch-users@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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