[asterisk-users] Asterisk V/s FreeSwitch

[isrlgb at gmail.com]

I run about 150 cc on a xen vps  with no problem mostly with no transcoding but I could have 15 channels transcoding and 15 channels are recorded 

I have a fs server on it too but not much more traffic so can't compare

If asterisk would use the sofia sip stack it would probably be about the same but the license doesn't match so asterisk brewed there own 
I remember one of the developers writing a blog or a post (I can't find the link)  were he compiled asterisk to use the sofia stack and got very nice results  

-----Original Message-----
From: "Jeff Brower" <jbrower at signalogic.com>
Sender: asterisk-users-bounces at lists.digium.com
Date: Wed, 8 Feb 2012 07:54:04 
To: Brynjolfur Thorvardsson<binni at itanet.nu>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
	<asterisk-users at lists.digium.com>
Cc: <asterisk-users at lists.digium.com>
Subject: Re: [asterisk-users] Asterisk V/s FreeSwitch

Brynjolfur-

> According to this article here:
>
> http://anders.com/cms/266
>
> the difference mainly lies in how FreeSwitchs handles open
> channels in comparison with Asterisk. FS uses one thread
> per channel while * keeps jumping between threads. At least
> that's how I understand it.

If the difference really is 10:1, then I doubt that threads vs. linked lists completely explains it.

But the difference may not be that much, as some other posts indicate.  I would suggest to Virendra to make sure he's
comparing identical configurations:  machine type/speed/mem, same type of calls, same amount of call RTP handling
(G711, no echo can, no recording, no DTMF, etc), latest versions of both softwares, and so on.  That would be a good
test.

Since the metric in this case is concurrent calls, not CPS, it could be that for some reason, Asterisk's RTP coding
isn't as efficient.

-Jeff

> Fra: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] På vegne af virendra
> bhati
> Sendt: 8. februar 2012 06:34
> Til: Asterisk Users Mailing List - Non-Commercial Discussion
> Emne: Re: [asterisk-users] Asterisk V/s FreeSwitch
>
> thanks Gilles,
>
> After reading these web links. it's pretty clear that FreeSwitch is batter then Asterisk feature, quality wise. But
> asterisk is easy to used.
>
> But the question is still open from my end.
>
> How FreeSwitch can support 1000CC but asterisk not ?
>
> Because FreeSwitch used XML as configuration and asterisk plan text file ?
> FreeSwitch used sofia_sip and asterisk used sip ?
> Asterisk is PBX and FreeSwitch is SoftSwitch ?
>
> On Tue, Feb 7, 2012 at 9:10 PM, Gilles <codecomplete at free.fr<mailto:codecomplete at free.fr>> wrote:
> On Tue, 7 Feb 2012 17:08:18 +0530, virendra bhati <virbhati at gmail.com<mailto:virbhati at gmail.com>>
> wrote:
>>Why FreeSwitch can handle more then 1,000CC and asterisk only 25CC ? What
>>technology FreeSwitch is used and asterisk don't. I don't know it's the
>>right or wrong but this question come to my mind...
> Provided Asterisk, even in release 1.8 or 10, does handle much fewer
> concurrent calls than Freeswitch, you might find the answer in those
> articles:
>
> "How does FreeSWITCH compare to Asterisk?"
> www.freeswitch.org/node/117<http://www.freeswitch.org/node/117>
>
> "Asterisk vs FreeSWITCH"
> www.richappsconsulting.com/blog/blog-detail/asterisk-vs-freeswitch/<http://www.richappsconsulting.com/blog/blog-detail/asterisk-vs-freeswitch/>
>
> "Asterisk vs. FreeSWITCH"
> www.anders.com/cms/266<http://www.anders.com/cms/266>
>
> "Open Source VoIP: Asterisk or FreeSwitch?"
> www.zdnet.com/blog/greenfield/open-source-voip-asterisk-or-freeswitch/233<http://www.zdnet.com/blog/greenfield/open-source-voip-asterisk-or-freeswitch/233>
>
> "FreeSwitch vs Asterisk"
> www.dslreports.com/forum/r23246683-FreeSwitch-vs-Asterisk<http://www.dslreports.com/forum/r23246683-FreeSwitch-vs-Asterisk>
>
>
>
> Thanks and regards
>
>  Virendra Bhati
> +91-8885268942
> Software Engineer
> E-mail-: virbhati at gmail.com<mailto:virbhati at gmail.com>
> Skype id:- virbhati2


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