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" <[email protected]> Sender: [email protected] Date: Wed, 8 Feb 2012 07:54:04 To: Brynjolfur Thorvardsson<[email protected]> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Cc: <[email protected]> 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: [email protected] > [mailto:[email protected]] 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 > <[email protected]<mailto:[email protected]>> wrote: > On Tue, 7 Feb 2012 17:08:18 +0530, virendra bhati > <[email protected]<mailto:[email protected]>> > 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-: [email protected]<mailto:[email protected]> > Skype id:- virbhati2 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
