<div dir="ltr"><div>Thanks Michael,<br><br><br></div><div>Few things I forgot to ask, the scale-ability I was referring to as horizontal scale and thanks for your input on that. <br><br>For point#2, mod_curl you mentioned for realtime, will it sustain more than 500-1K CPS and will it stress out the httpd/apache service?<br></div><div>For point#3, you mean to say Freeswitch doesn't use system files for media like Asterisk does?<br></div><div>For point#4, multi-threading; I forgot to mentioned whether Freeswitch supports multi-processing? As Asterisk, it uses/runs on single process whereas OpenSIPs/Kamilio, we can set multiple child processes.<br></div><div><div><div><div class="gmail_extra"><br><div class="gmail_quote"><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">From: Michael Jerris <<a href="mailto:mike@jerris.com">mike@jerris.com</a>><br>To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>><br>Cc: <br>Date: Thu, 13 Oct 2016 10:04:56 -0400<br>Subject: Re: [Freeswitch-users] FreeSwitch General Info<br><br>
> On Oct 13, 2016, at 9:52 AM, Ahmed Munir <<a href="mailto:ahmedmunir007@gmail.com">ahmedmunir007@gmail.com</a>> wrote:<br>
><br>
> Hi,<br>
><br>
> Currently I'm browsing different platforms online for open source VoIP technologies as currently using Asterisk as PBX.<br>
><br>
> I would like to know following things about FreeSwitch;<br>
><br>
> 1- FreeSwitch whether supports realtime integration sip profiles/extensions (using DB) and dialplan? (Just like in Asterisk)<br>
<br>
yes, most frequently using mod_xml_curl which uses http to pull xml on a per dialplan/directory lookup basis<br>
<br>
> 2- On average, how many extensions and active calls (CPS) it can sustain?<br>
<br>
This widely varies based on what you are doing. Calls with no media passed through the switch would be measured much more in calls per second than total calls, and could be in the thousands, calls with heavy video transcoding would be in the double digits on a big box. What you are doing could lie anywhere on that spectrum. Our recommendation is always to test your scenario and figure out sizing for yourself. Typically people with heavy load are very satisfied with our performance..<br>
<br>
> 3- For a call channel, will it create or use system sockets (openfile)? As you know that Asterisk create/open and consume the system sockets for each call.<br>
<br>
Certainly we use sockets for media. We do not use them in the way asterisk does.<br>
<br>
> 4- Will it supports single thread or multithreading?<br>
<br>
FreeSWITCH is heavily multithreaded. Each call will use at least 2 threads, one for each call leg.<br>
<br>
<br>
> 5- Is it scale-able in term of setting up as a cluster(s)?<br>
<br>
Sip itself allows for this. Depends on your exact scenario how to architect in a way to make this scale horizontally, but it can be done in most scenarios<br>
<br>
><br>
> I'll be glad to hear it from you.<br>
><br>
> --<br>
> Regards,<br>
><br>
> Ahmed Munir Chohan<br><br>______________________________<wbr>_________________<br>
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<br></blockquote></div><br><br clear="all"><br>-- <br><div class="gmail_signature" data-smartmail="gmail_signature">Regards,<br><br>Ahmed Munir Chohan<br><br></div>
</div></div></div></div></div>