[Freeswitch-users] FreeSwitch General Info

Thu Oct 13 20:57:04 MSD 2016

On Thu, Oct 13, 2016 at 11:41 AM, Michael Jerris <mike at jerris.com> wrote:

>
> On Oct 13, 2016, at 12:31 PM, Ahmed Munir <ahmedmunir007 at gmail.com> wrote:
>
> Few things I forgot to ask, the scale-ability I was referring to as
> horizontal scale and thanks for your input on that.
>
> 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?
>
>
> depends on the box.  Making the requests and parsing them has overhead,
> sure.  As I said, you need to test on your target hardware with your call
> flows to see for sure how it will size.  1kcps on one box is probably high
> to the point of having issues on single box.  Doing so across multiple
> boxes would be advised, i don’t like to put that many calls onto a single
> point of failure.  Will it stress the apache service…. that would be a
> factor of what you have to do over there, and 100% dependent on that code
> answering the request, its impossible for me to answer that question
> generically.
>

xml_curl is a default module implementing a gateway to retrieve config from
a black box http service.  It will only perform as good as that http
service which itself could be a cluster/caching etc.
You can also create custom modules to connect any way you wish into config
and dialplan lookup etc.  (I helped write Asterisk realtime, and at the
time it was a bolt on afterthought, FS is lucky that it was designed from
the beginning with the knowledge gained from Asterisk experience)
We have several other abstractions for integrating with external systems
but there are too many to discuss here.

>
> For point#3, you mean to say Freeswitch doesn't use system files for media
> like Asterisk does?
>
>
> Sure we can use system files for media.  You were asking about additional
> sockets, we have media sockets, if you are opening files of course that
> would use file handles as well.
>

Sockets are used for RTP, HTTP/WSS, Media Files, etc.  The use of many
sockets does not play a role in the performance of software assuming you
tune the kernel to match the load needed.
Most socket related instability comes from the use of the select syscall
which is limited FD numbers below 1024

>
> 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.
>
>
> We are not forking.  You could manually launch multiple processes if you
> choose.
>

In Linux, there is no longer a difference between threads and processes,
they are identical.  (See "man clone")
http://man7.org/linux/man-pages/man2/clone.2.html

 From "man fork"

   *C library/kernel differences*
       Since version 2.3.3, rather than invoking the kernel's *fork*() system
       call, the glibc *fork*() wrapper that is provided as part of the NPTL
       threading implementation invokes clone(2)
<http://man7.org/linux/man-pages/man2/clone.2.html> with flags that
provide the
       same effect as the traditional system call.  (A call to *fork*() is
       equivalent to a call to clone(2)
<http://man7.org/linux/man-pages/man2/clone.2.html> specifying *flags*
as just *SIGCHLD*.)
       The glibc wrapper invokes any fork handlers that have been
       established using *pthread_atfork*(3).

>
>
> From: Michael Jerris <mike at jerris.com>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Cc:
>> Date: Thu, 13 Oct 2016 10:04:56 -0400
>> Subject: Re: [Freeswitch-users] FreeSwitch General Info
>>
>> > On Oct 13, 2016, at 9:52 AM, Ahmed Munir <ahmedmunir007 at gmail.com>
>> wrote:
>> >
>> > Hi,
>> >
>> > Currently I'm browsing different platforms online for open source VoIP
>> technologies as currently using Asterisk as PBX.
>> >
>> > I would like to know following things about FreeSwitch;
>> >
>> > 1- FreeSwitch whether supports realtime integration sip
>> profiles/extensions (using DB) and dialplan? (Just like in Asterisk)
>>
>> yes, most frequently using mod_xml_curl which uses http to pull xml on a
>> per dialplan/directory lookup basis
>>
>> > 2- On average, how many extensions and active calls (CPS) it can
>> sustain?
>>
>> 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..
>>
>> > 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.
>>
>> Certainly we use sockets for media.  We do not use them in the way
>> asterisk does.
>>
>> > 4- Will it supports single thread or multithreading?
>>
>> FreeSWITCH is heavily multithreaded.  Each call will use at least 2
>> threads, one for each call leg.
>>
>>
>> > 5- Is it scale-able in term of setting up as a cluster(s)?
>>
>> 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
>>
>> >
>> > I'll be glad to hear it from you.
>> >
>> > --
>> > Regards,
>> >
>> > Ahmed Munir Chohan
>>
>
>
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