[OpenSIPS-Users] Freeswitch vs Asterisk

Wed Dec 8 19:23:21 CET 2010

Don't know much about freeswitch but now a lot about asterisk.

We use it in a call center environment with predictive dialing.

Typical usage is 3 dialers with 90-120 calls each for outbound.

I see no crashes at all, but I don't use any application other than dialing.

In our business customer pbx`s i find asterisk ok for about 30 users.

Anything more than that is crash show.

And when it starts to crash some times it's almost impossible to trace why.

However it's a great platform to develop fast and easy voice aps through ami
or agi especially for beginners.

Lots of information around books etc. etc.

I look forward to try freeswitch in more heavy things I think it will solve
a lot of stability things.

From: users-bounces at lists.opensips.org
[mailto:users-bounces at lists.opensips.org] On Behalf Of Dave Singer
Sent: Wednesday, December 08, 2010 7:53 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Freeswitch vs Asterisk

We have both asterisk and Freeswitch in production. The primary place where
we have * installed is as a pbx for our business customers (where we started
doing business and didn't know any better). We are still using * for them
for two reasons: migration time and voicemail app I feel is still better in
a couple points. They are low volume usage so crashes are very rare.

We also have some boxes where we connect to telecom PRI circuits where the
API for FS doesn't support some params we need to set. So we are stuck there
for now. There systems handle moderate volume, 30 - 90 simultaneous calls.
This call volume has proved to be deadly to asterisk and we have to restart
asterisk daily or suffer a crash in the middle of peek times.

We use FreeSwitch as the workhorse with a custom routing module combined
with Opensips as a class 4 switch (whole sale trunking service). With high
powered servers (latest dual xeon quad core, 16GB ram, and 10Gbit ethernet)
it can handle thousands of simultaneous calls. They run for months without
problem (would be longer but for reboots for upgrades, etc., not FS
crashes).
We also have a class 5 system that handles residential users which uses FS
and opensips for failover. Again no FS crashes.

FS is also our conference server for all our services.

We started out using * building the business PBXs. Later found FS as we were
developing the residential system and converted to using it.

Coming from * to FS has some difficulties because of the different ways of
doing things like the flow of the dialplan where all conditions are
evaluated at the time of entry to the dialplan, not as each line is executed
(executing another extension solved this problem for me).

I do think FS has a little higher learning curve, I have found it better in
almost every area, especially stability and flexibility.

Well, those are my 2 cents. :-D

Dave

On Tue, Dec 7, 2010 at 11:27 AM, Michael Collins <msc at freeswitch.org> wrote:

Comments inline. (Full disclosure: I am on the FreeSWITCH team, so if I come
off as biased then you know why. ;)

On Tue, Dec 7, 2010 at 8:29 AM, paul.gore.j at gmail.com
<paul.gore.j at gmail.com> wrote:

We use freeswitch in prod alone, no opensips yet. I would say fs is
definetly more scalable than *.
Stability wise seems like fs is on par with *.

YMMV, but a large percentage of FreeSWITCH users have abandoned Asterisk
specifically because of stability issues, like random and inexplicable
crashes.

* has substantially better interface for control over socket connection -
it's easier to implement and it's more consistent.

This statement is patently false. The FreeSWITCH event socket interface is
incredibly powerful and is absolutely more consistent than the AMI. Those
wondering about inconsistencies in the AMI should listen to a seasoned AMI
developer talk about the challenges:

http://www.viddler.com/explore/cluecon/videos/29/

Configuration wise, I think * is easier, xml- based approach in fs is
cumbersome and has no real advantage over *.

This one really is like Coke vs. Pepsi. Some people hate XML, some people
hate INI-style config files. Personally, I've done both and now that I'm
accustomed to FreeSWITCH's XML files I find them much easier to read than
Asterisk's config files. There is one "real advantage" to using XML for
configs and that is that machines and humans can both produce XML, so it's
relatively simple to let a machine generate XML-based configs on the fly.
(FreeSWITCH uses "mod_xml_curl" as the basis for dynamic configuration -
it's very cool and I recommend that you check it out.)

We have endless problems with fs nat handling, lots of no audio issues with
end users behind a nat. That's why we want to try opensips solution for
that.

Almost all NAT problems stem from phones which don't handle NAT properly or
NAT devices that scramble ports and IP addresses when packets pass through.
FreeSWITCH has several NAT-busting tools to assist the system admin. Some
tools are for when FS is behind NAT, others are for when the phones are
behind NAT. Bottom line is this: if the NAT device and the phones are not
horribly broken then FS works great with NAT and in many cases "just works."
However, when you start mixing crazy scenarios with broken phones then bad
things will happen. Example: Polycom phones are wonderful except that they
don't support rport - FS has a mechanism to assist with this but if you turn
it on to "fix" the Polycom phones then it will break all other phone types.
(There is a limit to the amount of pandering that the FS devs will do in
order to interop with broken devices. In many cases they simply say "NO" to
doing stupid things in order to work with broken devices. If you must work
with such a device then perhaps FreeSWITCH isn't for you.)

All that being said, the FreeSWITCH developers have a simple mantra that
they follow to the letter: Use what works for your situation. If Asterisk
works for you then by all means use it! You won't hurt our feelings. (I work
daily with the FreeSWITCH dev team.) If you have people knowledgeable in
Asterisk or FreeSWITCH then it might be advantageous to go with the project
for which you have more resources. In any case, if you are interested in
FreeSWITCH we have a great IRC channel (#freeswitch on irc.freenode.net), an
actively mailing list, and a small but growing international community of
users. You are most welcome to join us to see what we're about.

Happy VoIPing!

-Michael S Collins

IRC:mercutioviz

-----Original Message-----
From: James Mbuthia
Sent:  12/07/2010 8:54:51 AM
Subject:  [OpenSIPS-Users] Freeswitch vs Asterisk

Hi guys,

I want to integrate my Opensips implementation with either Asterisk or
Freeswitch to do the following functions

- Act as a Media server
- Connect to the PSTN
- Act as a B2BUA

There's been alot of hype about Freeswitch and I wanted to know from people
who've integrated it to OpenSIPS how it compares to Asterisk especially in
the case of installation and intergration, scalability and ease of
maintenance.  Any info would be a huge help

regards,
james

:::0:a0e8dc7ff9acb0ae85abefba43f14c73:-1:x:::

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