system toolbox » voip

Asterisk vs. FreeSWITCH

~Anders Brownworth - Wednesday, May 28, 2008 2:00 PM

Over the past several months I have been heavily involved in implementing FreeSWITCH (which just celebrated its first 1.0 release) as an Asterisk replacement. I come from a solid Asterisk production background with several hundred implementations in the field so I would classify my knowledge of Asterisk as fairly good. I must say though that I am very impressed with FreeSWITCH, particularly in the area of performance and flexibility. Given the strengths, I wouldn't be surprised to see FreeSWITCH migration announcements coming out of major Asterisk based PBX vendors within the next year.

Running Asterisk in production with standard hardware, I'm not able to reliably handle over 250 concurrent calls. But with the same hardware, I have tested to 1,000 concurrent calls using FreeSWITCH without any issues. Evidence from others on the FreeSWITCH mailing list claim 3,000 concurrent calls without issues which would seem to approach wire speed! Anecdotally, the first limit you reach with FreeSWITCH might not be with the CPU!

So is FreeSWITCH a drop-in replacement for Asterisk? Not quite, but it is headed somewhat in that direction. Let's look at what it can do versus Asterisk and how it is configured.

http://onruby.flempo.com/2008/10/28/chose-freeswitch-over-asterisk/

Our PBX/IVR web-based generator Telfa has been moved from Asterisk to Freeswitch. Why?

Asterisk just seems to come from a different world than what I am used to. Inflexible and problematic. Very long configuration files with ancient syntax. Now I’m far from pretending I’ve used Asterisk enough to understand it pros and cons well, but I have a decent software development experience and I can tell when something “smells.” I didn’t want to build our system (that I want to be flexible and scale well) on some old technology that is only living from its past.

And (most importantly) there are many people experienced with both Asterisk and Freeswitch favoring the latter: Anders Brownworth, Jonathan Palley (creator of Telegraph, a Rails plugin that lets you talk to Freeswitch), or of course the creator of Freeswitch (and former Asterisk developer!) Anthony Minessale himself.

http://rubyforge.org/pipermail/telegraph-users/2008-July/000136.html

Hello All -
In light of recent discussions on the mailing list, I would like to
let everyone that I have released an initial version of Telegraph for
FreeSWITCH. This code base is completely refactored and reflects my
vision of where Telegraph + Voice/Web Integration should go. It
currently does not support Asterisk, but read on and you will see why
it easily can and should. You can find more API info here:
http://code.google.com/p/telegraph/. I will be presenting at ClueCon
and discussing this.

Some motivations and information about this new version of Telegraph.

a) After dealing with constant Asterisk head-aches, I have ditched it
in favor of FreeSWITCH. Why (warning, the following is my truthful
promotion and endorsement of FS)?
i) Asterisk was built as a PBX. If you are using Telegraph/Rails
you are almost certainly not using it as a PBX. FS is, surprise, a
SWITCH. There is a fundamental and powerful difference.
ii) FreeSWITCH has a great modular architecture, Asterisk is a mess
that endlessly locks. Spend a few hours comparing their code. You
will truly be astonished.
ii) FS is considerably more stable. Hard to believe since its so
new but in my experience we rarely deal with FS problems while
Asterisk was always a culprit.
iv) Unlike Asterisk's AMI/AGI, the FreeSWITCH controlling interfaces are clean and standardized. Take a look at the ami.rb code
to understand what I am talking about when it comes to Asterisk's
quirks.

b) The new Telegraph is built using EventMachine, not TCP.
EventMachine is faster, more stable and doesn't require the use of
Ruby's notoriously bad threads. We have been running it in production
for quiet a while and, again, its considerably more stable than the
TCP library based version.

c) Telegraph is now modular. The FS specific code and the overall
control/rails integration code has been separated. This is why, in
theory, adding an EventMachine based Asterisk interface would be quiet
simple.

We have been running this code with FS in production for the last few
months and its been incredibly smooth. If you don't have a reason to
use Asterisk over FreeSWITCH, use FS. If your reason is that you are
more comfortable with Asterisk, learn FS...you will thank yourself.

There are some folks already working on updating Telegraph to work
with Asterisk in the new codebase. If you are interested in helping I
can connect you. I don't believe in writing code that you do not
really use. Right now, everything I and my company does is with FS.
I could develop for Asterisk (and am happy to consult) but it should
come from someone who is really using this stuff....

Hope this helps!

http://wiki.sangoma.com/wanpipe-freeswitch-config

http://lists.freeswitch.org/pipermail/freeswitch-users/2006-August/000285.html

http://www.richappsconsulting.com/blog/blog-detail/asterisk-vs-freeswitch/

...

Asterisk issues and bugs I have found in the software

I ran into some problems when I tried to deploy Asterisk on a VPS first (Amazon EC2). When I tried to do conferencing, I needed the MeetMe application that provides conferences, and MeetMe depends on Zaptel (which is another piece of software that provides the timing that MeetMe needs). I tried to install Zaptel, but Zaptel refused to work on XEN (the virtualization software that our VPS uses). So I didn't have any other choice but to go with a dedicated server for our VoIP needs. FreeSWITCH works great on a VPS with all the conferencing features, and everything out of the box - no zaptel, ztdummy or anything that could interfere is needed.

...

After all of the work that I put into making Asterisk work on our server, I keep discovering more and more problems with it. The first problem I discovered after everything was working was the DTMF problem. One day an employee asked me why his password wasn't working. I went and looked at the server and his password was there and everything looked good. Then I tried to debug it and I put a function on the dialplan that returned the digits that you send to the server and to my surprise, the server was printing bad/broken digits. So after a lot of research, I have come with the conclusion that the DTMF in Asterisk was broken.

....
One of the nice things about FreeSWITCH is the community. The developers themselves and the users all hang in one channel - #freeswitch (at freenode). The mailing list is also a very nice place. In both places they are very friendly and supportive, unlike the Asterisk/Digium community. They are also open-minded. At one point I asked for a feature (mod_yaml) and the creator of FreeSWITCH (Anthony Minessale) came up with this feature in less than 3 hours - without even knowing what YAML was when the feature was asked for. It was really impressive. He came up with a whole new feature that he hadn't even heard of yet, while the Asterisk devs can't even fix little bugs. Here is the proof of this: lists.freeswitch.org/pipermail/freeswitch-users/2008-June/003954.html

Another nice thing about FreeSWITCH is that it doesn't suffer from the "Not Invented Here" (NIH) syndrome that Asterisk suffers from. FreeSWITCH uses the Sofia SIP stack, a 100% RFC compliant SIP stack, which is very complete, robust and mature. Nokia also actively contributes in this stack along with the FS developers. They also use XML and PCRE (Perl Compatible Regular Expressions) for the dialplan, the Apache Portable Runtime (APR) and SQLite. FreeSWITCH has the ability to load scripts written in LUA, Python, Perl, PHP and XML in the dialplan as "applications” so you have a lot of flexibility and freedom in how you want to configure your VoIP system.

I am very happy that I made the decision to switch to FreeSWITCH. Employees in the company have already reported better audio quality and stability. FreeSWITCH is clearly the winner here. The project is still young and it has a long way to go, but the core and the foundations are very mature and strong and that makes it a great platform.

http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ

XMPP telephony signaling library that is capable of communicating with Google’s GoogleTalk. With a single Jabber account you can receive endless simultaneous calls from GoogleTalk clients and gateway those calls to IVR or another voice protocol like SIP or H.323

ISDN

ISDN comes in two general forms: PRI and BRI. As of this writing, only PRI is supported.

pogledao po source-u

hernad@nmraka-5:~/devel/freeswitch/freeswitch-snapshot/src/mod/endpoints/mod_skypiax$

WHAT IS SKYPIAX

This software (Skypiax) uses the Skype API but is not endorsed, certified or otherwise approved in any way by Skype.

Skypiax is an endpoint (channel driver) that uses the Skype client as an interface to the Skype network, and allows incoming and outgoing Skype calls to/from FreeSWITCH (that canbe bridged, originated, answered, etc. as in all other endpoints, e.g. sofia/SIP).

Skypiax works in FreeSWITCH (FS) on both Linux and Windows, at both 8khz and 16khz (Skype client has 16khz audio I/O). Skypiax works on Asterisk too, at 8khz, on Linux and Windows (through CygWin).

Think of Skypiax as similar to OpenZAP for analog lines. For each channel you need an interface (a Skype client). So, for example, two concurrent calls would need two channels, and therefor two Skype clients running on your FreeSWITCH server.

If your Skype client(s) have Skype credits, then Skypiax works for SkypeOut calls as well.

On Linux the Skype client uses a lot of CPU. To lower its CPU consumption, you can use the Xvfb "fake" X server and (more importantly) the snd-dummy ALSA "fake" sound driver. Scripts are provided for this, though for a low number of channels it should work just fine with normal X servers and ALSA drivers.

On a Linux machine with 3GB ram and a quad core intel6600, we had no problem with 20 concurrent calls, and plenty of head room for perhaps 100 more, (not tested).

On Windows, no need to do anything special the Skype client is lighter on the CPU.

Skypiax is now beta, usable for testing and finding bugs :-).

http://www.freeswitch.org/node/138

...

Missing FS features i'm thinking to working on:
- T.38 (well i don't need it i uses a Linksys SPA to use them directly);
- Queue/Agent (can be implemented using an external daemon, nice to develop a separate call center solution);

So this is the idea of what i got about the asterisk and freeswitch project:

- asterisk: half-open source code indeed the project seems to move in the direction the company that controls him moves to instead what the community demans; poorly source code, lots of bug but also lots of interesting features; supports for Analog/PRI card and some crap support for BRI; stable versions are like beta version and beta version are like experimental code;

- freeswitch: open source code but with heavy review of the quality of the source, perhaps 3 people behind him its a bit too few but perhaps its just because there aren't yet many critical core developers; high quality implementation and design, quickly fix of bugs; lacks support for BRI cards and need more test about OpenZap stuffs; stable versions are really stables;

That's the opinion i built myself from direct work 20/24 hours everyday.

http://wiki.freeswitch.org/wiki/Specsheet

http://wiki.freeswitch.org/wiki/Mod_fax

http://www.freeswitch.org/node/164

FreeSWITCH Now Compiles With AstLinux

Submitted by mcollins on Tue, 02/17/2009 - 19:03.::

We have just received word from Kristian Kielhofner that FreeSWITCH now compiles under AstLinux!

As you may know, AstLinux is a custom Linux distribution centered primarily around Asterisk. However, it lends itself to many different applications, including embedded systems and booting from USB thumb drives. Kristian reports that this configuration takes about 41MB:

• FreeSWITCH with default mods (but not spidermonkey)
• Other mods like mod_xml_curl, mod_snom, mod_lua, and others
• Sample config files
• 8kHz sound files
• Native sounds (G723, G729, GSM, PCMU, PCMA) in 8kHz

If you are looking for a stable, well-tested, lightweight Linux distribution for your FreeSWITCH install then please give AstLinux a try. Kristian will appreciate any feedback that you have to offer. Let's all help make AstLinux a reliable option for FreeSWITCH!

Introduction to Ubuntu Hardy 8.04 and FreeSWITCH

openwrt FreeSWITCH package

New FreeSWITCH Modules, Courtesy of a Great Community!

Submitted by mcollins on Thu, 12/11/2008 - 18:27.
::

We would like to call attention to some new modules that have been added by enterprising members of our community:

• mod_vmd - Voice message beep detection (by Eric Des Courtis from Benbria)

• mod_http - API for fast HTTP operations (by Eric Des Courtis from Benbria)

• mod_limit - Improved limit functionality using hashes (by Mathieu René)

• munin plugin - A plugin to graph channel usage (by William King)

Open Source VoIP: Asterisk or FreeSwitch?

http://wiki.freeswitch.org/wiki/Mod_nibblebill

http://www.wikipbx.org/wikipbx?cat=features

• Multiple 'accounts' per server instance - each account is effectively a completely independent PBX. (multi-tenant)
• Layered configuration - XML files goes on top of what is stored in database. Allows you to use a database, but stays out of your way if you choose to use flat files.
• Configure extensions, SIP endpoints and gateways via web interface
• Simple Voicemail system written in Python as an IVR (w/ source code)
• View/hangup/transfer live calls
• View call history (CDR records) over the web interface
• Web interface for managing (add/edit/update) IVR's written in Python, Javascript or Java
• Easily record "sound clips" for use in dialplan or IVR's
• Inject audio or text-to-speech into live calls

Mozilla public license

http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ

Q: Can I use freeSwitch with isdn BRI/BRA lines (S0 Basic Rate Interface)?

Initial BRI support has been added to the default ISDN module (ozmod_isdn). TE (user) and NT (net) mode are supported, including dialtone in NT mode and overlap receiving. This has been tested/developed with zaptel + bristuff / dahdi (ozmod_zt) and a single port HFC-S PCI card. NOTE: Advanced features (transfer, hold etc.) are not supported by the ozmod_isdn ISDN stack.

http://www.fredshack.com/docs/freeswitch.html

http://mit.edu/sip/sip.edu/

http://alexn.org/docs/dialer.html

We are trying to migrate one of our existing Asterisk setups to FreeSWITCH. These are some tips to walk you thorough the hard part of learning the inners of FreeSWITCH for creating a simple dialer.

As you will learn, FreeSWITCH is a little overwhelming, while being flexible and easy to use.

Being a software developer, I prefer the flexibility of a programming language, rather than working with XML configuration files. Fortunately the standard FreeSWITCH distribution comes with both a SpiderMonkey engine (javascript) and Lua embeded. And it works great for our needs.

ovo isto ali na drugom site-u ?

http://www.sinapticode.com/freeswitch-tips-creating-dialer

http://wikipbx.subwiki.com/installation-manual

• Multiple 'accounts' per server instance - each account is effectively a completely independent PBX. (multi-tenant)
• Layered configuration - XML files goes on top of what is stored in database. Allows you to use a database, but stays out of your way if you choose to use flat files.
• Configure extensions, SIP endpoints and gateways via web interface
• Simple Voicemail system written in Python as an IVR (w/ source code)
• View/hangup/transfer live calls
• View call history (CDR records) over the web interface
• Web interface for managing (add/edit/update) IVR's written in Python, Javascript or Java
• Easily record "sound clips" for use in dialplan or IVR's
• Inject audio or text-to-speech into live calls

• FreeSWITCH
• Twisted.web2 Web Server
• Django WSGI and O/R layer
• PostgreSQL / MySQL / others (any DB the Django O/R layer supports)
• MochiKit Javascript/AJAX library
• mod_xml_curl - HTTP/XML "callback based" dialplan interface
• FreePY Event Socket connection library (Twisted)
• Silk Icon Set

Talk to freeswitch from Adobe AIR application through XMLRPC