[OpenSIPS-Users] Freeswitch vs Asterisk

[Michael Collins]

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:::
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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