[OpenSIPS-Users] Freeswitch vs Asterisk

David J. david at styleflare.com
Fri Dec 10 16:36:16 CET 2010


Ok this is a really pointless discussion; Please use Asterisk or 
FreeSWITCH forum for these things. This is not a debate forum.

Thanks to everyone for thei wonderful feedback;



On 12/10/10 10:31 AM, Laszlo wrote:
> Hmm, it's like Ferrari owners talking about which one is better: 
> Volkswagen or Toyota :)
>
> 2010/12/10 Aloysius Lloyd <lloyd.aloysius at gmail.com 
> <mailto:lloyd.aloysius at gmail.com>>
>
>     Paul,
>
>     I do not quite understand what is "find me" doing with NAT
>
>     Thanks
>     Lloyd
>
>
>     On Fri, Dec 10, 2010 at 10:11 AM, Jeff Pyle
>     <jpyle at fidelityvoice.com <mailto:jpyle at fidelityvoice.com>> wrote:
>
>         Guys,
>
>         Point taken.  Personally I prefer Coke over Pepsi.
>
>
>         - Opensips user Jeff
>
>
>         On 12/10/10 10:04 AM, "paul.gore.j at gmail.com
>         <mailto:paul.gore.j at gmail.com>" <paul.gore.j at gmail.com
>         <mailto:paul.gore.j at gmail.com>>
>         wrote:
>
>         >I haven't seen many posts from frustrated peole, majority of
>         them come
>         >from people either selling fs based services or part of fs
>         development
>         >team.
>         >From my experience with fs 1.0.4 it was crashing every 2
>         months, 1.0.6 is
>         >better, I already posted crashing rate for our use case.
>         >I haven't experienced any stabilty issues with * 1.6 yet, but
>         it only
>         >sees light traffic.
>         >FS is a great piece of software but it does have issues,
>         sometimes even
>         >simplest things like "find me" function work flawlessly in *
>         and pain in
>         >the ass to impelement in fs due to either bad nat handling or
>         some other
>         >bugs.
>         >
>         >
>         >-----Original Message-----
>         >From: Erik Dekkers
>         >Sent:  12/10/2010 3:28:11 AM
>         >To: 'paul.gore.j at gmail.com <mailto:paul.gore.j at gmail.com>';
>         'OpenSIPS users
>         > mailling list'
>         >Subject:  RE: [OpenSIPS-Users] Freeswitch vs Asterisk
>         >
>         >The reason people are yelling on the internet "Freeswitch is
>         much better
>         >than asterisk" is pure frustration.
>         >They have used asterisk for years, were faced with crashes
>         and since they
>         >are using freeswitch they don't see those crashes anymore
>         (apart from the
>         >reason of those crashes).
>         >No wonder they tell everyone freeswitch is better than
>         asterisk. From
>         >their point of view asterisk is bad.
>         >
>         >It's not Mr. Collins opinion that asterisk is worse than
>         freeswitch. It
>         >are the ex-asterisk people who are saying that, think about that.
>         >
>         >-----Oorspronkelijk bericht-----
>         >Van: users-bounces at lists.opensips.org
>         <mailto:users-bounces at lists.opensips.org>
>         >[mailto:users-bounces at lists.opensips.org
>         <mailto:users-bounces at lists.opensips.org>] Namens
>         paul.gore.j at gmail.com <mailto:paul.gore.j at gmail.com>
>         >Verzonden: donderdag 9 december 2010 16:27
>         >Aan: OpenSIPS users mailling list
>         >Onderwerp: Re: [OpenSIPS-Users] Freeswitch vs Asterisk
>         >
>         >I just want to reply to mr Collins with FS: your post looks
>         very much
>         >like advertisement, and I have seen that "fs is so much
>         better than *"
>         >all over internet from people connected to fs. That is
>         unethical to say
>         >the least.
>         >In fact we have exprerienced fs crashes with core dump at
>         least  once in
>         >6 months and we process just under 40K calls/month.
>         >As to "nat tools" which you mentioned they just do not work.
>         In fact
>         >usually * box works much better for natted users.
>         >As to xml curl interface - we do use it, and it's a pathetic
>         way to feed
>         >a dialplan to a switch, since it's inefficient resource wise,
>         but there
>         >was no other way available for real time solution where's *
>         supports real
>         >time db out of the box.
>         >Trust me we do have development experience with both * socket
>         interface
>         >and fs one, and in my opinion * solution is far better and
>         has far less
>         >bugs.
>         >
>         >-----Original Message-----
>         >From: James Mbuthia
>         >Sent:  12/08/2010 5:55:42 PM
>         >Subject:  Re: [OpenSIPS-Users] Freeswitch vs Asterisk
>         >
>         >From the comments mentioned it seems FS meets my core
>         requirements which
>         >are scalability and stability. I don't have the financial and
>         manpower
>         >resources for a large scale implementation so am looking at
>         getting a
>         >high end server and a solution that can scale well until I
>         can through in
>         >more resources. It seems also FS is more stable than * which
>         is a huge
>         >plus for a small operation like mine and since I only need
>         few features
>         >from the solutions available then FS makes more sense
>         >
>         >On Wed, Dec 8, 2010 at 8:29 PM, Michael Collins
>         <msc at freeswitch.org <mailto:msc at freeswitch.org>>
>         >wrote:
>         >
>         >> Dave,
>         >>
>         >> Thanks for your two cents. :)
>         >>
>         >> Regarding the PRI stuff, Sangoma is really doing a lot with
>         FreeTDM
>         >> (the replacement for OpenZAP) and it will be a
>         full-featured PRI
>         >> stack. If you're missing anything in the PRI implementation
>         then
>         >> Moises Silva would definitely want to hear about it.
>         >>
>         >> On the voicemail stuff we have heard similar reports. In
>         fact, we have
>         >> an intrepid community member who is building "Jester Mail"
>         as a FS
>         >> alternative to Asterisk's Comedian mail. The basic idea is
>         that Jester
>         >> Mail will be 100% customizable such that you can drop in FS
>         as a
>         >> replacement for Asterisk and your voicemail users would be
>         none the
>         >>wiser.
>         >>
>         >> By early next year you will probably have more options if
>         you wish to
>         >> swap out your remaining Asterisk servers.
>         >>
>         >> -MC
>         >>
>         >>
>         >> On Wed, Dec 8, 2010 at 9:53 AM, Dave Singer
>         >><dave.singer at wideideas.com
>         <mailto:dave.singer at wideideas.com>>wrote:
>         >>
>         >>> 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 <mailto: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
>         <mailto:paul.gore.j at gmail.com> <
>         >>>> paul.gore.j at gmail.com <mailto: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
>         <http://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
>         >>>>>
>         >
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