[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
> >>>>>
> >
> >_______________________________________________
> >Users mailing list
> >Users at lists.opensips.org <mailto:Users at lists.opensips.org>
> >http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> >
> >_______________________________________________
> >Users mailing list
> >Users at lists.opensips.org <mailto:Users at lists.opensips.org>
> >http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org <mailto:Users at lists.opensips.org>
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org <mailto:Users at lists.opensips.org>
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.opensips.org/pipermail/users/attachments/20101210/5ca5fc2e/attachment-0001.htm>
More information about the Users
mailing list