[OpenSIPS-Users] Freeswitch vs Asterisk

Fri Dec 10 16:27:44 CET 2010

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> 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" <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'; '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] Namens 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>
> >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>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>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
> >>>>>
> >
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> >
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