[OpenSIPS-Users] Freeswitch vs Asterisk

Fri Dec 10 16:31:13 CET 2010

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>

> 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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>>
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