[OpenSIPS-Users] Freeswitch vs Asterisk

Fri Dec 10 17:57:40 CET 2010

I have 5 Asterisk servers with an average of 6000 calls/day. All work at
least eight months without stopping, somene with stops planned for
maintenance (upgrade, patch, etc...).

Mounting the operating system properly and taking care of the major faults I
see no problem with Asterisk.

The two problems I faced with it Asterisk is up to 200 simultaneous calls,
where Asterisk behaves in a weird way and AMI don't send some events.

This is my experience with Asterisk. I may be correct as I could be wrong.

Best regards,
Rodrigo Lang.

2010/12/10 Laszlo <laszlo at voipfreak.net>

> 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
>>> >>>>>
>>> >
>>> >_______________________________________________
>>> >Users mailing list
>>> >Users at lists.opensips.org
>>> >http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>> >
>>> >
>>> >_______________________________________________
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>>> >Users at lists.opensips.org
>>> >http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
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>>
>>
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-- 
Rodrigo Lang
Opening your mind - Just another Open Source
site<http://openingyourmind.wordpress.com/>
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